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Asterisk Autobuilder a5841a1561 Importing files for 11.0.0-beta1 release. 
git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.0.0-beta1@371123 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10 20:16:10 +00:00

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2012-08-10 Asterisk Development Team <asteriskteam@digium.com>
* Asterisk 11.0.0-beta1 Released.
2012-08-10 20:08 +0000 [r371121] Matthew Jordan <mjordan@digium.com>
* Created Asterisk 11 branch.
Because its one greater then 10.
2012-08-10 19:54 +0000 [r371120] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, channels/chan_misdn.c, channels/chan_sip.c,
main/channel_internal_api.c, main/features.c,
include/asterisk/channel.h, channels/sig_pri.c,
funcs/func_callerid.c, main/cli.c: Add private representation of
caller, connected and redirecting party ids. This patch adds the
feature "Private representation of caller, connected and
redirecting party ids", as previously discussed with us (DATUS)
and Digium. 1. Feature motivation Until now it is quite difficult
to modify a party number or name which can only be seen by
exactly one particular instantiated technology channel
subscriber. One example where a modified party number or name on
one channel is spread over several channels are supplementary
services like call transfer or pickup. To implement these
features Asterisk internally copies caller and connected ids from
one channel to another. Another example are extension
subscriptions. The monitoring entities (watchers) are notified of
state changes and - if desired - of party numbers or names which
represent the involving call parties. One major feature where a
private representation of party names is essentially needed, i.e.
where a party name shall be exclusively signaled to only one
particular user, is a private user-specific name resolution for
party numbers. A lookup in a private destination-dependent
telephone book shall provide party names which cannot be seen by
any other user at any time. 2. Feature Description This feature
comes along with the implementation of additional private party
id elements for caller id, connected id and redirecting ids
inside Asterisk channels. The private party id elements can be
read or set by the user using Asterisk dialplan functions. When a
technology channel is initiating a call, receives an internal
connected-line update event, or receives an internal redirecting
update event, it merges the corresponding public id with the
private id to create an effective party id. The effective party
id is then used for protocol signaling. The channel technologies
which initially support the private id representation with this
patch are SIP (chan_sip), mISDN (chan_misdn) and PRI
(chan_dahdi). Once a private name or number on a channel is set
and (implicitly) made valid, it is generally used for any further
protocol signaling until it is rewritten or invalidated. To
simplify the invalidation of private ids all internally generated
connected/redirecting update events and also all
connected/redirecting update events which are generated by
technology channels -- receiving regarding protocol information -
automatically trigger the invalidation of private ids. If not
using the private party id representation feature at all, i.e. if
using only the 'regular' caller-id, connected and redirecting
related functions, the current characteristic of Asterisk is not
affected by the new extended functionality. 3. User interface
Description To grant access to the private name and number
representation from the Asterisk dialplan, the CALLERID,
CONNECTEDLINE and REDIRECTING dialplan functions are extended by
the following data types. The formats of these data types are
equal to the corresponding regular 'non-private' already existing
data types: CALLERID: priv-all priv-name priv-name-valid
priv-name-charset priv-name-pres priv-num priv-num-valid
priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid
priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE:
priv-name priv-name-valid priv-name-pres priv-name-charset
priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr
priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag
REDIRECTING: priv-orig-name priv-orig-name-valid
priv-orig-name-pres priv-orig-name-charset priv-orig-num
priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan
priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type
priv-orig-subaddr-odd priv-orig-tag priv-from-name
priv-from-name-valid priv-from-name-pres priv-from-name-charset
priv-from-num priv-from-num-valid priv-from-num-pres
priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid
priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag
priv-to-name priv-to-name-valid priv-to-name-pres
priv-to-name-charset priv-to-num priv-to-num-valid
priv-to-num-pres priv-to-num-plan priv-to-subaddr
priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd
priv-to-tag Reported by: Thomas Arimont Review:
https://reviewboard.asterisk.org/r/2030/
2012-08-10 17:56 +0000 [r371113] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Fix a comparison that was causing presence
tests to fail. A recent change made it so that device state
changes that were not actual "changes" would not get reported to
subscribers. The problem was that this inadvertently blocked
presence updates as well.
2012-08-10 16:49 +0000 [r371059-371091] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c, /: remove ALREADYGONE flag on ooh323 call
data by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone
there really. This indication arrive from asterisk core not h.323
stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov
Patches: ASTERISK-19308.patch ........ Merged revisions 371089
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 371090 from
http://svn.asterisk.org/svn/asterisk/branches/10
* addons/ooh323c/src/ooGkClient.c, /: Send re-register packets by
GRQ (gatekeeper request) interval (close issue ASTERISK-20094)
Patches: ASTERISK-20094-2.patch ........ Merged revisions 371060
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 371061 from
http://svn.asterisk.org/svn/asterisk/branches/10
* addons/ooh323c/src/ooTimer.c: restore calling cb functions by
timer expire this was broken in rev 369602
2012-08-10 02:07 +0000 [r371052] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Fix pickup extension channel reference error.
You cannot unref a pointer and then expect to ref it again later.
* Fix potential NULL pointer deref if the call pickup search
fails.
2012-08-09 21:35 +0000 [r371036-371043] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: Introdue 'ooh323 show gk' cli command that
show status of connection to H.323 Gatekeeper (GkClient state)
* addons/ooh323c/src/ooGkClient.c, /: Fix to resend GRQ/RRQ if RRJ
(registration reject) is received (close issue ASTERISK-20094)
Patches: ASTERISK-20094.patch ........ Merged revisions 371011
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 371022 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-08-09 19:22 +0000 [r371030] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /, configure,
include/asterisk/autoconfig.h.in, configure.ac,
channels/sig_pri.c, channels/sig_ss7.c: Use better libss7
detection test and move libpri compile test. ........ Merged
revisions 371012 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 371013 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-08-09 18:28 +0000 [r371010] Alexandr Anikin <may@telecom-service.ru>
* /, addons/ooh323c/src/ooh323ep.c: change opening h323 logfile
with append mode instead of overwrite ........ Merged revisions
370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 370989 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-08-09 17:40 +0000 [r370987] Kinsey Moore <kmoore@digium.com>
* /, apps/app_meetme.c: Correct documentation for the MeetMe x flag
The documentation for the x flag for MeetMe incorrectly described
its function as closing down the conference when the last marked
user left. It actually causes the users with that flag to leave
the conference when the last marked user exits. The functionality
of this flag is not changing. ........ Merged revisions 370985
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 370986 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-08-09 14:52 +0000 [r370979] Mark Michelson <mmichelson@digium.com>
* main/pbx.c, channels/chan_sip.c, include/asterisk/pbx.h,
channels/sip/include/sip.h: Extend extension state callbacks to
have more information. Quote from review board: This patch
extends the extension state callbacks so that monitoring channels
(as chan_sip) get more information of the devices which are
responsible for an extension state change. The additional
information is needed by chan_sip to present names/numbers of the
caller and callee in an early-state SIP notification. Users of
extenstion state callback not interested in the additional
information are not affected by the changes. Motivation: to
present the involved party's name/number in an early-state
nofification (used by the notified device as a pickup offer) one
after another so that a user can see which call he will pick up
in an undirected pickup. Such a pickup offer to a user shall
indicate the same call (number/name-A calls number/name-B) as the
call which would be picked up when an undirected pickup is
executed. Users interested in additional state info must use the
new functions ast_extension_state_add_extended() resp.
ast_extension_state_add_destroy_extended() to register an
extended state callback. When the callback is registered this
way, an extra member device_state_info of struct
ast_state_cb_info is passed to the callback in addition to the
aggregated extension state. This container holds an object for
every device of the monitored extension hint consisting of the
device name, the device state and a channel reference to the
channel which (presumably) caused the device state. The
information is used by chan_sip for early-state notifications.
When the state of a device changes and the new state contains
AST_EVENT_RINGING, an early-state notification is sent to the
subscribed devices with the caller/callee names/numbers of the
oldest ringing channel of the monitored extension. The notified
user may then invoke a direct pickup, which will pickup exactly
this channel. Users of the old non-extended callbacks will only
be called when the aggregated state did change (same behavior as
before). Users of the extended callback will also be called when
the state is unchanged but does contain AST_EVENT_RINGING. That
could be the case if two channels are ringing at one device and
one of them hangs up, so the aggregated state does not change.
This way the monitoring channel can create a new early-state
notification with the now ringing party-ids. Review:
https://reviewboard.asterisk.org/r/2048 This contribution comes
from Guenther Kelleter
2012-08-09 14:36 +0000 [r370978] Jonathan Rose <jrose@digium.com>
* pbx/pbx_dundi.c, CHANGES: DUNDi: Add CLI commands DUNDi show
cache and DUNDi show hints (closes issue ASTERISK-18390) Reported
by: Peter Racz Patches: dundi_cli_cache.patch.v2 uploaded by
Peter Racz (license #6290)
ASTERISK-18390_dundi_cli_cache_jrose_mods_v2.diff uploaded by
Jonathan Rose (license #6182)
2012-08-08 22:45 +0000 [r370955] Michael L. Young <elgueromexicano@gmail.com>
* /, apps/app_chanspy.c: Fix Not Unreferencing A Spied Channel When
a channel hangs up while being spied upon and the option to exit
the ChanSpy application when the spied on channel hangs up is
set, ast_autochan_destroy is not being called and therefore a
reference to the spied upon channel is not removed. The symptom
being reported was that when using func_group in the dialplan and
calling "group show channels" at the cli, the spied upon channel
was still being shown while "core show channels" showed that the
channel was not up. This patch calls ast_autochan_destroy when a
spied upon channel hangs up and the option to exit the ChanSpy
application is set, removing the reference to the channel
allowing the count for the group that the spied channel was part
of to be decremented. (closes issue ASTERISK-17515) Reported by:
Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young
Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael
L. Young (license 5026) ........ Merged revisions 370952 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370954 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-08-08 22:41 +0000 [r370951-370953] Mark Michelson <mmichelson@digium.com>
* CHANGES: Move a SIP change up to the other SIP changes in the
CHANGES file.
* main/channel.c, main/pbx.c, main/manager.c, pbx/pbx_spool.c,
apps/app_originate.c, include/asterisk/channel.h,
include/asterisk/pbx.h, CHANGES, res/res_clioriginate.c: Allow
support for early media on AMI originates and call files. This is
based on the work done by Olle Johansson on review board. The
idea is that the channel specified in an AMI originate or call
file is typically not connected to the outgoing extension until
the channel has been answered. With this change, an EarlyMedia
header can be specified for AMI originates and an early_media
option can be specified in call files. With this option set, once
early media is received on a channel, it will be connected with
the outgoing extension. (closes issue ASTERISK-18644) Reported by
Olle Johansson Review: https://reviewboard.asterisk.org/r/1472
2012-08-08 21:22 +0000 [r370943] Terry Wilson <twilson@digium.com>
* main/manager.c, CHANGES: Add AMI_CLIENT dialplan function
Implementation of a dialplan function for checking manager
accounts. Right now it only returns the number of logged in
sessions for a manager account, but other attributes can be added
later. Patch by: Olle Johansson Review:
https://reviewboard.asterisk.org/r/421/
2012-08-08 20:47 +0000 [r370927] Joshua Colp <jcolp@digium.com>
* main/rtp_engine.c: Create the payload type if it does not exist
when setting information based on the 'm' line. An rtpmap
attribute is not required for defined payload numbers.
2012-08-08 20:32 +0000 [r370926] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_analog.h: Convert sig_analog to use a global
callback table.
2012-08-08 20:30 +0000 [r370925] Kinsey Moore <kmoore@digium.com>
* main/channel.c, /: Do not define a cause that doesn't actually
exist AST_CAUSE_NOTDEFINED is a placeholder for usage when there
is no cause information. As such, it should not be defined and
translatable as a cause. ........ Merged revisions 370923 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370924 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-08-08 20:17 +0000 [r370887-370902] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /,
channels/sig_analog.h: Fix the analog dial *0 flash-hook of
bridged peer feature. The flash-hook the bridged peer feature now
correctly determines if the bridged peer is another chan_dahdi
channel, that it is an analog channel, and that it has the
correct signaling for an FXO port. It now also flash-hooks the
correct channel. ........ Merged revisions 370900 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370901 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c:
Convert sig_pri to use a global callback table.
* channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c:
Convert sig_ss7 to use a global callback table.
2012-08-07 21:58 +0000 [r370881] Damien Wedhorn <voip@facts.com.au>
* build_tools/cflags-devmode.xml, channels/chan_skinny.c: Rewrite
of skinny debugging. Debugging messages and associated controls
only compiled in if configured with --enable-dev-mode. Debug
messages provide more detail (including thread id) and are
grouped so the user/dev can limit the type of messages displayed.
Functionally no real change to chan_skinny. Review:
https://reviewboard.asterisk.org/r/2040/
2012-08-07 19:59 +0000 [r370860] Joshua Colp <jcolp@digium.com>
* main/rtp_engine.c, include/asterisk/rtp_engine.h: Payload and RTP
code are must remain separate since in non-Asterisk format cases
they differ.
2012-08-07 19:26 +0000 [r370851-370859] Kinsey Moore <kmoore@digium.com>
* /: Recorded merge of revisions 370858 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Add
missing AST_CAUSE_* -> text translations ........ Merged
revisions 370856 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* main/channel.c: Add missing AST_CAUSE_* -> text translations A
few of these were missing from the list and are necessary for the
Who Hung Up? functionality.
2012-08-07 17:47 +0000 [r370832-370845] Joshua Colp <jcolp@digium.com>
* main/rtp_engine.c: Fix a bug uncovered by the test suite where
the RTP payload number was not getting set.
* res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c,
channels/chan_motif.c, include/asterisk/rtp_engine.h: Reduce
memory consumption significantly for users of the RTP engine API
by storing only the payloads present and in use instead of every
possible one. Review: https://reviewboard.asterisk.org/r/2052/
2012-08-07 12:46 +0000 [r370820-370831] Matthew Jordan <mjordan@digium.com>
* main/channel.c, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, channels/chan_misdn.c,
channels/chan_sip.c, main/channel_internal_api.c,
channels/misdn/chan_misdn_config.h, main/features.c,
configs/misdn.conf.sample, include/asterisk/channel.h,
configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h,
channels/misdn_config.c: Add named callgroups/pickupgroups This
patch adds named calledgroups/pickupgroups to Asterisk. Named
groups are implemented in parallel to the existing numbered
callgroup/pickupgroup implementation. However, unlike the
existing implementation, which is limited to a maximum of 64
defined groups, the number of defined groups allowed for named
callgroups/pickupgroups is effectively unlimited. Named groups
are configured with the keywords "namedcallgroup" and
"namedpickupgroup". This corresponds to the numbered group
definitions of "callgroup" and "pickupgroup". Note that as the
implementation of named groups coexists with the existing
numbered implementation, a defined named group of "4" does not
equate to numbered group 4. Support for the named groups has been
added to the SIP, DAHDI, and mISDN channel drivers. Review:
https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther
Kelleter(license #6372)
* contrib/realtime/mysql/voicemail_data.sql: Revert r370820 That
change is wrong, wrong, wrong.
* contrib/realtime/mysql/voicemail_data.sql: Update the MySQL
voicemail_data contrib script to reflect Asterisk 11 changes All
voicemails now have a 'msg_id' included in their metadata. The
ODBC message storage backend now requires this column; as such,
the MySQL contrib script that creates the voicemail_data table
has been updated with the appropriate column information.
2012-08-06 15:18 +0000 [r370801] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Improve debug message for temporary
outbound proxies. Thanks to Paul Belanger for pointing this out.
........ Merged revisions 370797 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370798 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-08-03 21:52 +0000 [r370773] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c, channels/sip/config_parser.c,
channels/sip/include/sip.h: Multiple revisions 370769-370771
........ r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri,
03 Aug 2012) | 24 lines Fix error in the "IPorHost" section of a
SIP dialstring. This is based on the review request posted by
Walter Doekes (referenced lower in the commit message) The main
fix here is to treat the IPorHost portion of the dial string as a
temporary outbound proxy. This ensures requests get sent to the
proper location. Due to the age of the request, some parts were
no longer relevant. For instance, the request moved outbound
proxy parsing code into a single method. This is done in a
previous commit, so it was not necessary to do again. Also, the
review request fixed some errors with regards to request routing
for CANCEL and ACK requests. This has also been fixed in more
recent commits. (closes issue ASTERISK-19677) reported by Walter
Doekes Review https://reviewboard.asterisk.org/r/1859 ........
r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug
2012) | 3 lines Remove unused variable. ........ r370771 |
mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5
lines Seriously? Another compilation error fixed. Somebody beat
me. ........ Merged revisions 370769-370771 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370772 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-08-02 15:51 +0000 [r370740] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Fix regression from r370636 When the
chan_sip cleanup went in, a typo was included that caused some
subscriptions of non-Polycom phones to be limited to the same
capabilities as Polycom phones. This resolves the failures in the
test suite resulting from this regression.
2012-08-01 19:37 +0000 [r370726] Mark Michelson <mmichelson@digium.com>
* main/manager.c: Fix a possible crash due to passing NULL to
ast_variables_dup()
2012-08-01 18:52 +0000 [r370720] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/astobj2.h, main/astobj2.c: Make astobj2.h not
include linkedlists.h. Using astobj2 does not require
linkedlists.h be included even though astob2 uses linked lists
internally.
2012-08-01 02:26 +0000 [r370699] Kinsey Moore <kmoore@digium.com>
* /, utils/extconf.c: Revert alloca changes for utils These changes
were a tad overzealous in the utils directory. Unfortunately,
these don't compile with a "make". ........ Merged revisions
370697 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 370698 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-31 22:28 +0000 [r370681-370691] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
channels/sip/include/sip.h: Add headers from SIPAddHeader to
outbound REFER requests. This is a patch from kkm from review
board. This is useful for adding headers to REFER requests that
emanate from a Transfer() dialplan application call. This also
fixes some uses of the Referred-by header, removing an extra set
of angle brackets. I've modified the reporter's original patch to
not require any additions to the sip_refer header and to just
remove the referred_by_name from sip_refer since it is no longer
needed or used. (closes Issue ASTERISK-17639) reported by Kirill
Katsnelson Patches: 019059-sip-refer-addheaders-trunk-353549.diff
uploaded by Kirill Katsnelson (license #5845) Review:
https://reviewboard.asterisk.org/r/1159
* main/manager.c, configs/manager.conf.sample, CHANGES: Add
"setvar" option to manager.conf. With this option set, channel
variables can be set on every manager originate. The Variable
header can still be used to set additional channel variables for
individual calls if desired. This work was completed by Olle
Johansson on review board. I have applied the review feedback and
am committing it in order to get this into trunk before Asterisk
11 is branched. Review: https://reviewboard.asterisk.org/r/1412
2012-07-31 21:20 +0000 [r370677] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Schedule pokes of registered SIP peers
within a given timespan after SIP reload With a large number of
SIP peers registered, performing a SIP reload causes a flood of
SIP OPTIONS request packets. These are immediately sent out, and,
as responses come back, can cause peers to be flagged as 'lagged'
due to handling of the many response messages. This fix prevents
this "packet storm" and schedules the pokes for a random time.
That time varies between 1 ms and the peer's qualify time, or, if
the qualify time is unknown, the global qualifyfreq setting. The
committed patch has some very small modifications to the patch
schmidts wrote for the review. (closes issue ASTERISK-19154)
Reported by: Nicolo Mazzon patches: issue19154.patch license
#6034 uploaded by schmidts Review:
https://reviewboard.asterisk.org/r/1652 ........ Merged revisions
370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 370672 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-31 20:33 +0000 [r370664] Russell Bryant <russell@russellbryant.com>
* main/event.c: Move event cache updates into event processing
thread. Prior to this patch, updating the device state cache was
done by the thread that originated the event. It would update the
cache and then queue the event up for another thread to dispatch.
This thread moves the cache updating part to be in the same
thread as event dispatching. I was working with someone on a
heavily loaded Asterisk system and while reviewing backtraces of
the system while it was having problems, I noticed that there
were a lot of threads contending for the lock on the event cache.
By simply moving this into a single thread, this helped
performance *a lot* and alleviated some deadlock-like symptoms.
Review: https://reviewboard.asterisk.org/r/2066/
2012-07-31 20:21 +0000 [r370655] Kinsey Moore <kmoore@digium.com>
* /, main/say.c, main/threadstorage.c, funcs/func_strings.c,
channels/chan_iax2.c, main/config.c, channels/chan_dahdi.c,
pbx/pbx_spool.c, channels/sig_analog.c, main/strcompat.c,
main/features.c, pbx/pbx_ael.c, main/http.c, pbx/pbx_realtime.c,
channels/chan_alsa.c, channels/sig_ss7.c, main/db.c,
include/asterisk/utils.h, main/pbx.c, funcs/func_cut.c,
tests/test_linkedlists.c, funcs/func_channel.c, apps/app_macro.c,
apps/app_mixmonitor.c, main/asterisk.c, apps/app_voicemail.c,
addons/app_mysql.c, apps/app_meetme.c, apps/app_dictate.c,
main/utils.c, funcs/func_logic.c, cdr/cdr_pgsql.c,
channels/chan_gtalk.c, res/res_jabber.c,
res/res_http_websocket.c, res/ael/pval.c, main/channel.c,
main/manager.c, apps/app_osplookup.c, res/res_agi.c,
apps/app_minivm.c, main/logger.c, main/app.c,
addons/chan_mobile.c, apps/app_while.c, res/res_config_pgsql.c,
channels/chan_sip.c, apps/app_festival.c, pbx/pbx_lua.c,
channels/sig_pri.c, apps/app_getcpeid.c, funcs/func_global.c,
channels/chan_jingle.c, main/tcptls.c,
apps/app_directed_pickup.c, main/file.c, main/callerid.c,
apps/app_sms.c, main/astmm.c, main/event.c, pbx/pbx_dundi.c,
include/asterisk/strings.h, utils/extconf.c, main/dsp.c,
addons/res_config_mysql.c: Clean up and ensure proper usage of
alloca() This replaces all calls to alloca() with ast_alloca()
which calls gcc's __builtin_alloca() to avoid BSD semantics and
removes all NULL checks on memory allocated via ast_alloca() and
ast_strdupa(). (closes issue ASTERISK-20125) Review:
https://reviewboard.asterisk.org/r/2032/ ........ Merged
revisions 370642 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370643 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-31 19:57 +0000 [r370644] Mark Michelson <mmichelson@digium.com>
* CHANGES, pbx/pbx_config.c: Add "dialplan remove context" and
modify "dialplan add include" From corruptor's review board
posting: "I've noticed that we can remove particular extension
from context with dialplan remove extension command but in order
to remove all extensions in the context we should delete them on
by one. I've created dialplan remove context command which uses
ast_context_destroy to destroy the whole context with all
extensions. I've created to functions for in pbx_config.c:
handle_cli_dialplan_remove_context which actually removes context
and complete_dialplan_remove_context which completes input. They
are based on other similar functions and pretty trivial but I can
be mistaken somewhere. "I've also modified dialplan add include
<context2> into <context1>. I've made it similar dialplan add
extension ... command. It creates <context1> if it doesn't exist
and I've also modified complete_dialplan_add_include and removed
check for existance of <context2> because we can include
non-existent context into another one. (I usually include empty
(non-existent) contexts in advance). Should we raise warning in
this case as it's raised while reading extensions.conf? "I use
those functions with AMI. I think manager commands should be
created in addition to those CLI commands." I've addressed the
latest comments on review board and have made some other coding
guidelines-related cleanup. I also have modified the CHANGES file
to mention these new commands. (closes issue ASTERISK-19292)
reported by Andrey Solovyev Patches: dialplan_add_include.patch
uploaded by Andrey Solovyev (license #5214)
dialplan_remove_context.patch uploaded by Andrey Solovyev
(license #5214) Review: https://reviewboard.asterisk.org/r/2042
2012-07-31 19:10 +0000 [r370636] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c, channels/sip/security_events.c,
channels/sip/include/sip.h: Clean up chan_sip This clean up was
broken out from https://reviewboard.asterisk.org/r/1976/ and
addresses the following: - struct sip_refer converted to use the
stringfields API. - sip_{refer|notify}_allocate ->
sip_{notify|refer}_alloc to match other *alloc functions. -
Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not
get_pidf_msg_text_body3 but get_content, to match add_content. -
get_body doesn't get the request body, renamed to
get_content_line. - get_body_by_line doesn't get the body line,
and is just a simple if test. Moved code inline and removed
function. - Remove camelCase in struct sip_peer peer state
variables, onHold -> onhold, inUse -> inuse, inRinging ->
ringing. - Remove camelCase in struct sip_request rlPart1 ->
rlpart1, rlPart2 -> rlpart2. - Rename instances of pvt->randdata
to pvt->nonce because that is what it is, no need to update
struct sip_pvt because _it already has a nonce field_. - Removed
struct sip_pvt randdata stringfield. - Remove useless (and
inconsistent) 'header' suffix on variables in
handle_request_subscribe. - Use ast_strdupa on Event header in
handle_request_subscribe to avoid overly complicated strncmp
calls to find the event package. - Move get_destination check in
handle_request_subscribe to avoid duplicate checking for packages
that don't need it. - Move extension state callback management in
handle_request_subscribe to avoid duplicate checking for packages
that don't need it. - Remove duplicate append_date prototype. -
Rename append_date -> add_date to match other add_xxx functions.
- Added add_expires helper function, removed code that manually
added expires header. - Remove _header suffix on
add_diversion_header (no other header adding functions have
this). - Don't pass req->debug to request handle_request_XXXXX
handlers if req is also being passed. - Don't pass req->ignore to
check_auth as req is already being passed. - Don't create a
subscription in handle_request_subscribe if p->expiry == 0. -
Don't walk of the back of referred_by_name when splitting string
in get_refer_info - Remove duplicate check for no dialog in
handle_incoming when sipmethod == SIP_REFER, handle_request_refer
checks for that. Review: https://reviewboard.asterisk.org/r/1993/
Patch-by: gareth
2012-07-30 23:26 +0000 [r370565-370598] Richard Mudgett <rmudgett@digium.com>
* main/test.c: Tweak unit test warning message.
* funcs/func_presencestate.c, main/test.c: Fix some presence-state
unit test typos.
* apps/app_confbridge.c: DECLINE to load confbridge if the config
fails to load.
* channels/chan_misdn.c, /: Release B channel allocation on error
path in chan_misdn. ........ Merged revisions 370563 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370564 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-30 14:52 +0000 [r370548] Jonathan Rose <jrose@digium.com>
* /, apps/app_meetme.c: app_meetme: Change app_meetme support level
to extended from deprecated (closes issue ASTERISK-20134)
Reported by: Leif Madsen ........ Merged revisions 370547 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-30 13:45 +0000 [r370534-370541] Russell Bryant <russell@russellbryant.com>
* tests/test_event.c: Fix ast_event_new unit test. One of my recent
commits broke this test. The error was:
[test_event.c:event_new_test:214]: Events expected to be
identical have different size: 69 != 59 The difference in size
occurred because the first event had the EID IE added to the
event twice. ast_event_new() now always adds it automatically.
Previously it only added it if there were no IEs specified, which
was kind of weird.
* include/asterisk/event_defs.h, res/res_corosync.c, main/event.c:
Add a "corosync ping" CLI command. This patch adds a new CLI
command to the res_corosync module. It is primarily used as a
debugging tool. It lets you fire off an event which will cause
res_corosync on other nodes in the cluster to place messages into
the logger if everything is working ok. It verifies that the
corosync communication is working as expected. I didn't put
anything in the CHANGES file for this, because this module is new
in Asterisk 11. There is already a generic "res_corosync new
module" entry in there so I figure that covers it just fine.
* addons/app_mysql.c, CHANGES: Allow specifying a port number for
the MySQL server. This patch allows you to specify a port number
for the MySQL server. It's useful if a MySQL server is running on
a non-standard port. Even though this module is deprecated in
favor of func_odbc, someone asked for this feature and it seems
pretty harmless to add. It has been tested using a number of
combinations of with/without a port number specified in the
dialplan and changing the port number for mysqld.
2012-07-26 15:31 +0000 [r370510-370518] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c, CHANGES: chan_sip: Add SIPpeerstatus command
to AMI This patch was submitted by mnicholson a while back. It
adds a new AMI action which allows users to request SIP peer
status on demand similar to existing PeerStatus events and to the
output you would see from CLI with sip show peer Review:
https://reviewboard.asterisk.org/r/1098/
* /, res/res_agi.c: res_agi: Add message indicating need for \n
character in verbose message The while loop responsible for
reading AGI messages from a fastAGI service can end up looping
indefinitely when an AGI script fails to indicate the end of a
message with a \n character. This patch adds an indication that
we are expecting a \n character to end the message to make it
more clear to users that this is necessary if they are receiving
this warning over and over. (issue ASTERISK-20061) Reported by:
Eike Kuiper ........ Merged revisions 370494 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370495 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-25 14:27 +0000 [r370481-370488] Kevin P. Fleming <kpfleming@digium.com>
* main/Makefile: Repair editline builds using in-tree editline
sources. The previous change to the build system for using a
system-provided editline library was missing a crucial include
directory for building against the copy of the library in the
Asterisk source tree.
* main/Makefile: Use an absolute path when referring to the
embedded editline directory. This patch changes the build system
to refer to the embedded editline directory using an absolute
path, which will resolve a problem seen on the CentOS automated
build agents.
* build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, main/Makefile,
main/editline/configure, configure.ac, main/editline/readline
(removed), main/editline/readline.c, main/editline/configure.in,
CHANGES, makeopts.in, main/editline/readline.h (added),
main/asterisk.c, contrib/scripts/install_prereq, main/cli.c:
Enable usage of system-provided NetBSD editline library if
available. This patch changes the Asterisk configure script and
build system to detect the presence of the NetBSD editline
library (libedit) on the system. If it is found, it will be used
in preference to the version included in the Asterisk source
tree. (closes issue ASTERISK-18725) Reported by: Jeffrey C. Ollie
Review: https://reviewboard.asterisk.org/r/1528/ Patches:
0001-Allow-linking-building-against-an-external-editline.patch
uploaded by jcollie (license #5373) (heavily modified by
kpfleming)
2012-07-25 03:51 +0000 [r370474] Terry Wilson <twilson@digium.com>
* main/pbx.c, /: Revert a change that broke compilation 1) There is
no such function as ast_ref() 2) The patch was originally
credited as the one uploaded by Guenther Kelleter (license 6372)
via issue AST-921, but the patch committed was not the patch
referenced on the issue. 3) Guenther Kelleter's patch was
actually correct. It moved the ast_free above the
presencechange_cleanup label. I am not committing his change as
it is not technically necesary--calling ast_free(NULL) is
perfectly safe and I worry that moving the ast_free outside of
the label could lead to future bugs if someone ever adds another
failure conditional and expects 'goto presencechange_cleanup;' to
clean up after everything.
2012-07-24 21:30 +0000 [r370466] Jonathan Rose <jrose@digium.com>
* main/pbx.c, /: Don't attempt free of NULL ptr in pbx.c
handle_presencechange (closes issue AST-921) Reported by:
Guenther Kelleter Patches: nullptr.patch uploaded by Guenther
Kelleter (license 6372)
2012-07-24 19:12 +0000 [r370453] Kevin P. Fleming <kpfleming@digium.com>
* tests/test_acl.c: Silence a warning message from older versions
of GCC. Revision 370426 introduced the use of a nested function
in tests/test_acl.c, but the lack of the 'auto' scope specifier
on the function and a forward declaration resulted in compilation
errors on the automated test systems.
2012-07-24 17:16 +0000 [r370433] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* /, channels/chan_oss.c: chan_oss: fix "sample rate" error message
Merged revisions 370428 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 Merged
revisions 370432 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-24 16:54 +0000 [r370426-370431] Kevin P. Fleming <kpfleming@digium.com>
* main/frame.c, /: Rewrite a comment that didn't adequately explain
the code it was documenting. ........ Merged revisions 370429
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 370430 from
http://svn.asterisk.org/svn/asterisk/branches/10
* CHANGES: Update CHANGES for list/negation ACL feature.
* tests/test_acl.c, main/acl.c: Allow permit/deny ACL lines to
contain multiple items and negated entries. Rules in ACLs
(specified using 'permit' and 'deny') can now contain multiple
items (separated by commas), and items in the rule can be negated
by prefixing them with '!'. This simplifies Asterisk Realtime
configurations, since it is no longer necessray to control the
order that the 'permit' and 'deny' columns are returned from
queries. Review: https://reviewboard.asterisk.org/r/1592/ Initial
patch contributed by Tilghman Lesher Unit tests written by Kevin
P. Fleming
2012-07-24 16:15 +0000 [r370419-370420] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c: Build is underway so logging can go away.
* res/res_rtp_asterisk.c: Temporarily enable pj logging to console
for debugging pjnath issue exposed by build slave.
2012-07-24 08:53 +0000 [r370413] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c: Remove code, that operate with cdr in
attempt_transfer(). That was removed somewhere between 1.2 and
1.4 and acidentaly put back in chan_unistim. (closes issue
ASTERISK-19628) Reported by: Igor Olhovskiy
2012-07-23 21:27 +0000 [r370407] Kevin P. Fleming <kpfleming@digium.com>
* codecs/Makefile, build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, configure.ac,
codecs/codec_ilbc.c, CHANGES, makeopts.in: Enable usage of
system-provided iLBC library. The WebRTC version of the iLBC
codec is now package as a library and is available on some
platforms. This patch allows codec_ilbc to be built against that
library if it is present. Review:
https://reviewboard.asterisk.org/r/1964/
2012-07-23 21:15 +0000 [r370387] Matthew Jordan <mjordan@digium.com>
* tests/test_abstract_jb.c (added), main/abstract_jb.c,
funcs/func_jitterbuffer.c, include/asterisk/abstract_jb.h: Unit
tests for the Jitter Buffer API; remove unnecessary resync This
patch includes the following: * Unit tests for the abstract
Jitter Buffer API. This includes both fixed and adaptive flavors,
testing nominal creation, frame input, frame retrieval,
resyncing; off nominal frame input overflow, out of order, and
others. * Tweaks to the abstract_jb API to remove the unnecessary
resync_threshold parameter from the create function
(resync_threshold is already in the struct passed into the create
function) * Ensure the fixed jitter buffer is empty before
destroying it, to avoid an ASSERT * Don't "resync" the adaptive
jitter buffer. The mechanism that was being used actually causes
the jitter buffer to think its being overflowed by going around
the jitterbuf API and attempting to 'resynch' it improperly. If a
resync is needed, the jitter buffer will do it properly by
itself. Note that this is only an optimization needed for trunk,
as the worst that happens is the loss of three voice packets
before the adaptive jitter buffer will resync anyway. Review:
https://reviewboard.asterisk.org/r/2035
2012-07-23 21:10 +0000 [r370386] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add
separate configuration options for subscription and registration
minexpiry and maxexpiry. This offers more fine-grained control
over how long subscriptions last without negatively affecting the
expiration range for registrations. Uploaded by: Guenther
Kelleter(license #6372) Review:
https://reviewboard.asterisk.org/r/2051
2012-07-23 21:10 +0000 [r370385] Kevin P. Fleming <kpfleming@digium.com>
* /, funcs/func_shell.c: Improve documentation for the SHELL()
dialplan function. ........ Merged revisions 370383 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370384 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-23 21:02 +0000 [r370382] Mark Michelson <mmichelson@digium.com>
* UPGRADE.txt: Add notes to UPGRADE.txt about addition of msg_id to
VoiceMails.
2012-07-23 00:15 +0000 [r370354] Joshua Colp <jcolp@digium.com>
* UPGRADE.txt: Update UPGRADE.txt with notes about ICE support and
res_xmpp.
2012-07-22 23:37 +0000 [r370353] Matthew Jordan <mjordan@digium.com>
* CHANGES: Update CHANGES for Asterisk 11 This updates the CHANGES
file with things that were committed for Asterisk 11, but were
not noted in that file.
2012-07-22 17:03 +0000 [r370347] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c, channels/chan_sip.c,
configs/sip.conf.sample, channels/sip/include/sip.h: Prevent
multiple local candidates from being added with the same
information and add support for disabling ICE on a per-peer
basis. (closes issue ASTERISK-20088) Reported by: wimpy Review:
https://reviewboard.asterisk.org/r/2044/
2012-07-21 13:25 +0000 [r370341] Terry Wilson <twilson@digium.com>
* main/config_options.c, apps/app_confbridge.c,
apps/confbridge/conf_config_parser.c: Fix segfault introduced by
conversion to ACO API The value "none" is specified in the config
file as a valid value for the "video_mode" option. The code prior
to the ACO conversion did not check for "none", but just ignored
it and relied on the default zero value. The parsing with ACO is
more strict, so without handling "none" specifically, parsing
would fail. When parsing failed, but the module loaded anyway,
the config info would never be stored, and one place in the code
did not check for this case and would segfault. It was also
possible that the aco_info struct's internals would be destroyed
and used as well. This patch keeps the module from loading after
parse failures, adds the "none" option to "video_mode", registers
CLI functions only after parsing has completed, checks the config
data for NULL before accessing it, and returns -1 on some
allocation failures when initializing. (closes issue
ASTERISK-20159) Reported by: Birger "WIMPy" Harzenetter Tested
by: Birger "WIMPy" Harzenetter Patches: confbridge_fix3.txt
uploaded by Terry Wilson
2012-07-20 19:36 +0000 [r370335] Jonathan Rose <jrose@digium.com>
* channels/chan_iax2.c: chan_iax2: Fix a segfault introduced by
call ID logging Didn't previously check that a non NULL IAX
channel was stored in the array at the requested position before
attempting iax_pvt_callid_get (closes issue ASTERISK-20145)
Reported by: Birger "WIMPy" Harzenetter
2012-07-20 19:08 +0000 [r370329] Matthew Jordan <mjordan@digium.com>
* apps/app_dial.c: Clean up ManagerEvent Dial documentation The
paragraph describing the SubEvent belongs with the SubEvent
parameter itself, and not with its enum values. The order of
parsing was placing the description after the last enum, which
isn't correct.
2012-07-20 18:37 +0000 [r370328] Kinsey Moore <kmoore@digium.com>
* channels/chan_misdn.c: Fix build error in chan_misdn from commit
370316 chan_misdn was not updated properly to account for a
change in parameters for HANGUPCAUSE functionality. It now builds
properly.
2012-07-20 16:25 +0000 [r370322] Joshua Colp <jcolp@digium.com>
* res/res_http_websocket.exports.in: Export the
ast_websocket_set_nonblock function for use by other modules.
2012-07-20 15:48 +0000 [r370316] Kinsey Moore <kmoore@digium.com>
* funcs/func_hangupcause.c (added), main/channel.c,
channels/chan_dahdi.c, channels/sig_analog.c, main/rtp_engine.c,
channels/chan_sip.c, main/channel_internal_api.c, UPGRADE.txt,
include/asterisk/channel.h, channels/chan_iax2.c,
channels/sig_pri.c, include/asterisk/frame.h, channels/sig_ss7.c:
Add hangupcause translation support The HANGUPCAUSE hash (trunk
only) meant to replace SIP_CAUSE has now been replaced with the
HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan functions to better
facilitate access to the AST_CAUSE translations for
technology-specific cause codes. The HangupCauseClear application
has also been added to remove this data from the channel. (closes
issue SWP-4738) Review: https://reviewboard.asterisk.org/r/2025/
2012-07-20 15:40 +0000 [r370309-370315] Richard Mudgett <rmudgett@digium.com>
* CHANGES: Update CHANGES about adding the AccountCode header to
the AMI Hangup event. (issue ASTERISK-19963)
* main/channel.c: Add the AccountCode header to the AMI Hangup
event. It's harder to correlate the Newchannel and Hangup AMI
events without specifying "AccountCode" in both. (closes issue
ASTERISK-19963) Reported by: Oleg A. Arkhangelsky Patches:
hangup_acctcode.diff (license #6397) patch uploaded by Oleg A.
Arkhangelsky
2012-07-19 23:21 +0000 [r370303] Terry Wilson <twilson@digium.com>
* include/asterisk/config_options.h,
apps/confbridge/include/confbridge.h, main/config_options.c,
apps/confbridge/conf_config_parser.c: Convert app_confbridge to
use the config options framework Review:
https://reviewboard.asterisk.org/r/2024/
2012-07-19 22:25 +0000 [r370298] Richard Mudgett <rmudgett@digium.com>
* /, main/cel.c: Fix compiler warnings. gcc (GCC) 4.2.4 has
problems casting away constness. ........ Merged revisions 370275
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 370277 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-19 22:17 +0000 [r370272-370278] Matthew Jordan <mjordan@digium.com>
* channels/chan_sip.c, res/res_xmpp.c, doc/appdocsxml.dtd,
main/message.c, main/xmldoc.c: Add the ability to specify
technology specific documentation A number of applications/AMI
commands in Asterisk have specific behavioral differences
depending on the resource or channel technology those
applications are executed on. For example, the MessageSend
application/ command is technology agnostic, but how the channel
drivers that support that functionality behave is dependant on
the protocols and channel driver implementation. Prior to this
patch, those details were either documented in the
application/command documentation itself, or were left
undocumented. This patch adds a new element to the documentation
schema, <info/>. An info node is essentially a piece of
technology specific reference information that can be included by
any top level XML documentation node. For example, the
MessageSend application can now include XMPP/SIP specific
information, where that technology specific information can be
defined in chan_motif/res_xmpp/ chan_sip. Likewise, that
information can also be included in the MessageSend AMI command.
Review: https://reviewboard.asterisk.org/r/2049
* /, main/cel.c: Fix compilation error when MALLOC_DEBUG is enabled
To fix a memory leak in CEL, a channel datastore was introduced
whose destruction function pointer was pointed to the ast_free
macro. Without MALLOC_DEBUG enabled this compiles as fine, as
ast_free is defined as free. With MALLOC_DEBUG enabled, however,
ast_free takes on a definition from a different place then
utils.h, and became undefined. This patch resolves this by using
a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this
calls ast_free; when MALLOC_DEBUG is not enabled, this is defined
to be ast_free, which is defined to be free. (issue AST-916)
Reported by: Thomas Arimont ........ Merged revisions 370273 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370274 from
http://svn.asterisk.org/svn/asterisk/branches/10
* res/res_rtp_asterisk.c, /: Handle extremely out of order RFC 2833
DTMF The current implementation of RFC 2833 DTMF handling in
res_rtp_asterisk will, if a packet arrives out of order, drop the
packet. This is to prevent duplicate ton generation in the
Asterisk core. Since the RTP layer does not buffer data itself,
this is the only option the RTP layer currently has for handling
packets that arrive out of order. For the most part, this doesn't
matter. For a particular digit, so long as a BEGIN packet arrives
before the first END packet, the digit will be produced. If
subsequent BEGIN packets arrive interleaved with the ENDs, they
will be dropped; likewise, if the BEGIN or END packets themselves
are out of order, those packets are dropped but sufficient
information is conveyed to the Asterisk core to produce the
appropriate digit. For certain sequences of DTMF packets - most
notably when, for a particular digit, an END packet arrives
before any BEGIN packet for that digit - this is a real problem.
When an END arrives before any BEGINs, the END packet is dropped
- but at the same time, it causes subsequent BEGIN packets for
that digit to be ignored. When the next in order END packet
arrives, it too is dropped - Asterisk believes that there was no
initial BEGIN. The solution this patch provides is to trust the
END packet to convey the information needed for the Asterisk core
to produce the DTMF digit. If we receive an END packet, and it: *
Has a timestamp greater then the last timestamp received from an
END packet * Does not have the same sequence number as the last
received sequence number (and is thus not an END packet
retransmission) Then we send the END frame up to the Asterisk
core. It contains enough DTMF information for Asterisk to produce
the digit. On the other hand, if we receive a BEGIN or
continuation packet that occurs with a timestamp equal to or less
then the last END timestamp, then we've received something out of
order - but we already have received enough information to
produce the digit. These packets are dropped. Much thanks goes to
Olle Johansson (oej) for providing the idea for this solution.
Review: https://reviewboard.asterisk.org/r/2033/ (closes issue
ASTERISK-18404) Reported by: Stephane Chazelas Tested by: Matt
Jordan ........ Merged revisions 370252 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370271 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-19 20:37 +0000 [r370246-370265] Jonathan Rose <jrose@digium.com>
* main/named_acl.c, configs/acl.conf.sample: named_acl: Remove
systemname option from acl.conf, use asterisk.conf value Review:
https://reviewboard.asterisk.org/r/2057/
* main/channel_internal_api.c: CallID Logging: Remove new
line/carriage return from callID change test event
2012-07-19 12:14 +0000 [r370234-370240] Joshua Colp <jcolp@digium.com>
* res/Makefile, res/pjproject/build/os-auto.mak.in: Use the
bruteforce method to get debugging enabled for pjproject.
* res/Makefile: Turn on debugging for pjproject so we can get a
better idea of what is causing the generic CCSS test crash.
2012-07-18 19:48 +0000 [r370225] Jonathan Rose <jrose@digium.com>
* main/channel_internal_api.c: callid logging: Issue test events
when the callid is changed for a channel Review:
https://reviewboard.asterisk.org/r/2054/
2012-07-18 19:18 +0000 [r370187-370211] Kevin P. Fleming <kpfleming@digium.com>
* /, main/cel.c: Resolve severe memory leak in CEL logging modules.
A customer reported a significant memory leak using Asterisk 1.8.
They have tracked it down to
ast_cel_fabricate_channel_from_event() in main/cel.c, which is
called by both in-tree CEL logging modules (cel_custom.c and
cel_sqlite3_custom.c) for each and every CEL event that they log.
The cause was an incorrect assumption about how data attached to
an ast_channel would be handled when the channel is destroyed;
the data is now stored in a datastore attached to the channel,
which is destroyed along with the channel at the proper time.
(closes issue AST-916) Reported by: Thomas Arimont Review:
https://reviewboard.asterisk.org/r/2053/ ........ Merged
revisions 370205 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370206 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/channel.c, addons/app_mysql.c, main/pbx.c,
funcs/func_curl.c, /, main/ccss.c, funcs/func_odbc.c,
funcs/func_lock.c, apps/app_macro.c, channels/chan_iax2.c,
apps/app_mixmonitor.c, apps/app_stack.c, funcs/func_global.c,
res/res_odbc.c: Ensure that all ast_datastore_info structures are
'const'. While addressing a bug, I came across a instance of
'struct ast_datastore_info' that was not declared 'const'. Since
the API already expects them to be 'const', this patch changes
the declarations of all existing instances that were not already
declared that way. ........ Merged revisions 370183 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370184 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-18 15:15 +0000 [r370171-370177] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c: Fix a crash in pjnath when starting an
ICE connectivity check and immediately destroying the ICE
session. The initial ICE connectivity check is scheduled as a
timer item that is to be executed immediately. It is possible for
this timer item to start executing while the ICE session it is
working on is destroyed. To reduce the chance of this any timer
items that need to be immediately executed will be executed
within the thread that has started the initial ICE connectivity
check.
* channels/chan_sip.c, include/asterisk/rtp_engine.h: Fix a crash
occurring as a result of excess stack usage. This fix involves
moving the allocation of some temporary codec structures to the
heap and also reduces the number of maximum payloads to something
more sane for both regular and low memory builds. (closes issue
ASTERISK-20140) Reported by: jonnt
2012-07-18 07:17 +0000 [r370165] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c, configs/unistim.conf.sample, CHANGES:
Added option 'interdigit_timer' to unistim.conf to make able
controll hardcoded dial timeout constant.
2012-07-17 19:05 +0000 [r370152-370157] Joshua Colp <jcolp@digium.com>
* res/res_xmpp.c: Add pubsub unsubscription support so
subscriptions do not linger for MWI and device state progatation.
The pubsub code did not attempt to remove subscriptions at all.
This has now changed so that if a client is being disconnected it
will unsubscribe. It will also unsubscribe at connection time so
if it unexpectedly disconnected duplicate subscriptions will not
occur. (closes issue ASTERISK-19882) Reported by: mattvryan
* include/asterisk/xmpp.h, res/res_xmpp.c: Fix a crash as a result
of propagating MWI or device state over XMPP when the client is
disconnected. The MWI and device state propagation code wrongly
assumes that an XMPP client connection will remain established at
all times. This fix corrects that by making the lifetime of the
subscription the same as the lifetime of the connection itself.
As the connection is established and disconnected the
subscription itself is created and destroyed. (closes issue
ASTERISK-18078) Reported by: elguero
2012-07-16 19:58 +0000 [r370133] Walter Doekes <walter+asterisk@wjd.nu>
* /, channels/chan_sip.c: Code cleanup and bugfix in chan_sip
outboundproxy parsing. The bug was clearing the global
outboundproxy when a peer-specific outboundproxy was bad. The
cleanup reduces duplicate code. Review:
https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark
Michelson ........ Merged revisions 370131 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370132 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-16 19:14 +0000 [r370111-370126] Joshua Colp <jcolp@digium.com>
* res/res_xmpp.c: Fix an issue where a service discovery request
could crash Asterisk. A server sending a service discovery
request to us may or may not put a from attribute in the message.
If the from attribute is present use it in the to attribute for
the result. If the from attribute is not present do not add a to
attribute. (issue ASTERISK-16203) Reported by: wubbla
* res/res_xmpp.c: Fix a bug where some XMPP servers would reject
authentication. We need to use the user portion of the JID and
not the full configured username.
* res/res_xmpp.c: Add missing namespace for old non-SASL based
authentication.
* channels/chan_sip.c: Fix a bug exposed by the testsuite where
text streams would no longer be parsed correctly.
2012-07-16 14:02 +0000 [r370083] Kinsey Moore <kmoore@digium.com>
* /, UPGRADE-10.txt, CHANGES, UPGRADE-1.8.txt: Add comments about
the BUILD_NATIVE change This is a significant change and mention
of it should have gone into UPGRADE.txt and CHANGES. ........
Merged revisions 370081 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370082 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-16 12:58 +0000 [r370072-370073] Joshua Colp <jcolp@digium.com>
* res/res_xmpp.c: Fix an issue where specifying the resource in the
username would cause authentication to fail.
* channels/sip/sdp_crypto.c, channels/chan_sip.c,
channels/sip/security_events.c,
include/asterisk/http_websocket.h, configs/sip.conf.sample,
CHANGES, res/res_http_websocket.c, channels/sip/include/sip.h:
Add support for SIP over WebSocket. This allows SIP traffic to be
exchanged over a WebSocket connection which is useful for rtcweb.
Review: https://reviewboard.asterisk.org/r/2008
2012-07-16 07:38 +0000 [r370066-370067] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c: Deactivate timer for dialing entered
number on hook switch hang up. (closes issue ASTERISK-19554)
Reported by: Stefano Villani
* channels/chan_unistim.c, contrib/unistimLang/fr.po (added),
CHANGES: Add French translation for chan_unistim phones on-screen
menus.
2012-07-13 18:41 +0000 [r370055-370060] Joshua Colp <jcolp@digium.com>
* include/asterisk/format.h, res/res_format_attr_h263.c (added),
res/res_format_attr_h264.c (added): Reduce memory consumption and
add the H.264 and H.263 modules I shamefully neglected to add.
* main/format.c, channels/chan_sip.c, main/translate.c,
include/asterisk/format.h, res/res_format_attr_silk.c,
res/res_format_attr_celt.c: Add support for parsing SDP
attributes, generating SDP attributes, and passing it through.
This support includes codecs such as H.263, H.264, SILK, and
CELT. You are able to set up a call and have attribute
information pass. This should help considerably with video calls.
Review: https://reviewboard.asterisk.org/r/2005/
2012-07-13 00:05 +0000 [r370048] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* contrib/scripts/live_ast: live_ast: don't set working directory
contrib/scripts/live_ast currently assumes that it is being run
from the top-level directory of the source tree. It creates a
script that will also set the working directory. This fix avoids
the need to set the working directory if the caller sets
LIVE_AST_BASE_DIR instead. It relies on realpath for that. If
realpath is not available, it will fall back to the original
behaviour. Review: https://reviewboard.asterisk.org/r/2027/
2012-07-12 21:43 +0000 [r370043] Terry Wilson <twilson@digium.com>
* include/asterisk/config_options.h,
configs/config_test.conf.sample, main/config_options.c,
tests/test_config.c: Handle deprecated (aliased) option names
with the config options api Add a simple way to register
"deprecated" option names that alias to a different "current"
name. Review: https://reviewboard.asterisk.org/r/2026/
2012-07-12 20:28 +0000 [r370037] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /: Add missing
ast_hangup() calls on some analog exception paths. Make starting
analog_ss_thread() or __analog_ss_thread() failure paths hangup
the channel. ........ Merged revisions 370017 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370025 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-12 20:06 +0000 [r369995-370016] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Include Expires header for SIP PUBLISH
requests RFC3903 requres SIP PUBLISH requests to have Expires
headers, so add them. Review:
https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth
........ Merged revisions 370014 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 370015 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c: Prevent double uri_escaping in chan_sip
when pedantic is enabled If pedantic mode is enabled, outbound
invites will have double-escaped contacts. This avoids setting an
already-escaped string into a field where it is expected to be
unescaped. (closes issue ASTERISK-20023) Reported by: Walter
Doekes ........ Merged revisions 369993 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369994 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-12 14:38 +0000 [r369972-369974] Michael L. Young <elgueromexicano@gmail.com>
* /, funcs/func_math.c: Correct Documentation For DEC Function The
documentation for DEC in func_math.c was incorrect. Looks like a
copy and paste error. (Closes issue ASTERISK-20095) Reported by:
Billy Chia Tested by: Michael L. Young Patches: func_math.patch
uploaded by Billy Chia (license 6381) ........ Merged revisions
369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 369971 from
http://svn.asterisk.org/svn/asterisk/branches/10
* funcs/func_math.c: Reverting last merge since it wasn't completed
properly.
* funcs/func_math.c: Correct Documentation For DEC Function The
documentation for DEC in func_math.c was incorrect. Looks like a
copy and paste error. (Closes issue ASTERISK-20095) Reported by:
Billy Chia Tested by: Michael L. Young Patches: func_math.patch
uploaded by Billy Chia (license 6381) ........ Merged revisions
369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-07-11 18:33 +0000 [r369959] Jonathan Rose <jrose@digium.com>
* include/asterisk/acl.h, channels/chan_sip.c,
include/asterisk/config.h, main/acl.c,
include/asterisk/channel.h, configs/manager.conf.sample,
channels/chan_iax2.c, CHANGES, main/named_acl.c (added),
main/config.c, main/loader.c, configs/iax.conf.sample,
main/manager.c, include/asterisk/event_defs.h,
configs/extconfig.conf.sample, configs/sip.conf.sample,
channels/sip/include/sip.h, main/asterisk.c,
configs/acl.conf.sample (added): Named ACLs: Introduces a system
for creating and sharing ACLs This patch adds Named ACL
functionality to Asterisk. This allows system administrators to
define an ACL and refer to it by a unique name. Configurable
items can then refer to that name when specifying access control
lists. It also includes updates to all core supported consumers
of ACLs. That includes manager, chan_sip, and chan_iax2. This
feature is based on the deluxepine-trunk by Olle E. Johansson and
provides a subset of the Named ACL functionality implemented in
that branch. For more information on this feature, see acl.conf
and/or the Asterisk wiki. Review:
https://reviewboard.asterisk.org/r/1978/
2012-07-11 17:16 +0000 [r369940] Tilghman Lesher <tilghman@meg.abyt.es>
* /, main/ast_expr2.h, main/ast_expr2f.c, res/ael/ael_lex.c,
funcs/func_realtime.c, main/ast_expr2.c: Allow the REALTIME()
function to report errors back to the caller. Also, do more error
checking on the arguments specified to the REALTIME() function
and clarify the documentation. While I was editing the file, a
few coding guidelines fixups, as well. Review:
https://reviewboard.asterisk.org/r/2031/ ........ Merged
revisions 369937 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369938 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-11 17:14 +0000 [r369939] Matthew Jordan <mjordan@digium.com>
* main/features.c: Don't perform an XInclude to a document node
that may not always be present Because some of the manager events
are defined in the top of the source, due to the macro calls not
containing all necessary information to have the documentation
colocated with the call itself, several include statements were
failing when built with 'make'. While this did not cause any
problems in compilation or validation, it did result in a number
of warnings being dumped to stderr. This patch changes those
references such that they always resolve, regardless of the
documentation build options.
2012-07-11 16:42 +0000 [r369936] Joshua Colp <jcolp@digium.com>
* channels/chan_motif.c: Do not consider failure to read the
configuration file in chan_motif to be a show stopper for loading
Asterisk by returning decline instead of failure. (closes issue
ASTERISK-20103) Reported by: Terry Wilson
2012-07-11 02:06 +0000 [r369905-369910] Matthew Jordan <mjordan@digium.com>
* main/cdr.c, main/channel.c, channels/sig_analog.c, main/logger.c,
channels/sig_pri.c, main/asterisk.c, main/loader.c: Fix
validation errors when producing documentation using default
build script The awk script parses out the first instance of the
DOCUMENTATION tag that it finds within a file. If a file did not
previously have a DOCUMENTATION tag but received one due to it
having an AMI event, then the XML fragment associated with the
AMI event was erroneously placed in the resulting XML file.
Without the python scripts, these XML fragments will not
validate. This patch adds DOCUMENTATION tags at the top of those
files that did not previously have them to prevent the awk script
from pulling AMI event documentation.
* main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c,
channels/chan_local.c, channels/sig_analog.c, main/manager.c,
channels/chan_agent.c, main/features.c, main/logger.c,
channels/sig_pri.c, doc/appdocsxml.dtd, main/asterisk.c,
main/loader.c: Add some additional documentation for core AMI
events This patch adds some basic documentation for a number of
modules. This includes core source files in Asterisk (those in
main), as well as chan_agent, chan_dahdi, chan_local, sig_analog,
and sig_pri. The DTD has also been updated to allow referencing
of AMI commands.
2012-07-10 15:36 +0000 [r369900] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c: Fix failing SDP_offer_answer test Asterisk
now generates image stream declinations with the same transport
case that it used to before the stream declination improvements.
(udptl vs UDPTL) (closes issue SWP-4736)
2012-07-10 15:25 +0000 [r369873-369898] Joshua Colp <jcolp@digium.com>
* channels/chan_motif.c: Add additional description stanza names
from the old Google Talk protocol which is used with Google
Voice. (closes issue ASTERISK-20114) Reported by: Malcolm
Davenport
* channels/chan_motif.c: Respect codec preference order when adding
codecs to a media description. This change allows an endpoint in
motif.conf to be configured with a preference of G.722 and
fallback of ulaw. With Google this allows communication with
Google Talk clients to use G.722 while when using Google Voice
ulaw will be used. (closes issue ASTERISK-20114) Reported by:
Malcolm Davenport
2012-07-10 13:40 +0000 [r369872] Kinsey Moore <kmoore@digium.com>
* main/pbx.c, /, apps/app_stack.c: Improve Goto and GotoIf related
documentation Correct documentation on labeliftrue and
labeliffalse parameters of GotoIf() and update several other
locations that use the same syntax. (closes issue ASTERISK-20007)
Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged
revisions 369869 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369871 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-10 13:34 +0000 [r369870] Matthew Jordan <mjordan@digium.com>
* main/libasteriskssl.c: Fix initial loading problem with res_curl
When the OpenSSL duplicate initialization issues were resolved in
r351447, res_curl could fail to load if it checked
SSL_library_init after SSL initialization completed. This is due
to the SSL_library_init stub returning a value of 0 for success,
as opposed to a value of 1. OpenSSL uses a value of 1 to indicate
success - in fact, SSL_library_init is documented to always
return 1. Interestingly, the CURL libraries actually checked the
return value - the fact that nothing else that depends on OpenSSL
was having problems loading probably means they don't check the
return value. (closes issue AST-924) Reported by: Guenther
Kelleter patches: (AST-924.patch license #6372 uploaded by
Guenther Kelleter)
2012-07-10 11:49 +0000 [r369837-369864] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c, channels/chan_motif.c: Add required items
for Google video support. This adds legacy STUN support for RTCP
sockets, adds RTCP candidates to the Google transport
information, and adds required codec parameters. (closes issue
ASTERISK-20106) Reported by: Malcolm Davenport
* main/stun.c: When receiving a STUN binding request send one out
as the Google Talk client uses this as a method to determine if
the remote party is still reachable or not. Failure to do this
results in the Google Talk client ignoring RTP packets after a
specific period of time. This is also done as a result of
receiving a STUN binding request so that the username information
can be used from the inbound request, thus not requiring it to be
stored on a per candidate basis. (closes issue ASTERISK-20107)
Reported by: Malcolm Davenport
* channels/chan_sip.c: Add support for exposing the received
contact URI and also for setting the request URI in messages.
(closes issue AST-911)
* channels/chan_motif.c: Force the clock rate of G.722 to be 16000
when using the Google transports as it is 8000 elsewhere. (closes
issue ASTERISK-20105) Reported by: Malcolm Davenport
* configs/motif.conf.sample: Document that multiple endpoints using
the same connection is not supported. (closes issue
ASTERISK-20104) Reported by: Malcolm Davenport
2012-07-09 17:07 +0000 [r369820] Jason Parker <jparker@digium.com>
* configs/sip_notify.conf.sample, /: Add Digium phones context to
sip_notify sample config. This makes it so that they can be
reconfigured remotely. (closes issue ASTERISK-19910) ........
Merged revisions 369818 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369819 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-09 16:44 +0000 [r369811-369817] Joshua Colp <jcolp@digium.com>
* res/res_rtp_asterisk.c: Fix an issue where media would not flow
for situations where the legacy STUN code is in use. The STUN
packets should *not* be blocked by strict RTP. (closes issue
ASTERISK-20102) Reported by: Malcolm Davenport
* res/res_xmpp.c: Add additional namespaces for Google Talk which
are used for the gmail client. (closes issue ASTERISK-20101)
Reported by: Malcolm Davenport
* channels/chan_motif.c: Fix dependency to be on res_xmpp. Long ago
in a galaxy far far away it used to use res_jabber.
2012-07-09 14:54 +0000 [r369794] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: chan_sip: Fix small behavioral change
accidentally introduced in r369750 When removing the warning for
AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed
the return value, which would likely make the indication not be
sent in audio. This fixes that while still removing the warning
message. ........ Merged revisions 369792 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369793 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-07 17:06 +0000 [r369769] Joshua Colp <jcolp@digium.com>
* res/res_xmpp.exports.in (added), include/asterisk/xmpp.h,
channels/chan_motif.c (added), UPGRADE.txt,
channels/chan_gtalk.c, res/res_xmpp.c, CHANGES, res/res_jabber.c,
configs/motif.conf.sample (added): Add a new unified Jingle,
Google Jingle, and Google Talk channel driver written from
scratch called chan_motif. This channel driver is a replacement
for both chan_gtalk and chan_jingle but adds additional features
not found in either. These features include full configuration
reload, video, full codec support, bidirectional cause code
mapping, hold, unhold, and ringing indication. It is also
compliant with the current published Jingle and Google Jingle
specifications. The original Google Talk protocol is also
supported for Google Voice interoperability. You may ask yourself
though where the name motif comes from... and I would say to
you... music! motif: a perceivable or salient recurring fragment
or succession of notes Sorta like a jingle! Review:
https://reviewboard.asterisk.org/r/1917/
2012-07-06 22:03 +0000 [r369765] Kinsey Moore <kmoore@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c,
channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c:
Remove unnecessary generation of informational cause frames It is
not necessary to generate information cause code frames on every
protocol event that occurs. This removes all the instances where
the frame was not conveying a cause code and was instead just
conveying a protocol-specific message. This also corrects the
generation of the message associated with disconnects for MFC/R2
to use the MFC/R2 specific text for the disconnect cause.
2012-07-06 21:28 +0000 [r369764] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: chan_sip: Add case for FLASH control
frames so that we don't display a warning. chan_sip channels can
receive flash control frames when connected to analog phones and
possibly for other reasons. There really isn't a reason to warn
when these frames are received, we can safely ignore them.
Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license
6182) ........ Merged revisions 369750 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369751 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-06 18:49 +0000 [r369710-369733] Mark Michelson <mmichelson@digium.com>
* main/tcptls.c, /: Remove a superfluous and dangerous freeing of
an SSL_CTX. The problem here is that multiple server sessions
share a SSL_CTX. When one session ended, the SSL_CTX would be
freed and set NULL, leaving the other sessions unable to
function. The code being removed is superfluous because the
SSL_CTX structures for servers will be properly freed when
ast_ssl_teardown is called. (closes issue ASTERISK-20074)
Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded
by Mark Michelson (license #5049) Testers: Trevor Helmsley
........ Merged revisions 369731 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369732 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/bridging.c: Fix bridging thread leak. The bridge thread
was exiting but was never being reaped using pthread_join(). This
has been fixed now by calling pthread_join() in
ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by
Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012
........ Merged revisions 369708 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369709 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-06 14:32 +0000 [r369703] Joshua Colp <jcolp@digium.com>
* res/pjproject/pjnath/include/pjnath/ice_session.h,
res/pjproject/pjnath/src/pjnath/ice_session.c: Import revision
4196 from pjproject trunk. Fix a crash issue when starting ICE
connectivity checks and immediately destroying the ICE session.
This was exposed by the SIP CCSS test. Full fix for this issue
will be worked on as a medium to long term roadmap item. pjroject
issue viewable at https://trac.pjsip.org/repos/ticket/1548
2012-07-05 21:36 +0000 [r369681] Matthew Jordan <mjordan@digium.com>
* res/res_stun_monitor.c, CHANGES: Add 'stun show status' command
This patch adds a new CLI command, 'stun show status'. This
command will show a table describing all known STUN servers and
statuses. (closes issue ASTERISK-18046) Reported by: Jeremy
Kister Tested by: Jeremy Kister patches:
(stun-show-status-v4-trunk.patch license #6232 uploaded by Jeremy
Kister) Review: https://reviewboard.asterisk.org/r/2001
2012-07-05 19:36 +0000 [r369677] Richard Mudgett <rmudgett@digium.com>
* res/pjproject/pjmedia/include/pjmedia,
res/pjproject/pjsip/include/pjsip,
res/pjproject/pjlib/include/pj/compat,
res/pjproject/pjmedia/include/pjmedia-codec: Make res/pjproject
ignore more files.
2012-07-05 19:36 +0000 [r369676] Kinsey Moore <kmoore@digium.com>
* /, apps/app_voicemail.c: AST-2012-011: Resolve heap corruption
issue with voicemail The heard and deleted arrays in the
voicemail state structure were not handled properly following the
memory leak fix in r354890 and a fix for an invalid free in
r356797. This could result in accessing and writing into freed
memory. The allocation for these arrays has been reworked to
avoid the possibility of invalid frees, access of freed memory,
and crashes that were occurring as a result of this. Locking
around accesses and modifications of the voicemail state
structure members dh_arraysize, heard, and deleted has been added
to prevent simultaneous modification and access when IMAP storage
is in use. If IMAP storage is not in use, this locking is not
compiled in. Review: https://reviewboard.asterisk.org/r/1994/
(closes issue ASTERISK-19923) Reported by: Dan Delaney Tested by:
Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by
kmoore (license 6273) ........ Merged revisions 369652 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369653 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-05 19:32 +0000 [r369666-369673] Richard Mudgett <rmudgett@digium.com>
* res/pjproject/pjsip/src/pjsip-ua,
res/pjproject/pjsip-apps/src/ipjsystest/ipjsystest.xcodeproj,
res/pjproject/pjnath/src/pjnath-test,
res/pjproject/third_party/build/speex,
res/pjproject/third_party/build/gsm/output,
res/pjproject/pjmedia/include/pjmedia-codec,
res/pjproject/third_party/build/baseclasses,
res/pjproject/third_party/build/srtp,
res/pjproject/pjsip-apps/src/samples,
res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin,
res/pjproject/pjlib/include/pj++,
res/pjproject/tests/pjsua/scripts-call,
res/pjproject/third_party/srtp/doc,
res/pjproject/pjsip-apps/src/pocketpj/output,
res/pjproject/pjnath/bin,
res/pjproject/third_party/srtp/crypto/replay,
res/pjproject/pjsip/include/pjsip,
res/pjproject/third_party/build/speex/speex,
res/pjproject/build.symbian, res/pjproject/third_party/bin,
res/pjproject/pjsip/src/pjsua-lib,
res/pjproject/third_party/srtp/include,
res/pjproject/third_party/portaudio/doc, res/pjproject/lib,
res/pjproject/pjmedia/include/pjmedia-videodev,
res/pjproject/pjlib/bin,
res/pjproject/third_party/srtp/crypto/cipher,
res/pjproject/third_party/build/speex/output,
res/pjproject/pjlib-util/src/pjlib-util,
res/pjproject/third_party/portaudio/test,
res/pjproject/third_party/build/gsm,
res/pjproject/third_party/portaudio/include,
res/pjproject/pjsip-apps/src/pjsua_wince,
res/pjproject/pjsip/include/pjsip-simple,
res/pjproject/pjmedia/src/pjmedia-codec,
res/pjproject/tests/pjsua,
res/pjproject/pjsip-apps/src/pocketpj/res,
res/pjproject/pjsip-apps/src/3rdparty_media_sample,
res/pjproject/third_party/gsm/inc,
res/pjproject/pjsip-apps/build/wince-evc4,
res/pjproject/pjsip-apps/src/ipjsua/Resources-iPad,
res/pjproject/third_party/portaudio/src/hostapi,
res/pjproject/third_party/portaudio/build, res/pjproject/build,
res/pjproject/third_party/build/resample,
res/pjproject/third_party/speex/include,
res/pjproject/pjsip/src/pjsip,
res/pjproject/pjlib/build/wince-evc4,
res/pjproject/pjsip-apps/src/symbian_ua_gui/group,
res/pjproject/pjsip-apps/src/symbian_ua,
res/pjproject/tests/pjsua/wavs,
res/pjproject/third_party/portaudio/src/os/win,
res/pjproject/pjsip-apps/src/ipjsua/Classes,
res/pjproject/pjmedia/include/pjmedia,
res/pjproject/tests/pjsua/scripts-sendto,
res/pjproject/third_party/gsm/src,
res/pjproject/third_party/portaudio/build/msvc,
res/pjproject/pjsip-apps/src/confbot,
res/pjproject/pjnath/src/pjturn-client,
res/pjproject/pjlib-util/build/output,
res/pjproject/third_party/BaseClasses,
res/pjproject/third_party/portaudio/src/hostapi/wasapi,
res/pjproject/third_party/portaudio/src/hostapi/wdmks,
res/pjproject/pjlib/src/pj/compat,
res/pjproject/third_party/srtp/crypto/include,
res/pjproject/third_party/speex/include/speex,
res/pjproject/third_party/gsm/add-test,
res/pjproject/pjsip/build,
res/pjproject/pjsip-apps/src/pjsua_wince/output,
res/pjproject/third_party/gsm/lib, res/pjproject/pjsip,
res/pjproject/pjsip-apps/src/pjsystest,
res/pjproject/third_party/portaudio/src,
res/pjproject/third_party/speex/libspeex,
res/pjproject/pjsip/build/wince-evc4/output,
res/pjproject/pjlib-util/src/pjlib-util-test,
res/pjproject/pjsip-apps/src/symsndtest,
res/pjproject/third_party/srtp/tables,
res/pjproject/third_party/g7221, res/pjproject/pjmedia/include,
res/pjproject/pjlib/include/pj,
res/pjproject/third_party/build/portaudio/output,
res/pjproject/pjsip-apps/bin,
res/pjproject/pjsip-apps/src/ipjsua/ipjsua.xcodeproj,
res/pjproject/pjsip-apps/src/pjsua,
res/pjproject/third_party/srtp/test,
res/pjproject/pjsip/include/pjsip-ua,
res/pjproject/third_party/resample,
res/pjproject/third_party/build/ilbc,
res/pjproject/pjmedia/src/pjmedia-audiodev,
res/pjproject/pjsip-apps/src/ipjsua,
res/pjproject/third_party/srtp/srtp,
res/pjproject/third_party/build/milenage,
res/pjproject/pjmedia/src/pjmedia, res/pjproject/pjlib-util,
res/pjproject/third_party/portaudio/src/common,
res/pjproject/third_party/portaudio/bindings/cpp,
res/pjproject/pjlib-util/build/wince-evc4/output,
res/pjproject/third_party/srtp/crypto/kernel,
res/pjproject/tests/pjsua/scripts-pres, res/pjproject/pjnath,
res/pjproject/pjsip/build/output,
res/pjproject/pjsip-apps/build/output,
res/pjproject/pjsip-apps/build, res/pjproject/tests/automated,
res/pjproject/pjnath/build/wince-evc4/output,
res/pjproject/third_party/portaudio/src/hostapi/asio,
res/pjproject/pjnath/include/pjnath,
res/pjproject/pjsip/src/test,
res/pjproject/pjsip-apps/src/symbian_ua_gui/gfx,
res/pjproject/pjsip/bin,
res/pjproject/third_party/build/portaudio,
res/pjproject/pjlib/build/output, res/pjproject/pjmedia/src,
res/pjproject/pjlib/src/pj, res/pjproject/pjlib,
res/pjproject/pjlib/build/wince-evc4/output,
res/pjproject/pjmedia/src/test/vectors,
res/pjproject/third_party/portaudio/src/hostapi/jack,
res/pjproject/pjmedia/src/pjmedia-codec/g722,
res/pjproject/third_party/portaudio/src/hostapi/coreaudio,
res/pjproject/pjmedia/build/output,
res/pjproject/pjlib-util/include/pjlib-util,
res/pjproject/third_party/portaudio/src/hostapi/asihpi,
res/pjproject/third_party/milenage, res/pjproject/pjnath/src,
res/pjproject/tests/pjsua/scripts-run,
res/pjproject/pjlib-util/build/wince-evc4,
res/pjproject/pjmedia/lib, res/pjproject/pjmedia/src/test,
res/pjproject/third_party/speex/symbian,
res/pjproject/third_party/speex/win32,
res/pjproject/third_party/srtp/crypto/test,
res/pjproject/pjlib-util/bin,
res/pjproject/third_party/portaudio/build/scons,
res/pjproject/tests/cdash,
res/pjproject/tests/pjsua/scripts-media-playrec,
res/pjproject/third_party/build/portaudio/src,
res/pjproject/pjlib/src, res/pjproject/third_party/mp3,
res/pjproject/pjnath/lib, res/pjproject/third_party/build/g7221,
res/pjproject/third_party/gsm/man,
res/pjproject/third_party/portaudio/src/os/unix,
res/pjproject/third_party/portaudio/bindings,
res/pjproject/pjsip-apps/src/python,
res/pjproject/pjnath/src/pjnath, res/pjproject/third_party/lib,
res/pjproject/third_party/portaudio/src/os/mac_osx,
res/pjproject/third_party/srtp/crypto/ae_xfm,
res/pjproject/pjsip-apps/bin/samples,
res/pjproject/pjnath/src/pjturn-srv,
res/pjproject/third_party/portaudio/pablio,
res/pjproject/pjlib/lib, res/pjproject/third_party/g7221/decode,
res/pjproject/pjlib/include/pj/compat,
res/pjproject/third_party/gsm,
res/pjproject/third_party/build/baseclasses/output,
res/pjproject/third_party/build/srtp/output,
res/pjproject/third_party/srtp, res/pjproject/pjnath/build,
res/pjproject/tests/pjsua/scripts-sipp, res/pjproject/pjsip-apps,
res/pjproject/pjnath/build/wince-evc4,
res/pjproject/third_party/srtp/crypto/rng,
res/pjproject/pjsip/build/wince-evc4,
res/pjproject/pjsip-apps/build/wince-evc4/output,
res/pjproject/third_party/gsm/tst,
res/pjproject/third_party/portaudio/src/hostapi/dsound,
res/pjproject/third_party/portaudio/testcvs,
res/pjproject/pjsip-apps/src/ipjsystest/Classes,
res/pjproject/pjlib/build, res/pjproject/third_party/portaudio,
res/pjproject/third_party/portaudio/src/hostapi/wmme,
res/pjproject/pjlib-util/docs,
res/pjproject/pjmedia/include/pjmedia-audiodev,
res/pjproject/pjsip-apps/src/vidgui,
res/pjproject/pjlib/src/pjlib-test,
res/pjproject/pjsip-apps/src/py_pjsua,
res/pjproject/third_party/portaudio/src/os,
res/pjproject/pjsip/include,
res/pjproject/pjmedia/build/wince-evc4,
res/pjproject/pjmedia/src/pjmedia-videodev,
res/pjproject/pjsip-apps/src, res/pjproject/third_party/speex,
res/pjproject/third_party/gsm/tls,
res/pjproject/third_party/g7221/common,
res/pjproject/tests/pjsua/tools,
res/pjproject/third_party/resample/include,
res/pjproject/third_party/build/samplerate/output,
res/pjproject/third_party/build/samplerate,
res/pjproject/third_party/gsm/bin,
res/pjproject/pjsip/src/pjsip-simple,
res/pjproject/third_party/g7221/encode,
res/pjproject/pjlib/src/pjlib-samples,
res/pjproject/pjsip-apps/lib,
res/pjproject/pjsip-apps/src/ipjsystest,
res/pjproject/pjlib-util/include,
res/pjproject/third_party/build/resample/output,
res/pjproject/third_party/build/ilbc/output,
res/pjproject/third_party/srtp/crypto,
res/pjproject/pjsip-apps/src/python/samples, res/pjproject/tests,
res/pjproject/pjsip-apps/src/symbian_ua_gui/sis,
res/pjproject/pjnath/include,
res/pjproject/pjsip-apps/src/symbian_ua_gui,
res/pjproject/pjmedia/build, res/pjproject/pjmedia,
res/pjproject/third_party/build/milenage/output,
res/pjproject/pjlib-util/build, res/pjproject/pjsip/src,
res/pjproject/pjmedia/build/wince-evc4/output,
res/pjproject/third_party/portaudio/src/hostapi/alsa,
res/pjproject/pjsip-apps/docs,
res/pjproject/pjsip-apps/src/symbian_ua_gui/inc,
res/pjproject/pjsip-apps/src/symbian_ua_gui/data,
res/pjproject/tests/pjsua/scripts-pesq,
res/pjproject/third_party/srtp/pjlib,
res/pjproject/pjlib/include, res/pjproject/pjnath/build/output,
res/pjproject/third_party/srtp/crypto/hash,
res/pjproject/build/vs, res/pjproject/pjlib/docs,
res/pjproject/third_party/build,
res/pjproject/third_party/resample/src,
res/pjproject/third_party, res/pjproject/pjlib/src/pjlib++-test,
res/pjproject/third_party/build/g7221/output,
res/pjproject/third_party/srtp/crypto/math,
res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/src/pocketpj,
res/pjproject/tests/pjsua/scripts-recvfrom,
res/pjproject/third_party/portaudio/build/dev-cpp,
res/pjproject/pjsip/include/pjsua-lib,
res/pjproject/pjsip-apps/src/symbian_ua_gui/src, res/pjproject,
res/pjproject/third_party/portaudio/src/hostapi/oss,
res/pjproject/pjlib-util/src, res/pjproject/third_party/ilbc:
Make res/pjproject ignore some generated files.
* include/asterisk/utils.h: Tweak some comments and whitespace in
utils.h
2012-07-05 18:11 +0000 [r369644] Jonathan Rose <jrose@digium.com>
* apps/app_mixmonitor.c: app_mixmonitor: Fix a reference leak in
manager_mixmonitor function Manager_mixmonitor included an early
return on failed executions of mixmonitor that would result in a
leaked channel reference. (closes issue ASTERISK-19943) Reported
by: Mark Murawski Patches: mixmonitor-trunk-368394.patch uploaded
by Mark Murawski (license 5791)
2012-07-05 17:03 +0000 [r369628] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Do not send a BYE when a provisional
response arrives during a re-INVITE Commits r369557 and r369579
were done to improve handling of re-INVITEs when the UA that was
supposed to receive the re-INVITE fails to respond. A limitation
of those patches occurred when a UA sent a provisional response
to the re-INVITE. This triggered a sending of a BYE in
check_pending. This patch tweaks the handling of the re-INVITE
such that a BYE is not sent in response to those messages. (issue
ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies
patches: (reinvite_tweak.diff license #5012 by Steve Davies)
........ Merged revisions 369626 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369627 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-05 11:42 +0000 [r369602-369620] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooCmdChannel.c,
addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c:
Fix dev mode ooh323 warnings
* addons/chan_ooh323.c, addons/ooh323c/src/ooq931.h,
addons/ooh323c/src/ooCalls.h, configs/chan_ooh323.conf.sample
(removed), addons/ooh323c/src/ooh323ep.c, CHANGES,
configs/ooh323.conf.sample (added),
addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c,
addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooLogChan.h,
addons/ooh323c/src/ooStackCmds.h, addons/ooh323c/src/ooh245.c,
addons/ooh323cDriver.c, addons/ooh323c/src/ooh245.h,
addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c:
Added direct media support to ooh323 channel driver options are
documented in config sample sample config rename to proper name -
ooh323.conf To change media address ooh323 send empty TCS if
there was completed TCS exchange or send facility
forwardedelements with new fast start proposal if not. Then close
transmit logical channels and renew TCS exchange. If new fast
start proposal is received then ooh323 stack call back channel
driver routine to change rtp address in the rtp instance. If
empty TCS is received then close transmit logical channels and
renew TCS exchange Review:
https://reviewboard.asterisk.org/r/1607/
* addons/ooh323cDriver.c: fix small mistake in the previous
* addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/ooCapability.c,
addons/ooh323c/src/decode.c, addons/ooh323c/src/perutil.c,
addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c,
addons/ooh323c/src/ooq931.c: Fix modern gcc warning Review:
https://reviewboard.asterisk.org/r/1767
2012-07-03 17:07 +0000 [r369559-369581] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: More improvements to re-INVITEs timing
out after a provisional response There is no need to call
check_pendings() on a final response to an INVITE when destroying
the scheduler entry as it will be done later during normal
processing. (issue ASTERISK-19992) ........ Merged revisions
369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 369580 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c, channels/sip/include/sip.h: Better handle
re-INVITEs with provisional but no final repsonses A previous
attempt at fixing this issue had negative side effects related to
attended transfers which this patch should resolve. Many thanks
to Steve Davies for all of the good suggestions and testing.
(closes issue ASTERISK-19992) Reported by: Steve Davies Tested
by: Steve Davies, Terry Wilson Review:
https://reviewboard.asterisk.org/r/2009/ ........ Merged
revisions 369557 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369558 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-07-02 14:06 +0000 [r369517-369527] Joshua Colp <jcolp@digium.com>
* configs/xmpp.conf.sample (added), include/asterisk/xmpp.h
(added), configs/cli_aliases.conf.sample, res/res_xmpp.c (added):
Add a cleaned up drop-in replacement for res_jabber called
res_xmpp. This provides the same externally facing functionality
but is implemented differently internally. This is currently not
built by default but this will be changed once chan_jingle2
(insert actual name in your head when reading this after it has
been merged) is in the tree. Review:
https://reviewboard.asterisk.org/r/1983/
* res/res_rtp_asterisk.c: Ensure the timer heap is protected by a
lock.
* res/pjproject/pjlib/include/pj/config_site.h: Enable IPv6 support
in pjproject.
* res/res_rtp_asterisk.c: Don't try to send connectivity checks on
RTCP if RTCP is no longer present and don't do multiple ICE
connectivity checks at once.
* res/pjproject/pjlib/src/pj/sock_qos_common.c (added),
res/pjproject/pjlib-util/src/pjlib-util/crc32.c (added),
res/pjproject/pjsip/src/pjsip-simple/xpidf.c (added),
res/pjproject/third_party/gsm/src/gsm_implode.c (added),
res/pjproject/tests/pjsua/scripts-sipp/uas-cancel-no-final.xml
(added), res/pjproject/build.symbian/pjmedia.mmp (added),
res/pjproject/third_party/build/portaudio/src/pa_hostapi.h
(added), res/pjproject/pjlib/src/pjlib-test/fifobuf.c (added),
res/pjproject/pjlib/src/pj/file_access_unistd.c (added),
res/pjproject/third_party/gsm/src/toast_ulaw.c (added),
res/pjproject/pjsip/include/pjsip/sip_transport_tls.h (added),
res/pjproject/pjsip/include/pjsip/sip_multipart.h (added),
res/pjproject/pjmedia/src/pjmedia/errno.c (added),
res/pjproject/pjsip-apps/src/pjsua_wince/pjsua_wince.vcp (added),
res/pjproject/third_party/speex/COPYING (added),
res/pjproject/pjlib/src/pj/os_core_darwin.m (added),
res/pjproject/third_party/ilbc/packing.c (added),
res/pjproject/third_party/build/portaudio/src/pa_mac_core_internal.h
(added),
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res/pjproject/tests/pjsua/scripts-sendto/140_sdp_with_direction_attr_in_session_1.py
(added),
res/pjproject/pjsip-apps/src/pjsystest/pjsystest_wince.rc2
(added), res/pjproject/pjlib/include/pj/compat/os_win32.h
(added), res/pjproject/pjmedia/include/pjmedia/doxygen.h (added),
res/pjproject/pjsip/src/test/main_rtems.c (added),
res/pjproject/pjlib-util/include/pjlib-util/scanner_cis_bitwise.h
(added), res/pjproject/pjsip-apps/src/ipjsystest/main.m (added),
res/pjproject/build.symbian/pjsip.mmp (added),
res/pjproject/third_party/speex/include/speex/speex_jitter.h
(added), res/pjproject/tests/pjsua/run.py (added),
res/pjproject/third_party/speex/symbian (added),
res/pjproject/tests/pjsua/scripts-pesq/200_codec_l16_8000_stereo.py
(added), res/pjproject/pjsip-apps/src/samples/auddemo.c (added),
res/pjproject/tests/pjsua/scripts-sendto/300_srtp_crypto_case_insensitive.py
(added), res/pjproject/third_party/g7221/common/basic_op.c
(added),
res/pjproject/pjnath/build/wince-evc4/pjnath_test_wince.vcp
(added), res/pjproject/third_party/g7221/common/basic_op.h
(added), res/pjproject/third_party/portaudio/config.guess
(added), res/pjproject/third_party/portaudio/src/os/unix (added),
res/pjproject/third_party/speex/libspeex/cb_search_sse.h (added),
res/pjproject/tests/pjsua/tools/Makefile (added),
res/pjproject/pjlib/src/pj/compat/longjmp_i386.S (added),
res/pjproject/third_party/portaudio/pablio (added),
res/pjproject/build.symbian/symbian_ua_udeb.pkg (added),
res/pjproject/README.txt (added),
res/pjproject/third_party/srtp/srtp.vcproj (added),
res/pjproject/pjnath/build (added),
res/pjproject/third_party/portaudio/src/hostapi/dsound (added),
res/pjproject/tests/automated/prepare.xml.template (added),
res/pjproject/pjsip/src/pjsua-lib/pjsua_pres.c (added),
res/pjproject/tests/pjsua/scripts-sipp/uas-reinv-and-ack(same-branch)-without-sdp.xml
(added), res/pjproject/pjlib/build (added),
res/pjproject/third_party/build/baseclasses/libbaseclasses.vcproj
(added),
res/pjproject/third_party/speex/include/speex/speex_preprocess.h
(added), res/pjproject/pjlib/src/pjlib-test (added),
res/pjproject/pjsip-apps/src/symbian_ua_gui/data/symbian_ua_gui.l01
(added), res/pjproject/pjlib/build/privkey.pem (added),
res/pjproject/pjmedia/src/pjmedia/alaw_ulaw_table.c (added),
res/pjproject/configure-legacy (added),
res/pjproject/tests/pjsua/scripts-sendto/200_ice_success_1.py
(added), res/pjproject/pjsip/include/pjsip/sip_transport.h
(added), res/pjproject/pjnath/src/pjturn-srv/server.c (added),
res/pjproject/pjmedia/build/os-linux.mak (added),
res/pjproject/pjlib/include/pj/compat/os_win32_wince.h (added),
res/pjproject/pjsip/src/pjsip-ua/sip_replaces.c (added),
res/pjproject/third_party/portaudio/src/common/pa_util.h (added),
res/pjproject/pjsip-apps/src/symbian_ua_gui/inc/symbian_ua_guiDocument.h
(added), res/pjproject/pjlib/src/pj/fifobuf.c (added),
res/pjproject/third_party/gsm/tls/sour1.dta (added),
res/pjproject/pjsip/include/pjsip/sip_types.h (added),
res/pjproject/pjlib/include/pj/compat/time.h (added),
res/pjproject/pjsip/src/pjsip/sip_auth_msg.c (added),
res/pjproject/tests/pjsua/scripts-sendto/001_torture_4475_3_1_1_1.py
(added), res/pjproject/pjsip/include/pjsip_ua.h (added),
res/pjproject/pjlib/build/Makefile (added),
res/pjproject/third_party/srtp/README (added),
res/pjproject/tests/pjsua/scripts-sendto/311_srtp1_recv_avp.py
(added), res/pjproject/pjsip-apps/src/pjsua/main_rtems.c (added),
res/pjproject/pjsip-apps/src/pocketpj/res/invisibl.bmp (added),
res/pjproject/pjlib/src/pjlib-test/rtems_network_config.h
(added), res/pjproject/third_party/srtp/crypto/math/stat.c
(added), res/pjproject/third_party/srtp/test/replay_driver.c
(added), res/pjproject/pjmedia/src/pjmedia-audiodev/audiotest.c
(added), res/pjproject/pjlib/src/pjlib++-test (added),
res/pjproject/pjsip-apps/src/samples/streamutil.c (added),
res/pjproject/pjmedia/src/pjmedia/ffmpeg_util.c (added),
res/pjproject/tests/pjsua/scripts-sendto/500_pres_subscribe_with_bad_event.py
(added), res/pjproject/third_party/srtp/install-sh (added),
res/pjproject/tests/pjsua/scripts-pesq/200_codec_speex_16000.py
(added),
res/pjproject/third_party/srtp/crypto/cipher/null_cipher.c
(added), res/pjproject/pjmedia/src/pjmedia/ffmpeg_util.h (added),
res/pjproject/pjlib-util/src (added),
res/pjproject/pjsip/include/pjsip/sip_config.h (added),
res/pjproject/pjlib/docs/doxygen.cfg (added): Add support for
ICE/STUN/TURN in res_rtp_asterisk and chan_sip. Review:
https://reviewboard.asterisk.org/r/1891/
2012-06-29 20:32 +0000 [r369512] Mark Michelson <mmichelson@digium.com>
* main/rtp_engine.c, /: Fix apparent copy and paste error where
incorrect "glue" is used. ........ Merged revisions 369511 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-29 17:02 +0000 [r369493] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, main/channel.c, main/autoservice.c, main/pbx.c,
channels/chan_local.c, funcs/func_channel.c,
main/channel_internal_api.c, main/features.c,
configs/cdr.conf.sample, include/asterisk/channel.h,
include/asterisk/pbx.h, CHANGES, apps/app_followme.c,
apps/app_queue.c: Hangup handlers - Dialplan subroutines that run
when the channel hangs up. Hangup handlers are an alternative to
the h extension. They can be used in addition to the h extension.
The idea is to attach a Gosub routine to a channel that will
execute when the call hangs up. Whereas which h extension gets
executed depends on the location of dialplan execution when the
call hangs up, hangup handlers are attached to the call channel.
You can attach multiple handlers that will execute in the order
of most recently added first. (closes issue ASTERISK-19549)
Reported by: Mark Murawski Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/2002/
2012-06-29 16:56 +0000 [r369492] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: With some configurations a transport is
not actually specified so assume UDP in these cases. ........
Merged revisions 369490 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369491 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-29 16:42 +0000 [r369489] Richard Mudgett <rmudgett@digium.com>
* main/channel_internal_api.c, .cleancount: Remove obsolete struct
ast_channel note. The opaquing the ast_channel struct no longer
requires .cleancount to be changed when the struct is changed. *
Bump .cleancount value one last time because of struct
ast_channel for old times sake.
2012-06-29 15:33 +0000 [r369473] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Make the address family filter specific
to the transport. (closes issue ASTERISK-16618) Reported by: Leif
Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........
Merged revisions 369471 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369472 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-28 01:12 +0000 [r369449-369454] Terry Wilson <twilson@digium.com>
* include/asterisk/config_options.h,
configs/config_test.conf.sample, main/config_options.c,
tests/test_config.c: Add the ability to set flags via the config
options api Allows the setting of flags via the config options
api. For example, code like this: #define OPT1 1 << 0 #define
OPT2 1 << 1 #define OPT3 1 << 2 struct thing { unsigned int
flags; }; and a config like this: [blah] opt1=yes opt2=no
opt3=yes Review: https://reviewboard.asterisk.org/r/2004/
* /, channels/chan_sip.c, channels/sip/include/sip.h: AST-2012-010:
Clean up after a reinvite that never gets a final response The
basic problem is that if a re-INVITE is sent by Asterisk and it
receives a provisional response, but no final response, then the
dialog is never torn down. In addition to leaking memory, this
also leaks file descriptors and will eventually lead to Asterisk
no longer being able to process calls. This patch just keeps
track of whether there is an outstanding re-INVITE, and if there
is goes ahead and cleans up everything as though there was no
outstanding reinvite. Review:
https://reviewboard.asterisk.org/r/2009/ (closes issue
ASTERISK-19992) Reported by: Steve Davies Tested by: Steve
Davies, Terry Wilson ........ Merged revisions 369436 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369437 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-26 21:45 +0000 [r369414] Jonathan Rose <jrose@digium.com>
* include/asterisk/logger.h, channels/chan_dahdi.c,
main/autoservice.c, main/pbx.c, channels/chan_local.c,
channels/sig_analog.c, main/channel_internal_api.c,
channels/chan_agent.c, main/features.c, main/logger.c,
channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c,
main/bridging.c, main/cli.c: Unique Call ID logging Phases III
and IV Adds call ID logging changes to specific channel drivers
that weren't handled handled in phase II of Call ID Logging. Also
covers logging for threads for threads created by systems that
may be involved with many different calls. Extra special thanks
to Richard for rigorous review of chan_dahdi and its various
signalling modules. review:
https://reviewboard.asterisk.org/r/1927/ review:
https://reviewboard.asterisk.org/r/1950/
2012-06-26 13:23 +0000 [r369370-369392] Matthew Jordan <mjordan@digium.com>
* /, main/adsi.c: Fix crash in unloading of res_adsi module When
res_adsi is unloaded, it removes the ADSI functions that it
previously installed by passing a NULL adsi_funcs pointer to
ast_adsi_install_funcs. This function was not checking whether or
not the adsi_funcs pointer passed in was NULL before
dereferencing it to check whether or not the version of the
functions matches what the core was expecting it. This patch
makes it so that the version is only checked if a potentially
valid adsi_funcs pointer was passed in. Passing in NULL removes
the installed functions, bypassing the version check. ........
Merged revisions 369390 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369391 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/manager.c: Update "manager show event" to support tab
completion Thank you rmudgett for pointing out that I was missing
this in the initial check-in for AMI event documentation
(r369346)
* main/cdr.c, /: Fix incorrect duration reporting in CDRs created
in batch mode Certain places in core/cdr.c would, if the duration
value were 0, calculate the duration as being the delta between
the current time and the time at which the CDR record was
started. While this does not typically cause a problem in
non-batch mode, this can cause an issue in batch mode where CDR
records are gathered and written long after those calls have
ended. In particular, this affects calls that were never
answered, as those are expected to have a duration of 0. Often,
this would result in CDR logs with a significant number of calls
with lengthy durations, but dispositions of "BUSY". Note that
this does not affect cdr_csv, as that backend does not use
ast_cdr_getvar and instead directly reports the duration value.
The affected core backends include cdr_apative_odbc and
cdr_custom; other extended or deprecated CDR backends may
potentially still directly manipulate the duration values. (issue
ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883)
Reported by: Thomas Arimont Tested by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1996/ ........ Merged
revisions 369351 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369369 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-25 19:26 +0000 [r369367] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c, channels/sip/include/sip.h: Re-fix how
local tag is generated when sending a 481 to an INVITE. Match our
local tag to whatever to-tag was sent in the initial INVITE.
Because the size of the to-tag may not fit in the buffer in the
sip_pvt, it has been changed to a string field. (closes issue
ASTERISK-19892) reported by Walter Doekes Review:
https://reviewboard.asterisk.org/r/1977 ........ Merged revisions
369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 369353 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-25 17:59 +0000 [r369346] Matthew Jordan <mjordan@digium.com>
* apps/app_dial.c, apps/app_meetme.c, configure.ac,
apps/app_userevent.c, CHANGES, apps/app_queue.c, Makefile,
build_tools/get_documentation.py (added), main/manager.c,
configure, build_tools/post_process_documentation.py (added),
include/asterisk/xmldoc.h, apps/app_confbridge.c, makeopts.in,
apps/app_stack.c, apps/app_chanspy.c, doc/appdocsxml.dtd,
main/xmldoc.c, apps/app_voicemail.c: Add AMI event documentation
This patch adds the core changes necessary to support AMI event
documentation in the source files of Asterisk, and adds
documentation to those AMI events defined in the core application
modules. Event documentation is built from the source by two new
python scripts, located in build_tools: get_documentation.py and
post_process_documentation.py. The get_documentation.py script
mirrors the actions of the existing AWK get_documentation
scripts, except that it will scan the entirety of a source file
for Asterisk documentation. Upon encountering it, if the
documentation happens to be an AMI event, it will attempt to
extract information about the event directly from the manager
event macro calls that raise the event. The
post_process_documentation.py script combines manager event
instances that are the same event but documented in multiple
source files. It generates the final core-[lang].xml file. As
this process can take longer to complete than a typical 'make
all', it is only performed if a new make target, 'full', is
chosen. Review: https://reviewboard.asterisk.org/r/1967/
2012-06-25 16:07 +0000 [r369329] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Fix Bridge application occasionally returning
to the wrong location. * Fix do_bridge_masquerade() getting the
resume location from the zombie channel. The code must not touch
a clone channel after it has masqueraded it. The clone channel
has become a zombie and is starting to hangup. (closes issue
ASTERISK-19985) Reported by: jamicque Patches:
jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: jamicque ........ Merged revisions 369327
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 369328 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-25 15:55 +0000 [r369304-369326] Mark Michelson <mmichelson@digium.com>
* include/asterisk/adsi.h, /, main/Makefile, res/res_adsi.c,
main/adsi.c (added), res/res_adsi.exports.in (removed): Multiple
revisions 369323-369324 ........ r369323 | mmichelson |
2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines Eliminate
embedding of res_adsi.so module. The way this is done is to stop
using the optional API. Instead, res_adsi.so, when loaded fills
in a table of function pointers. Review:
https://reviewboard.asterisk.org/r/1991 ........ r369324 |
mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2
lines Forgot to svn add this file in my last commit. ........
Merged revisions 369323-369324 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369325 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c: Be more consistent with the return code
for requests received from invalid domain. When Asterisk receives
an INVITE from an external domain when allowexternaldomains=no
send a 403 instead of a 404. This is consistent with Asterisk's
behavior when receiving a REGISTER in this situation. (Closes
issue ASTERISK-19601) Reported by Matthew Jordan Patches:
ASTERISK-19601-no401.patch uploaded by Mark Michelson (License
#5049) ........ Merged revisions 369302 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369303 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-23 00:33 +0000 [r369237-369296] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Fix F and F(x) action logic in Bridge
application.
* /, main/features.c: Fix Bridge application and AMI Bridge action
error handling. * Fix AMI Bridge action disconnecting the AMI
link on error. * Fix AMI Bridge action and Bridge application not
checking if their masquerades were successful. * Fix Bridge
application running the h-exten when it should not. * Made
do_bridge_masquerade() return if the masquerade was successful so
the Bridge application and AMI Bridge action could deal with it
correctly. * Made bridge_call_thread_launch() hangup the passed
in channels if the bridge_call_thread fails to start. Those
channels would have been orphaned. * Made builtin_atxfer() check
the success of the transfer masquerade setup. ........ Merged
revisions 369282 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369283 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_queue.c: Explicitly check caller hangup in app Queue
rather than a polluted res2 value. ........ Merged revisions
369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 369263 from
http://svn.asterisk.org/svn/asterisk/branches/10
* apps/app_queue.c: Fix F and F(x) action logic in Queue
application.
* apps/app_dial.c, /: Check if PBX was started and fix F and F(x)
action logic in Dial application. ........ Merged revisions
369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 369259 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/ccss.c: Check if PBX was started for generic CCSS recall.
........ Merged revisions 369238 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369239 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c: Change incorrect chan_sip zombie hangup
debug message. They are all zombies now. ........ Merged
revisions 369235 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369236 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-22 20:05 +0000 [r369217] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Don't crash on a guest directmedia call A
sip_pvt may not have relatedpeer set if a call doesn't match up
with a peer. If there is no relatedpeer, there is no direct media
ACL to apply, so just return that it is allowed. (closes issue
ASTERISK-20040) Reported by: Terry Wilson ........ Merged
revisions 369214 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369215 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-22 19:54 +0000 [r369184-369216] Kinsey Moore <kmoore@digium.com>
* channels/chan_dahdi.c: Fix wrong variable name in the R2
disconnect callback
* /, channels/chan_sip.c: Don't parse media stream state for SIP
video streams The sendonly/recvonly/sendrecv/inactive media
stream attributes were parsed for video, but nothing was ever
done with them. With this code removed, an UNSUPPORTED message is
produced when these attributes are used in conjunction with a
video stream which is the better behavior since they were never
really supported in the first place. ........ Merged revisions
369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 369206 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/chan_dahdi.c: Add HANGUPCAUSE hash implementation for
DAHDI MFC/R2 subtech This adds a minimal implementation of the
"Who Hung Up?" Asterisk 11 work to chan_dahdi.c for the MFC/R2
DAHDI subtech. Given the way that OpenR2 interfaces with
chan_dahdi, it is much harder to expose the type of protocol
information that is available in PRI, SS7, or other channel
technologies.
* channels/sig_analog.c, channels/sig_pri.c: Add HANGUPCAUSE hash
support for analog and PRI DAHDI subtechs This is part of the
DAHDI support for the Asterisk 11 "Who Hung Up?" project and
covers the implementation for the technologies implemented in
sig_analog.c and sig_pri.c. Tested on a local machine to verify
protocol and cause information is available. Review:
https://reviewboard.asterisk.org/r/1953/ (issue SWP-4222)
* channels/sig_ss7.c: Add "Who Hung Up?" implementation for DAHDI
SS7 subtechnology Testing was done on a local machine to verify
that protocol and cause information was being sent properly.
Review: https://reviewboard.asterisk.org/r/1955/ (issue SWP-4222)
2012-06-20 21:33 +0000 [r369166-369167] Richard Mudgett <rmudgett@digium.com>
* main/logger.c: Don't waste time initializing the whole
call_identifer_str[]. The array is either setup with a callid
string or only the first element needs to be initialized.
* channels/chan_misdn.c: Fix chan_misdn compile error.
2012-06-20 17:48 +0000 [r369148] Alexandr Anikin <may@telecom-service.ru>
* /, addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: fix
locking issue on empty callList (issue ASTERISK-19298) Reported
by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch ........
Merged revisions 369146 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369147 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-20 11:47 +0000 [r369142] Sean Bright <sean@malleable.com>
* apps/app_externalivr.c: Remove declaration of eivr_connect_socket
because it no longer exists.
2012-06-20 11:20 +0000 [r369141] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: use right definition for channel name
2012-06-20 03:18 +0000 [r369110-369126] Michael L. Young <elgueromexicano@gmail.com>
* main/manager.c, CHANGES: Add IPv6 Support To Manager This patch
adds IPv6 support to AMI. (Closes issue ASTERISK-19965) Reported
by: Michael L. Young Tested by: Michael L. Young Patches:
ami_ipv6_v3.diff uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/1968/
* main/netsock2.c, /, include/asterisk/netsock2.h: Fix NULL pointer
segfault in ast_sockaddr_parse() While working with
ast_parse_arg() to perform a validity check, a segfault occurred.
The segfault occurred due to passing a NULL pointer to
ast_sockaddr_parse() from ast_parse_arg(). According to the
documentation in config.h, "result pointer to the result. NULL is
valid here, and can be used to perform only the validity checks."
This patch fixes the segfault by checking for a NULL pointer.
This patch also adds documentation to netsock2.h about why it is
necessary to check for a NULL pointer. (Closes issue
ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael
L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded
by Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/1990/ ........ Merged
revisions 369108 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369109 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-19 23:36 +0000 [r369092] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c, /: check rtptimeouts in ooh323 channels as
per config file (rtp voice, video, udptl except rtcp) (closes
issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches:
19179-ooh323-ast10.patch ........ Merged revisions 369091 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-19 21:13 +0000 [r369086] Kinsey Moore <kmoore@digium.com>
* main/channel.c, channels/chan_dahdi.c, channels/chan_misdn.c,
main/rtp_engine.c, include/asterisk/channel.h,
channels/chan_iax2.c: Ensure that pvt cause information does not
break native bridging Channel drivers that allow native bridging
need to handle AST_CONTROL_PVT_CAUSE_CODE frames and previously
did not handle them properly, usually breaking out of the native
bridge. This change corrects that behavior and exposes the
available cause code information to the dialplan while native
bridges are in place. This required exposing the HANGUPCAUSE hash
setter outside of channel.c, so additional documentation has been
added.
2012-06-19 15:44 +0000 [r369068] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Fix request routing issue when
outboundproxy is used. Asterisk was incorrectly setting the
destination of CANCELs and ACKs for error responses to the URI of
the initial INVITE. This resulted in further requests, such as
INVITEs with authentication credentials, to be routed
incorrectly. Instead, when these CANCEL or ACKs are to be sent,
we should simply keep the destination the same as what it
previously was. There is no need to alter it any. (closes issue
ASTERISK-20008) Reported by Marcus Hunger Patches:
ASTERISK-20008.patch uploaded by Mark Michelson (license #5049)
........ Merged revisions 369066 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369067 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-18 22:56 +0000 [r369061] Kinsey Moore <kmoore@digium.com>
* main/features.c: Fix AST_CONTROL_PVT_CAUSE_CODE handling When the
IAX2 Who Hung Up? changes were added, they uncovered a bug in the
way AST_CONTROL_PVT_CAUSE_CODE was handled in
feature_request_and_dial(). This particular frame subtype was
being treated like more terminal control frames causing the
function to be exited prematurely.
2012-06-18 18:25 +0000 [r369057] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Fix monitoring calls put in a parking lot. *
Fix a regression that was introduced by -r366167 which
effectively disabled monitoring parked calls. (closes issue
ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett
........ Merged revisions 369043 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 369044 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-15 21:18 +0000 [r369034] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Various small chan_skinny fixes and
cleanup Added test to skinny_register to only allow device to
register against a device that is not already registered. Addback
l->device test for skinny_show_lines. Fixes segfault if a line is
configured but not configured to a device. Reverses part of
r368680. Removed redundant l->device tests in subsubstate and
dumpsub. l->device will always be valid if these routines are
called. Reverses 368948 - discussed with mjordan on irc. Some
indentation cleanup.
2012-06-15 17:13 +0000 [r369028] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c, channels/sip/include/sip.h: Allow chan_sip
to decline unwanted media streams This change replaces the static
array of four representable media streams with an AST_LIST so
that chan_sip can keep track of offered media streams. This
allows chan_sip to deal with offers containing multiple same-type
streams and many other situations without rejecting the SDP offer
in its entirety, yet still generating a valid response. This also
covers cases where Asterisk can not comprehend the offer if it is
in the correct format. Previously, chan_sip would reject SDP
offers or entirely ignore individual stream offers in an effort
to be more compatible which would often result in invalid SDP
responses. Review: https://reviewboard.asterisk.org/r/1988/
2012-06-15 16:30 +0000 [r369027] Jason Parker <jparker@digium.com>
* /, apps/app_voicemail.c: Fix voicemail API tests by using the
correct argument order for create/destroy. ........ Merged
revisions 369024 from
http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
........ Merged revisions 369026 from
http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
2012-06-15 16:20 +0000 [r369013] Kevin P. Fleming <kpfleming@digium.com>
* main/format.c, main/udptl.c, main/netsock2.c, main/autoservice.c,
main/rtp_engine.c, main/frame.c, main/security_events.c, /,
main/say.c, main/threadstorage.c, channels/console_video.c,
main/devicestate.c, main/astfd.c, main/taskprocessor.c,
main/format_pref.c, main/astobj2.c, main/indications.c,
main/config.c, main/loader.c, main/term.c,
apps/confbridge/conf_config_parser.c, main/cli.c,
channels/sig_analog.c, main/framehook.c, main/strcompat.c,
main/plc.c, main/fskmodem_int.c, main/syslog.c,
main/stdtime/localtime.c, main/bridging.c, main/db.c,
channels/sig_ss7.c, main/datastore.c, main/sched.c,
channels/sip/sdp_crypto.c, main/strings.c, main/pbx.c,
channels/vcodecs.c, channels/sip/security_events.c,
main/libasteriskssl.c, channels/iax2-provision.c,
pbx/dundi-parser.c, main/aoc.c, main/cel.c, utils/astdb2bdb.c,
channels/iax2-parser.c, main/chanvars.c, main/netsock.c,
build_tools/find_missing_support_level (added), main/data.c,
main/srv.c, channels/chan_misdn.c, main/privacy.c,
main/fixedjitterbuf.c, channels/sip/dialplan_functions.c,
main/test.c, main/audiohook.c, codecs/codec_dahdi.c, main/alaw.c,
main/asterisk.c, main/timing.c, main/global_datastores.c,
main/fskmodem_float.c, main/ccss.c,
channels/sip/reqresp_parser.c, main/xml.c,
channels/misdn/isdn_msg_parser.c, main/utils.c, main/autochan.c,
channels/misdn/isdn_lib.c, main/enum.c, main/presencestate.c,
main/fskmodem.c, channels/misdn_config.c, main/io.c,
main/channel.c, main/cdr.c, res/ael/pval.c, main/ulaw.c,
main/dial.c, main/format_cap.c, main/tdd.c,
channels/console_gui.c, main/heap.c, channels/misdn/ie.c,
main/logger.c, main/app.c, channels/console_board.c,
main/image.c, main/message.c, main/dns.c, main/lock.c,
main/stun.c, channels/sip/srtp.c, main/dnsmgr.c,
main/slinfactory.c, main/channel_internal_api.c,
main/translate.c, main/jitterbuf.c, main/acl.c,
utils/astdb2sqlite3.c, channels/sip/utils.c, channels/sig_pri.c,
apps/app_system.c, funcs/func_realtime.c, main/tcptls.c,
main/hashtab.c, funcs/func_presencestate.c,
apps/app_celgenuserevent.c, main/abstract_jb.c, main/callerid.c,
main/file.c, main/config_options.c, res/snmp/agent.c,
main/astmm.c, main/event.c, channels/misdn/portinfo.c,
channels/sip/config_parser.c, channels/vgrabbers.c, main/dsp.c,
main/xmldoc.c: Multiple revisions 369001-369002 ........ r369001
| kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11
lines Add support-level indications to many more source files.
Since we now have tools that scan through the source tree looking
for files with specific support levels, we need to ensure that
every file that is a component of a 'core' or 'extended' module
(or the main Asterisk binary) is explicitly marked with its
support level. This patch adds support-level indications to many
more source files in tree, but avoids adding them to third-party
libraries that are included in the tree and to source files that
don't end up involved in Asterisk itself. ........ r369002 |
kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3
lines Add a script to enable finding source files without
support-levels defined. ........ Merged revisions 369001-369002
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 369005 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-15 16:17 +0000 [r369007] Kinsey Moore <kmoore@digium.com>
* main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h: Add
HANGUPCAUSE hash support to IAX2 Continuing with the Who Hung Up?
project for Asterisk 11, this adds support to IAX2 for the
HANGUPCAUSE hash. Additionally, this breaks out some
functionality in frame.c for getting information about frame
types and subclasses. Review:
https://reviewboard.asterisk.org/r/1941/ (issue SWP-4222)
2012-06-15 15:33 +0000 [r369000] Jason Parker <jparker@digium.com>
* /, apps/app_voicemail.exports.in: Remove some symbol exports that
got missed in the removal of global symbols. (issue AST-807)
(issue AST-901) (issue AST-908) ........ Merged revisions 368998
from
http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
........ Merged revisions 368999 from
http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
2012-06-15 00:55 +0000 [r368972-368991] Richard Mudgett <rmudgett@digium.com>
* /: Remove remaining properties mmichelson left laying around from
phones branch merge.
* apps/app_dial.c, main/channel.c, include/asterisk/app.h,
main/ccss.c, main/app.c, apps/app_followme.c, apps/app_queue.c,
apps/app_stack.c: Allow non-normal execution routines to be able
to run on hungup channels. * Make non-normal dialplan execution
routines be able to run on a hung up channel. This is preparation
work for hangup handler routines. * Fixed ability to support
relative non-normal dialplan execution routines. (i.e., The
context and exten are optional for the specified dialplan
location.) Predial routines are the only non-normal routines that
it makes sense to optionally omit the context and exten. Setting
a hangup handler also needs this ability. * Fix Return
application being able to restore a dialplan location exactly.
Channels without a PBX may not have context or exten set. * Fixes
non-normal execution routines like connected line interception
and predial leaving the dialplan execution stack unbalanced.
Errors like missing Return statements, popping too many stack
frames using StackPop, or an application returning non-zero could
leave the dialplan stack unbalanced. * Fixed the AGI gosub
application so it cleans up the dialplan execution stack and
handles the autoloop priority increments correctly. * Eliminated
the need for the gosub_virtual_context return location. Review:
https://reviewboard.asterisk.org/r/1984/
* main/pbx.c: Make the Hangup application set a softhangup flag.
The Hangup application used to just return -1 to cause normal
dialplan execution to hangup a channel. For the non-normal
execution routines like predial and connected-line interception
routines, the hangup request would exit the routine early but
otherwise be ignored. * Made the Hangup application not allow
setting a cause code of zero. A zero cause code is not defined.
* include/asterisk/app.h: Move vm defines to group them better.
2012-06-14 19:40 +0000 [r368966] Jason Parker <jparker@digium.com>
* include/asterisk/app.h, /, tests/test_voicemail_api.c,
main/app.c, include/asterisk/app_voicemail.h (removed),
apps/app_voicemail.c: Multiple revisions 368963,368965 ........
r368963 | qwell | 2012-06-14 13:47:03 -0500 (Thu, 14 Jun 2012) |
14 lines Remove global symbol requirement from app_voicemail.
This uses the existing "function installation" stuff that already
existed for other functions, like getting message counts. (closes
issue AST-807) (issue AST-901) (issue AST-908) Review:
https://reviewboard.asterisk.org/r/1965/ ........ Merged
revisions 368962 from
http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
........ r368965 | qwell | 2012-06-14 14:04:57 -0500 (Thu, 14 Jun
2012) | 11 lines These functions that were moved need to be
static. Also wrap test functions in a #ifdef. (issue AST-807)
(issue AST-901) (issue AST-908) ........ Merged revisions 368964
from
http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11
........ Merged revisions 368963,368965 from
http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones
2012-06-14 17:34 +0000 [r368948] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_skinny.c: AST-2012-009: Fix crash in chan_skinny
due to Key Pad Button Message handling AST-2012-008 (r367844)
fixed a denial of service attack exploitable in the Skinny
channel driver that occurred when certain messages are sent after
a previously registered station sends an Off Hook message.
Unresolved in that patch is an issue in the Asterisk 10 releases,
wherein, if a Station Key Pad Button Message is processed after
an Off Hook message, the channel driver will inappropriately
dereference a NULL pointer. This patch fixes those places where
the message handling or the channel callback functions would
attempt to dereference the line's pointer to the device. (issue
ASTERISK-19905) Reported by: Christoph Hebeisen Tested by:
mjordan, Christoph Hebeisen Patches: AST-2012-009-10.diff
uploaded by mjordan (license 6283) ........ Merged revisions
368947 from http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-14 15:28 +0000 [r368929] Mark Michelson <mmichelson@digium.com>
* /, main/Makefile: Revert Makefile change to remove embedding
res_adsi.so The change has resulted in a linking error for
certain versions of GCC. This is much worse than the original
issue, so for now, temporarily revert the change. A more thorough
change will be sought out. ........ Merged revisions 368927 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368928 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-14 13:41 +0000 [r368920-368921] Terry Wilson <twilson@digium.com>
* include/asterisk/config_options.h, main/config_options.c: Add a
post_apply callback to the Config Options API This adds a
callback that only fires when changes have been successfully
applied via the Config Options API. Review:
https://reviewboard.asterisk.org/r/1980/
* include/asterisk/config_options.h, main/config_options.c: Add
filename alias support to the Config Options API This adds the
ability to handle a single filename alias for a config file. This
is useful if a config filename has changed, but the old filename
should be supported for backwards compatibility. Review:
https://reviewboard.asterisk.org/r/1981/
2012-06-13 21:17 +0000 [r368900] Mark Michelson <mmichelson@digium.com>
* /, funcs/func_volume.c: Fix a deadlock that occurs when
func_volume is used on a local channel. This was discovered by
trying to perform a call forward to an extension that makes use
of func_volume. When the local channel is optimized away, the
datastore on the local;2 channel would have its audiohook
destroyed rather than detaching the audiohook from the channel
and then destroying it. With this patch, func_volume's datastore
destructor takes the proper route of detaching the audiohook and
then destroying it. (closes issue ASTERISK-19611) reported by
Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark
Michelson (license #5049) ........ Merged revisions 368898 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368899 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-13 20:28 +0000 [r368896] Matthew Jordan <mjordan@digium.com>
* res/res_smdi.c, /, res/res_adsi.c: Mark res_smdi/res_adsi as
'core' supported modules Recently, various issues surrounding
weak symbols have caused problems with modules that rely on that
feature to be enabled in menuselect. This includes app_voicemail
and chan_dahdi, as they both rely upon res_smdi and res_adsi,
which, in certain circumstances, may not be enabled by default in
menuselect. Because res_smdi/res_adsi are dependencies for
chan_dahdi/app_voicemail, this patch marks both as 'core'
supported modules. This will allow both app_voicemail and
chan_dahdi to be enabled as well, regardless of whether or not
that system supports weak symbols. (issue AST-900) Reported by:
Thomas Arimont (issue AST-885) Reported by: Denis Alberto
Martinez ........ Merged revisions 368894 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368895 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-13 19:51 +0000 [r368886] Mark Michelson <mmichelson@digium.com>
* /, main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+
the result is that Asterisk has a phantom module loaded at
startup, claiming to be res_adsi. (closes issue ASTERISK-19920)
reported by Leif Madsen ........ Merged revisions 368873 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368885 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-13 14:55 +0000 [r368832-368855] Matthew Jordan <mjordan@digium.com>
* Makefile: Replace MODULES_DIR with ASTMODDIR in Makefile's
INSTALLDIRS Post Asterisk 10, the MODULES_DIR variable no longer
exists, and was replaced with ASTMODDIR.
* Makefile, /: Do not install empty directories; add ASTLIBDIR
r368830 modified the installation script to only create a
directory if that directory does not exist. If some directory
variable was empty, it would attempt to create the empty
location. It also failed to create the ASTLIBDIR directory. This
patch fixes it such that the correct directories are made and
only created if a value specifying them actually exists. ........
Merged revisions 368852 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368853 from
http://svn.asterisk.org/svn/asterisk/branches/10
* Makefile, /: Do not perform install on existing directories If a
directory already exists, performing a 'make install' will remove
the permissions associated with the current directory and replace
them with the permissions of the user executing the install. This
patch changes this behavior to only perform an install on the
directory if the directory does not exist. Thus, if a user later
changes the permissions on that directory, those permissions will
be preserved in subsequent installs. Review:
https://reviewboard.asterisk.org/r/1986 Review:
https://reviewboard.asterisk.org/r/1864 (closes issue
ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger,
Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded
by mjordan) ........ Merged revisions 368830 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368831 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-12 15:46 +0000 [r368809] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Set the Caller ID "tag" on peers even if
remote party information is present. On incoming calls, we were
setting the cid_tag on the dialog only if there was no remote
party information (Remote-Party-ID or P-Asserted-Identity)
present. The Caller ID tag is an invented parameter, though, and
should be set no matter the circumstance. (closes issue
ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884)
Reported by Trey Blancher ........ Merged revisions 368807 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368808 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-12 14:09 +0000 [r368793-368794] Matthew Jordan <mjordan@digium.com>
* /: Update merge property information
* channels/chan_sip.c: Fix deadlock in SIP transfers that involve a
REFER request In r367163, "send to voicemail" functionality was
added to the SIP channel driver. This required updating the party
redirecting information for the channel based on the headers
provided in the REFER request. When the redirecting party
information is updated on the channel, a call to
ast_indicate_data occurs. Because handle_request_refer still had
the sip_pvt locked, a deadlock could occur between the pbx_thread
and the do_monitor thread servicing the REFER request. This patch
preserves the proper locking order between the channel and the
sip_pvt by ensuring that the sip_pvt is unlocked prior to
updating the party redirecting information on the channel.
(closes issue AST-903) Reported by: Matt Jordan patches:
jira_ast_903_trunk.patch by rmudgett (license 5621)
2012-06-12 04:03 +0000 [r368784] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c, UPGRADE.txt: Parse ANI2 information from SIP
From header parameters ANI2 information is now parsed out of SIP
From headers when present in the oli, isup-oli, and ss7-oli
parameters and is available via the CALLERID(ani2) dialplan
function. (closes issue ASTERISK-19912) Patch-by: Rob Gagnon
Review: https://reviewboard.asterisk.org/r/1947/
2012-06-11 17:34 +0000 [r368772] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c, /,
channels/chan_sip.c, include/asterisk/channel.h,
channels/chan_iax2.c: Fix deadlock potential with
ast_set_hangupsource() calls. Calling ast_set_hangupsource() with
the channel lock held can result in a deadlock because the
function also locks the bridged channel. (issue ASTERISK-19537)
(closes issue AST-891) Reported by: Guenther Kelleter Tested by:
Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec
Davis ........ Merged revisions 368759 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368760 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-11 15:23 +0000 [r368722-368751] Kinsey Moore <kmoore@digium.com>
* channels/sip/sdp_crypto.c, /, channels/chan_sip.c, main/say.c,
res/res_fax.c, channels/sip/reqresp_parser.c, apps/app_queue.c,
main/loader.c, channels/chan_dahdi.c, res/res_config_odbc.c,
channels/sip/dialplan_functions.c, apps/app_directory.c,
pbx/pbx_config.c, res/res_odbc.c, res/res_speech.c,
apps/app_voicemail.c: Fix coverity UNUSED_VALUE findings in core
support level files Most of these were just saving returned
values without using them and in some cases the variable being
saved to could be removed as well. (issue ASTERISK-19672)
........ Merged revisions 368738 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368739 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /: Recorded merge of revisions 368721 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Fix
compilation in dev-mode Backport a compilation fix in md5.c from
trunk that only showed up in dev-mode under certain compiler
versions. ........ Merged revisions 368719 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
2012-06-08 21:08 +0000 [r368712-368714] Richard Mudgett <rmudgett@digium.com>
* main/manager.c, main/utils.c, include/asterisk/strings.h: Fix
error paths in action_hangup() for AMI Hangup action. * Check
allocation function return values for failure. Crashing is bad. *
Tweak ast_regex_string_to_regex_pattern() parameters for proper
ast_str usage.
* main/channel.c, include/asterisk/channel.h: Tweak
ast_channel_softhangup_withcause_locked() to take a typed
parameter.
2012-06-08 08:32 +0000 [r368688] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c: Fix MWI update so LED display correct
voicemail state after phone usage. Also fixes few warnings.
(closes issue #19675) Reported by: dbohling Patches: fixmwi.patch
uploaded by dbohling (license 6378)
2012-06-07 21:44 +0000 [r368680-368681] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Skinny cleanup (mwi_event_cb). Original
was testing for d->session, setting and testing again (all
nested). Removed duplicate testing and restructured function to
test/return and then the main code.
* channels/chan_skinny.c: Skinny cleanup. Removed d->registered
which was mirroring d->session. Changed relevant references to
use d->session instead. Moved setting and unsetting of l->device
from session register to device configuration. As such, l->device
will always be valid unless it is has not been configured to a
device. Revised various test where checking if a device is
registered to use l->device->session.
2012-06-07 20:39 +0000 [r368674-368675] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c: Fix app_queue debug message use of args.options
after the string has been parsed.
* apps/app_queue.c: Fix inverted test in app_queue for ringinuse.
Regression from -r367080 ringinuse commit. (issue ASTERISK-19536)
2012-06-07 20:32 +0000 [r368673] Terry Wilson <twilson@digium.com>
* main/udptl.c, include/asterisk/config_options.h, apps/app_skel.c,
main/config_options.c, tests/test_config.c: Fix reloading an
unchanged file with the Config Options API Adding multiple file
support broke reloading an unchanged file. This adds an enum for
return values for the aco_process_* functions and ensures that
the config is not applied if res is not ACO_PROCESS_OK. Review:
https://reviewboard.asterisk.org/r/1979/
2012-06-07 20:00 +0000 [r368668] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* formats/format_ogg_vorbis.c: Fix a typo in format_ogg_vorbis.c:
suport Review: https://reviewboard.asterisk.org/r/1970/
2012-06-07 15:43 +0000 [r368663] Terry Wilson <twilson@digium.com>
* include/asterisk/config_options.h, main/config_options.c,
tests/test_config.c: Add default handler documentation and
standardize acl handler Added documentation describing what flags
and arguments to pass to aco_option_register for default option
types. Also changed the ACL handler to use the flags parameter to
differentiate between "permit" and "deny" instead of adding an
additional vararg parameter. Review:
https://reviewboard.asterisk.org/r/1969/
2012-06-06 21:34 +0000 [r368646] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /: Fix POTS flash
hook to orignate a second call deadlock. A deadlock can occur
when a POTS phone tries to flash hook to originate a second call
for 3-way or transfer. If another process is scanning the
channels container when the POTS line flash hooks then a deadlock
will occur. * Release the channel and private locks when creating
a new channel as a result of a flash hook. (closes issue
ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett
........ Merged revisions 368644 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368645 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-06 19:25 +0000 [r368637] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Fix a specific scenario where ACKs are
not matched. If a dialog-starting INVITE contains a to-tag, then
Asterisk will respond with a 481. In this case, the resulting
incoming ACK would not be matched, so Asterisk would continue
retransmitting the 481 until the transaction times out. There
were two issues. Asterisk, upon creating a sip_pvt would generate
a local tag. However, when the time came to transmit the 481,
since there was a to-tag in the INVITE, Asterisk would place this
original to-tag in the 481 response. When the ACK came in,
Asterisk would attempt to match the to-tag in the ACK to the
generated local tag. Unfortunately, Asterisk never actually
transmitted a response with the generated local tag, so the
to-tag in the ACK would not match. The other problem was that
when the 481 was sent, nothing was set on the sip_pvt to indicate
what CSeq is expected in the ACK. To fix the first problem, we
zero out the to-tag seen in the incoming INVITE. This way,
Asterisk, when time to send a response, will send its generated
local tag instead. To fix the second problem, we set the
sip_pvt's pendinginvite to the CSeq of the INVITE when we send a
481. (closes issue ASTERISK-19892) Reported by Mark Michelson
........ Merged revisions 368625 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368629 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-06 17:22 +0000 [r368606] Matthew Jordan <mjordan@digium.com>
* /, build_tools/make_version: Add feature modifier to versions
produced from branches Certain branches, such as Certified
Asterisk, may have a modifier added to them that specifies the
features available in that branch. For branches, this modifier is
expected to be reflected in the location of the branch in
subversion. For example, a subversion of URL of
/certified/branches/1.8.11 would have a feature modifier of
'certified'. This is slightly different then how features are
determined for tags, where the feature is part of the actual tag
name, e.g., "10.5.0-digiumphones". In keeping with the
nomenclature used for tags, the feature specifier for branches is
translated and placed after the revision numbers. For the example
given previously, this would result in a branch version of
"Asterisk SVN-branch-1.8.11-cert-rXXXXXX". ........ Merged
revisions 368604 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368605 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-06 16:11 +0000 [r368588] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Ensure overlapping hold flags do not
conflict When changing between different modes of hold, the flags
were not being cleared out properly causing a failure to change
hold states. (closes issue ASTERISK-19919) Patch-by: Morten
Tryfoss Reported-by: Morten Tryfoss ........ Merged revisions
368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 368587 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-06 01:11 +0000 [r368566-368569] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Fix parked call performing a DTMF blind
transfer after being retrieved. When a parked call was retrieved
from the parking lot, it could not do a blind transfer because it
caused the involved calls to be hung up unconditionally. * Made
the ParkedCall application return the ast_bridge_call() return
value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc
........ Merged revisions 368567 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368568 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/features.c: Make builtin_blindtransfer() fully use
ast_async_goto() abilities.
2012-06-05 16:25 +0000 [r368550] Jonathan Rose <jrose@digium.com>
* CHANGES: Merge 'core' and 'core changes' sections in CHANGES
file.
2012-06-05 15:28 +0000 [r368519-368537] Kinsey Moore <kmoore@digium.com>
* /: Recorded merge of revisions 368536 from
http://svn.asterisk.org/svn/asterisk/branches/10 ........ Resolve
some build warnings My newly upgraded compiler caught these
usages of uninitialized values. They weren't actually used.
........ Merged revisions 368533 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
* /, apps/app_voicemail.c: Ensure that pages and emails are sent
using RFC822-compliant date format When localization was added to
app_voicemail, these headers were altered when they should have
remained in en_US format for RFC compliance. This reverts the
changes to those two lines. (closes issue ASTERISK-19876)
........ Merged revisions 368520 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368524 from
http://svn.asterisk.org/svn/asterisk/branches/10
* apps/app_dial.c, channels/chan_unistim.c, channels/chan_local.c,
channels/chan_sip.c, main/channel_internal_api.c,
main/features.c, include/asterisk/channel.h, apps/app_queue.c:
Convert AST_FLAG_ANSWERED_ELSEWHERE usage to
AST_CAUSE_ANSWERED_ELSEWHERE This was essentially duplicated
functionality where normal channels used
AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used
AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts
that usage into AST_CAUSE_ANSWERED_ELSEWHER usage. Review:
https://reviewboard.asterisk.org/r/1944 (closes issue
ASTERISK-19865) Patch-by: Birger Harzenetter
2012-06-04 22:12 +0000 [r368500] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Relay proper SIP responses on calling
side. Revision 351130 broke corect HANGUPCAUSE setting for the
404 case in chan_sip. Other cases were also potentially broken.
This patch fixes the relaying of causes to be what they used to
be. (closes issue ASTERISK-19914) Reported by Pavel Troller
Tested by Walter Doekes (via a reviewboard test to be committed
later) Patches: chan_sip.diff uploaded by Pavel Troller (license
#6302) ........ Merged revisions 368498 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368499 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-04 21:18 +0000 [r368472] Richard Mudgett <rmudgett@digium.com>
* /, UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue
ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call
........ Merged revisions 368469 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368470 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-04 20:53 +0000 [r368435-368467] Mark Michelson <mmichelson@digium.com>
* contrib/editors/asterisk.vim: Also have vim syntax-highlight
type=network.
* contrib/editors/asterisk.vim: Add vim syntax highlighting for
type=line, type=phone, and type=application. (closes issue
ASTERISK-19800) Reported by: Billy Chia Patches:
asterisk.vim.patch uploaded by Billy Chia (license #6381)
* main/channel.c, apps/app_mixmonitor.c: Remove some extra
debugging I forgot to remove in the merge of Digium phone
support.
* /: Remove automerge properties.
* /, contrib/realtime/mysql/voicemail_messages.sql,
main/presencestate.c (added), main/config.c, main/channel.c,
include/asterisk/callerid.h, include/asterisk/file.h,
main/manager.c, channels/chan_skinny.c,
include/asterisk/event_defs.h, include/asterisk/sip_api.h
(added), tests/test_voicemail_api.c (added), main/features.c,
apps/app_voicemail.exports.in, main/app.c, main/message.c,
channels/sip/include/sip.h, main/pbx.c, channels/chan_sip.c,
include/asterisk/presencestate.h (added),
include/asterisk/config.h, include/asterisk/app_voicemail.h
(added), configs/manager.conf.sample, apps/app_queue.c,
include/asterisk/manager.h, include/asterisk/app.h,
funcs/func_presencestate.c (added), include/asterisk/message.h,
main/file.c, main/callerid.c, main/event.c,
include/asterisk/pbx.h, tests/test_config.c,
channels/chan_sip.exports.in (added), apps/app_mixmonitor.c,
main/asterisk.c, apps/app_voicemail.c: Merge changes dealing with
support for Digium phones. Presence support has been added. This
is accomplished by allowing for presence hints in addition to
device state hints. A dialplan function called PRESENCE_STATE has
been added to allow for setting and reading presence. Presence
can be transmitted to Digium phones using custom XML elements in
a PIDF presence document. Voicemail has new APIs that allow for
moving, removing, forwarding, and playing messages. Messages have
had a new unique message ID added to them so that the APIs will
work reliably. The state of a voicemail mailbox can be obtained
using an API that allows one to get a snapshot of the mailbox. A
voicemail Dialplan App called VoiceMailPlayMsg has been added to
be able to play back a specific message. Configuration hooks have
been added. Configuration hooks allow for a piece of code to be
executed when a specific configuration file is loaded by a
specific module. This is useful for modules that are dependent on
the configuration of other modules. chan_sip now has a public
method that allows for a custom SIP INFO request to be sent
mid-dialog. Digium phones use this in order to display progress
bars when files are played. Messaging support has been expanded a
bit. The main visible difference is the addition of an AMI action
MessageSend. Finally, a ParkingLots manager action has been added
in order to get a list of parking lots.
2012-06-04 19:46 +0000 [r368421] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /: Fix potential deadlock between masquerade and
chan_local. * Restructure ast_do_masquerade() to not hold channel
locks while it calls ast_indicate(). * Simplify many calls to
ast_do_masquerade() since it will never return a failure now. If
it does fail internally because a channel driver callback
operation failed, the only thing ast_do_masquerade() can do is
generate a warning message about strange things may happen and
press on. * Fixed the call to ast_bridged_channel() in
ast_do_masquerade(). This change fixes half of the deadlock
reported in ASTERISK-19801 between masquerades and chan_iax.
(closes issue ASTERISK-19537) Reported by: rmudgett Tested by:
rmudgett Review: https://reviewboard.asterisk.org/r/1915/
........ Merged revisions 368405 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368407 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-02 21:13 +0000 [r368359] Joshua Colp <jcolp@digium.com>
* include/asterisk/utils.h, res/res_http_websocket.exports.in
(added), include/asterisk/http_websocket.h (added), main/utils.c,
res/res_http_websocket.c (added): Add res_http_websocket module
which implements the WebSocket protocol according to RFC 6455.
Review: https://reviewboard.asterisk.org/r/1952/
2012-06-01 23:53 +0000 [r368311] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_stack.c: Fix deadlock when Gosub used with alternate
dialplan switches. Attempting to remove a channel from
autoservice with the channel lock held will result in deadlock. *
Restructured gosub_exec() to not call ast_parseable_goto() and
ast_exists_extension() with the channel lock held. (closes issue
ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett
........ Merged revisions 368308 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368310 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-01 20:42 +0000 [r368268-368269] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_sip.c: Improve SDP offer/answer RFC compliance
Asterisk should not accept SDP offers that contain unknown RTP
profiles (for audio/video streams) or unknown top-level media
types. When it does, it answers with an SDP that does not match
the offer properly, and this will nearly always result in a
broken call. This patch causes such offers to be rejected.
Review: https://reviewboard.asterisk.org/r/1811/
* /, channels/chan_sip.c: Improve SDP parsing warning messages *
'Unsupported media type' is only reported when that is in fact
the case, not when a supported media type is included in an 'm'
line that has an invalid format. * All warning messages related
to parsing 'm' lines now include the 'm' line contents. * (minor
bugfix) newline added to port-number-zero warning messages. *
Warning messages improved to use RFC-specified terminology for
various items. * Warnings for offers that include more than one
port for a single media type now include the media type. Review:
https://reviewboard.asterisk.org/r/1811/ ........ Merged
revisions 368218 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368267 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-06-01 18:20 +0000 [r368181-368221] Terry Wilson <twilson@digium.com>
* configs/config_test.conf.sample (added): Add missing config for
config API test
* main/udptl.c, include/asterisk/utils.h,
include/asterisk/astobj2.h, configure.ac,
include/asterisk/config.h, main/astobj2.c, main/config.c,
Makefile, include/asterisk/config_options.h (added), configure,
main/asterisk.exports.in, apps/app_skel.c, main/config_options.c
(added), tests/test_config.c, makeopts.in,
configs/app_skel.conf.sample (added),
include/asterisk/stringfields.h: Add new config-parsing framework
This framework adds a way to register the various options in a
config file with Asterisk and to handle loading and reloading of
that config in a consistent and atomic manner. Review:
https://reviewboard.asterisk.org/r/1873/
2012-06-01 13:04 +0000 [r368143] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample,
channels/sip/include/sip.h: Help mitigate potential reinvite
glare scenarios. When Asterisk servers are set up back-to-back,
and direct media is to be used betweeen endpoints, it is fairly
common for the two Asterisk servers to send direct media
reinvites to each other simultaneously. This results in 491s and
ACKs being exchanged between the servers. While the media
eventually gets set up properly, the problem is that there can be
a noticeable delay for the streams to stabilize. This patch adds
a new directmedia option called "outgoing". With this set, an
immediate direct media reinvite will only be sent if the call
direction is outgoing. For incoming dialogs, an immediate direct
media reinvite will not be sent, but further "reactionary" direct
media reinvites may be sent. Review:
https://reviewboard.asterisk.org/r/1954
2012-06-01 03:30 +0000 [r368094] Michael L. Young <elgueromexicano@gmail.com>
* /, funcs/func_channel.c: Add documentation to function CHANNEL
for options echocan_mode and buffers The ability to set
"echocan_mode" and "buffers" through the dialplan was added to
chan_dahdi some time ago. This patch adds some documentation to
func_channel. (Closes issue ASTERISK-19911) Reported by: Dale
Noll Tested by: Michael L. Young Patches:
asterisk-19911-branch18.diff uploaded by Michael L. Young
(license 5026) Review: https://reviewboard.asterisk.org/r/1949/
........ Merged revisions 368092 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368093 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-31 18:39 +0000 [r368052] Richard Mudgett <rmudgett@digium.com>
* res/ael/pval.c, main/tcptls.c, main/manager.c,
res/res_config_odbc.c, /, channels/chan_sip.c,
channels/chan_agent.c, funcs/func_math.c, main/features.c,
apps/app_queue.c, channels/chan_iax2.c, pbx/pbx_config.c:
Coverity Report: Fix issues for error type REVERSE_INULL (core
modules) * Fixes findings: 0-2,5,7-15,24-26,28-31 (issue
ASTERISK-19648) Reported by: Matt Jordan ........ Merged
revisions 368039 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 368042 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-30 18:08 +0000 [r367908-367982] Richard Mudgett <rmudgett@digium.com>
* /: Use the DEADLOCK_AVOIDANCE() macro instead. (issue
ASTERISK-19854) ........ Merged revisions 367980 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367981 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/sig_pri.c, channels/sig_ss7.c: Fix deadlock when
executing CLI "pri show channels" and "ss7 show channels"
commands. * Fix sig_pri_lock_owner() to avoid deadlock properly.
* Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid
deadlock properly. * Code ss7_grab() better. (closes issue
ASTERISK-19854) Reported by: Jaxon Patches:
jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded
by rmudgett (Modified to do the same thing to sig_ss7) Tested by:
Jaxon ........ Merged revisions 367976 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367978 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_meetme.c: Coverity Report: Fix issues for error type
REVERSE_INULL (deprecated modules) * Fix only issue pointed out
by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user().
* Change use of %i to %d in sscanf() in find_user(). The use of
%i gives unexpected parsing because it can accept hex, octal, and
decimal integer formats. * Changed other uses of %i in
app_meetme() to use %d for consistency. (issue ASTERISK-19648)
Reported by: Matt Jordan ........ Merged revisions 367906 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367907 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-29 18:40 +0000 [r367845] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_skinny.c: AST-2012-008: Fix remote crash
vulnerability in chan_skinny When a skinny session is
unregistered, the corresponding device pointer is set to NULL in
the channel private data. If the client was not in the on-hook
state at the time the connection was closed, the device pointer
can later be dereferened if a message or channel event attempts
to use a line's pointer to said device. The patches prevent this
from occurring by checking the line's pointer in message handlers
and channel callbacks that can fire after an unregistration
attempt. (closes issue ASTERISK-19905) Reported by: Christoph
Hebeisen Tested by: mjordan, Damien Wedhorn Patches:
AST-2012-008-1.8.diff uploaded by mjordan (license 6283)
AST-2012-008-10.diff uploaded by mjordan (licesen 6283) ........
Merged revisions 367844 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-25 16:33 +0000 [r367783] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_iax2.c: AST-2012-007: Fix IAX receiving HOLD
without suggested MOH class crash. * Made schedule_delivery() set
the received frame f->data.ptr to NULL if the datalen is zero. *
Fix queue_signalling() memcpy() size error. * Made
queue_signalling() not use C++ keyword variable names. (closes
issue ASTERISK-19597) Reported by: mgrobecker Patches:
jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: rmudgett, Michael L. Young ........ Merged
revisions 367781 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367782 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-25 02:31 +0000 [r367732] Michael L. Young <elgueromexicano@gmail.com>
* /, channels/chan_sip.c: Fix pvt_sip for inbound call to use
peer's allowtransfer setting The pvt_sip allowtransfer was not
being set to that of the peer's setting. Therefore, the global
allowtransfer setting was being used instead which would lead to
calls not being transfered if the global setting was set to 'no'
despite the setting on the peer being 'yes' and vice versa, calls
would be allowed to transfer even if the peer's setting was 'no'
but the global setting was 'yes'. (Closes issue ASTERISK-19856)
Reported by: Jacek Tested by: Michael L. Young, Jacek Patches:
issue-asterisk-19856-branch10-v3.diff uploaded by Michael L.
Young (license 5026) Review:
https://reviewboard.asterisk.org/r/1923/ ........ Merged
revisions 367730 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367731 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-24 23:52 +0000 [r367693] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, /, apps/app_queue.c: Fix Dial I option ignored
if dial forked and one fork redirects. The Dial and Queue I
option is intended to block connected line updates and
redirecting updates. However, it is a feature that when a call is
locally redirected, the I option is disabled if the redirected
call runs as a local channel so the administrator can have an
opportunity to setup new connected line information.
Unfortunately, the Dial and Queue I option is disabled for *all*
forked calls if one of those calls is redirected. * Make the Dial
and Queue I option apply to each outgoing call leg independently.
Now if one outgoing call leg is locally redirected, the other
outgoing calls are not affected. * Made Dial not pass any
redirecting updates when forking calls. Redirecting updates do
not make sense for this scenario. * Made Queue not pass any
redirecting updates when using the ringall strategy. Redirecting
updates do not make sense for this scenario. * Fixed deadlock
potential with chan_local when Dial and Queue send redirecting
updates for a local redirect. * Converted the Queue stillgoing
flag to a boolean bitfield. (closes issue ASTERISK-19511)
Reported by: rmudgett Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1920/ ........ Merged
revisions 367678 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367679 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-24 18:56 +0000 [r367640] Jonathan Rose <jrose@digium.com>
* main/rtp_engine.c, channels/chan_sip.c,
include/asterisk/rtp_engine.h: chan_sip: fix problem
directmediapermit/deny uses the wrong address When remotely
bridging calls with directmedia, Asterisk would check the address
of the peers/users holding directmedia ACLs (set via
directmediapermit/directmediadeny) instead of the bridged peer.
This is similar to r366547, but trunk specific and involves
changes to the rtpengine instead of just chan_sip. (closes issue
AST-876) review: https://reviewboard.asterisk.org/r/1924/
2012-05-24 13:33 +0000 [r367563] Matthew Jordan <mjordan@digium.com>
* /, apps/app_confbridge.c: Fix crash in ConfBridge when user
announcement is played for more than 2 users A patch introduced
in r354938 made it so that ConfBridge would not attempt to play
sound files if those files did not exist. Unfortunately,
ConfBridge uses the same underlying function, play_sound_helper,
to playback both sound files and numbers to callers. When a
number is being played back, the name of the sound file is
expected to be NULL. This NULL value was passed into a function
that tested for the existance of a sound file and is not tolerant
to NULL file names, causing a crash. This patch fixes the
behavior, such that if a sound file does not exist we do not
attempt to play it, but we only attempt that check if the a sound
file was specified in the first place. If a sound file was not
specified, we use the 'play number' logic in the helper function.
(closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested
by: Florian Gilcher patches: asterisk-19899.diff uploaded by
mjordan (license 6283) ........ Merged revisions 367562 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-24 00:36 +0000 [r367477-367520] Richard Mudgett <rmudgett@digium.com>
* channels/iax2-parser.c: Made use IAX frame cache only for
cacheable frame types.
* main/pbx.c, /: Fix WaitExten(x,m(musicclass)) string termination.
The AST_CONTROL_HOLD MOH class from the WaitExten application can
now be queued onto a channel, passed over local channels with the
/m option, and passed over IAX channels. ........ Merged
revisions 367469 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367470 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-23 20:39 +0000 [r367419] Jonathan Rose <jrose@digium.com>
* main/pbx.c: logger: Fix a potential callid reference leak
discovered in development Uncovered a nasty reference leak while
I was writing some changes to chan_dahdi/sig_analog. Slapped
myself around a bit after seeing that I performed the unchecked
return causing this problem.
2012-05-23 20:30 +0000 [r367418] Mark Michelson <mmichelson@digium.com>
* main/tcptls.c, /: Only call SSL_CTX_free if DO_SSL is defined.
Thanks to Paul Belanger for pointing out this error. ........
Merged revisions 367416 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367417 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-23 13:46 +0000 [r367376] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c, channels/sip/include/sip.h: Re-add
LastMsgsSent value for SIP peers Previously, MWI logic utilized a
counter called 'lastmsgssent' to know whether or not MWI NOTIFY
requests had been sent to a specific peer. When MWI notifications
were changed to use the internal event framework, this value was
no longer needed for its original purpose. Hence, it was no
longer updated with the new/old message counts for a peer. The
value was previously removed for Asterisk 10; however, since it
was still present in Asterisk 1.8 and still useful for reporting
purposes, it was decided to re-add the value. This patch re-adds
the 'LastMsgsSent' field in the response to an AMI/CLI 'sip show
peer [peer]' command, and makes it so that the value of
lastmsgssent is updated appropriately. The value should now
display the new/old message counts for a particular peer. (closes
issue ASTERISK-17866) Reported by: Steve Davies patches by:
ast-17866-rb1272.patch (License #5041 by irroot) Modified
slightly for this commit Review:
https://reviewboard.asterisk.org/r/1939 ........ Merged revisions
367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 367369 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-22 17:29 +0000 [r367274-367309] Terry Wilson <twilson@digium.com>
* main/channel.c, /, include/asterisk/cel.h,
main/channel_internal_api.c, include/asterisk/channel.h,
main/cel.c, main/asterisk.c: Fix race condition for CEL
LINKEDID_END event This patch fixes to situations that could
cause the CEL LINKEDID_END event to be missed. 1) During a core
stop gracefully, modules are unloaded when ast_active_channels ==
0. The LINKDEDID_END event fires during the channel destructor.
This means that occasionally, the cel_* module will be unloaded
before the channel is destroyed. It seemed generally useful to
wait until the refcount of all channels == 0 before unloading, so
I added a channel counter and used it in the shutdown code. 2)
During a masquerade, ast_channel_change_linkedid is called. It
calls ast_cel_check_retire_linkedid which unrefs the linkedid in
the linkedids container in cel.c. It didn't ref the new linkedid.
Now it does. Review: https://reviewboard.asterisk.org/r/1900/
........ Merged revisions 367292 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367299 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c: Resolve crash in subscribing for MWI
notifications ASTOBJ_UNREF sets the variable to NULL after
unreffing it, so the variable should definitely not be used after
that. To solve this in the two cases that affect subscribing for
MWI notifications, we instead save the ref locally, and unref
them in the error conditions. (closes issue ASTERISK-19827)
Reported by: B. R Review:
https://reviewboard.asterisk.org/r/1940/ ........ Merged
revisions 367266 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367267 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-21 22:45 +0000 [r367227] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Made ast_queue_hangup() and
ast_queue_hangup_with_cause() lock instead of trylock. It made no
sense to trylock the channel and then unconditionally lock the
channel right after.
2012-05-21 20:35 +0000 [r367189] Kinsey Moore <kmoore@digium.com>
* channels/chan_iax2.c: Make chan_iax2 reject cause code
indications correctly If chan_iax2 does not reject the
PVT_CAUSE_CODE frames, the cause will not be stored properly.
2012-05-21 20:31 +0000 [r367163-367183] Mark Michelson <mmichelson@digium.com>
* include/asterisk/callerid.h, channels/chan_sip.c,
main/callerid.c: Revert revision 367163. This should have been
committed to my team trunk-digiumphones branch instead of trunk.
* include/asterisk/callerid.h, channels/chan_sip.c,
main/callerid.c: Add "send to voicemail" Digium phone
functionality to Asterisk. This change accommodates two methods
by which calls can be directed to a user's voicemail. * Incoming
calls can be redirected to any user's voicemail. * Established
calls can be blind transferred to any user's voicemail. Digium
phones indicate the desire to direct a call to voicemail by using
a Diversion header with a reason parameter of "send_to_vm". This
patch adds the "send_to_vm" reason as a valid redirecting reason.
In addition, chan_sip.c has been modified to update redirecting
information on the transferred channel by reading a Diversion
header on a REFER request. (closes issue AST-871) Reported by
Malcolm Davenport Review: https://reviewboard.asterisk.org/r/1925
2012-05-21 17:39 +0000 [r367124] Terry Wilson <twilson@digium.com>
* include/asterisk/astobj2.h: Minor documentation change
2012-05-18 19:39 +0000 [r367080] Jonathan Rose <jrose@digium.com>
* configs/queues.conf.sample, CHANGES, apps/app_queue.c: app_queue:
Per Member ringinuse option and deprecation of ignorebusy Adds a
number of methods for controlling the setting of 'ringinuse'
which is basically the same concept as the old ignorebusy
setting, only now the per member setting always controls whether
or not the member is actually ringed while in use. A CLI command
and a manager action have been added to change a given queue
member's ringinuse option while Asterisk is running and the an
argument has been added for adding members with deliberately set
ringinuse in queues.conf Some effort has been made to ensure
compatability with dialplans and databases still referring to
'ignorebusy'. (issue ASTERISK-19536) reported by: Philippe
Lindheimer Review: https://reviewboard.asterisk.org/r/1919/
2012-05-18 17:54 +0000 [r367010-367029] Mark Michelson <mmichelson@digium.com>
* channels/chan_dahdi.c, /, main/say.c: Address MISSING_BREAK
static analysis reports some more. This addresses core findings 4
and 6. Moises Silva helped me by stating that a break could be
safely added to the case where it is added in chan_dahdi.c In
say.c, I have added a comment indicating that static analysis
complains but that it is currently unknown if this is correct.
This fixes all core findings of this type. (closes issue
ASTERISK-19662) reported by Matthew Jordan ........ Merged
revisions 367027 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367028 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
Fix memory leak of SSL_CTX structures in TLS core. SSL_CTX
structures were allocated but never freed. This was a bigger
issue for clients than servers since new SSL_CTX structures could
be allocated for each connection. Servers, on the other hand,
typically set up a single SSL_CTX for their lifetime. This is
solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an
ssl_ctx on it, it is freed so that a new one can take its place.
2. A companion to ast_ssl_setup() called ast_ssl_teardown() has
been added so that servers can properly free their SSL_CTXs.
(issue ASTERISK-19278) ........ Merged revisions 367002 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 367003 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-18 15:51 +0000 [r366917-366955] Matthew Jordan <mjordan@digium.com>
* channels/chan_dahdi.c, /, channels/chan_sip.c, funcs/func_odbc.c,
main/cli.c: Fix more memory leaks This patch adds to what was
fixed in r366880. Specifically, it addresses the following: *
chan_sip: dispose of an allocated frame in off nominal code paths
in sip_rtp_read * func_odbc: when disposing of an allocated
resultset, ensure that any rows that were appended to that
resultset are also disposed of * cli: free the created return
string buffer in another off nominal code path * chan_dahdi: free
a frame that was allocated by the dsp layer if we choose not to
process that frame (issue ASTERISK-19665) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/1922/ ........
Merged revisions 366944 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 366948 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/netsock2.c, res/res_rtp_asterisk.c, main/pbx.c,
res/res_calendar_exchange.c, res/res_calendar_icalendar.c,
apps/app_page.c, /, funcs/func_dialgroup.c, channels/chan_sip.c,
apps/app_record.c, res/res_calendar_caldav.c, res/res_jabber.c,
apps/app_queue.c, channels/chan_iax2.c, main/enum.c,
main/editline/term.c, main/config.c, res/res_srtp.c, main/cli.c,
main/editline/tokenizer.c, main/data.c, channels/chan_dahdi.c,
funcs/func_odbc.c, main/features.c, apps/app_minivm.c,
main/editline/readline.c, channels/sip/config_parser.c,
main/xmldoc.c, res/res_calendar.c, apps/app_voicemail.c: Fix a
variety of memory leaks This patch addresses a number of memory
leaks in a variety of modules that were found by a static
analysis tool. A brief summary of the changes: * app_minivm: free
ast_str objects on off nominal paths * app_page: free the
ast_dial object if the requested channel technology cannot be
appended to the dialing structure * app_queue: if a penalty rule
failed to match any existing rule list names, the created rule
would not be inserted and its memory would be leaked * app_read:
dispose of the created silence detector in the presence of off
nominal circumstances * app_voicemail: dispose of an allocated
unique ID field for MWI event un-subscribe requests in off
nominal paths; dispose of configuration objects when using the
secret.conf option * chan_dahdi: dispose of the allocated frame
produced by ast_dsp_process * chan_iax2: properly unref peer in
CLI command "iax2 unregister" * chan_sip: dispose of the
allocated frame produced by sip_rtp_read's call of
ast_dsp_process; free memory in parse unit tests *
func_dialgroup: properly deref ao2 object grhead in nominal path
of dialgroup_read * func_odbc: free resultset in off nominal
paths of odbc_read * cli: free match_list in off nominal paths of
CLI match completion * config: free comment_buffer/list_buffer
when configuration file load is unchanged; free the same buffers
any time they were created and config files were processed *
data: free XML nodes in various places * enum: free context
buffer in off nominal paths * features: free ast_call_feature in
off nominal paths of applicationmap config processing * netsock2:
users of ast_sockaddr_resolve pass in an ast_sockaddr struct that
is allocated by the method. Failures in ast_sockaddr_resolve
could result in the users of the method not knowing whether or
not the buffer was allocated. The method will now not allocate
the ast_sockaddr struct if it will return failure. * pbx: cleanup
hash table traversals in off nominal paths; free ignore pattern
buffer if it already exists for the specified context * xmldoc:
cleanup various nodes when we no longer need them *
main/editline: various cleanup of pointers not being freed before
being assigned to other memory, cleanup along off nominal paths *
menuselect/mxml: cleanup of value buffer for an attribute when
that attribute did not specify a value * res_calendar*: responses
are allocated via the various *_request method returns and should
not be allocated in the various write_event methods; ensure
attendee buffer is freed if no data exists in the parsed node;
ensure that calendar objects are de-ref'd appropriately *
res_jabber: free buffer in off nominal path * res_musiconhold:
close the DIR* object in off nominal paths * res_rtp_asterisk: if
we run out of ports, close the rtp socket object and free the rtp
object * res_srtp: if we fail to create the session in libsrtp,
destroy the temporary ast_srtp object (issue ASTERISK-19665)
Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1922 ........ Merged revisions
366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 366881 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-18 14:27 +0000 [r366896] Jonathan Rose <jrose@digium.com>
* channels/sip/dialplan_functions.c: chan_sip: Fix a small
TEST_FRAMEWORK related error that prevents compiling Introduced
with r366842, a function call made only with TEST_FRAMEWORK
enabled was missing an argument since the function arguments were
changed.
2012-05-18 14:21 +0000 [r366843-366888] Kinsey Moore <kmoore@digium.com>
* /, channels/sip/config_parser.c: Reorder and renumber tests
appropriately It appears that a patch did not apply properly when
adding tests 12 and 13 and test 11 was duplicated. These tests
have been reordered and renumbered such that they make sense.
........ Merged revisions 366882 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 366884 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/channel.c: Make the new SIP_CAUSE backend behave more like
the original SIP_CAUSE There was a slight discrepancy in the
behaviors of the old SIP_CAUSE and the new SIP_CAUSE/HANGUPCAUSE
when a channel had been originated and had not yet been answered.
This caused the noload_res_srtp_attempt_srtp test to fail since
the SIP_CAUSE variable was never actually set. This behavior has
been restored.
2012-05-17 16:28 +0000 [r366842] Jonathan Rose <jrose@digium.com>
* include/asterisk/logger.h, main/channel.c,
channels/sip/include/dialog.h, main/pbx.c, channels/chan_sip.c,
main/channel_internal_api.c, main/logger.c,
include/asterisk/channel.h, CHANGES, channels/sip/include/sip.h,
main/cli.c: logger: Adds additional support for call id logging
and chan_sip specific stuff This patch improves the handling of
call id logging significantly with regard to transfers and adding
APIs to better handle specific aspects of logging. Also, changes
have been made to chan_sip in order to better handle the creation
of callids and to enable the monitor thread to bind itself to a
particular call id when a dialog is determined to be related to a
callid. It then unbinds itself before returning to normal
monitoring. review: https://reviewboard.asterisk.org/r/1886/
2012-05-17 13:21 +0000 [r366746] Matthew Jordan <mjordan@digium.com>
* channels/chan_dahdi.c, /, res/res_calendar_ews.c: Fix checking
bounds of array index after using it; improper sizeof This patch
fixes two problems pointed out by a static analysis tool. * In
chan_dahdi, when an event is handled the index of the sub channel
is first obtained. In very off nominal cases, the method that
determines the index can return a negative value. In the event
handling code, whether or not the index returned is valid was
being checked after that value was used to index into an array.
This patch makes it so the value is checked before any indexing
is done. * In res_calendar_ews, sizeof was being passed a pointer
instead of the struct to determine the amount of memory to
allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes
issue ASTERISK-19671) Reported by: Matt Jordan ........ Merged
revisions 366740 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 366741 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-16 18:00 +0000 [r366663-366700] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/astobj2.h: Remove missed idx parameter to some
ao2 global holder macros.
* include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c:
Change ao2 global array to ao2 global object holder. Review:
https://reviewboard.asterisk.org/r/1921/
2012-05-15 23:41 +0000 [r366599] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Correct misuse of ast_strip_quoted() when
getting a Diversion header's reason parameter. The use here was
assuming that the pointer would be updated, but the updated
string is actually returned by ast_strip_quoted() instead.
........ Merged revisions 366597 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 366598 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-15 19:36 +0000 [r366462-366546] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c: The predial routine must be run on the
local;1 channel. When ast_call() operates on a local channel, it
copies a lot of things from the local;1 channel to the local;2
channel. This includes among other things, channel variables and
party id information. Other reasons it was a bad idea to run
predial on the local;2 channel: 1) The channel has not been
completely setup. The ast_call() completes the setup. 2) The
local;2 caller and connected line party information is opposite
to any other channels predial runs on. (And it hasn't been setup
yet.) * Partially back out -r366183 by removing the chan_local
implementation of the struct ast_channel_tech.pre_call callback.
* CHANGES, apps/app_followme.c: Add predial support to FollowMe.
Like the new predial feature for Dial. This adds the same b/B
options to FollowMe. Review:
https://reviewboard.asterisk.org/r/1910/
* channels/chan_local.c: Make chan_local use the API call instead
of inlining its own version.
2012-05-14 20:15 +0000 [r366413] Mark Michelson <mmichelson@digium.com>
* /, pbx/dundi-parser.c: Fix two more coverity constant expression
result findings. These correspond to findings 0 and 1 in the core
findings of ASTERISK-19649. After contacting Mark Spencer, he was
unsure of what the intent behind these lines of code were, so
they are being axed. For Asterisk 1.8 and 10, the output of
debugging DUNDi frames will not be changed, but for trunk the
"Retry" portion will be omitted since it does not properly
distinguish retransmissions from initial frames. (closes issue
ASTERISK-19649) Reported by Matthew Jordan ........ Merged
revisions 366409 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 366412 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-14 19:44 +0000 [r366408] Kinsey Moore <kmoore@digium.com>
* channels/chan_unistim.c, apps/app_dial.c, main/rtp_engine.c,
channels/chan_vpb.cc, channels/chan_sip.c, UPGRADE.txt,
channels/chan_gtalk.c, channels/chan_console.c,
channels/chan_iax2.c, apps/app_queue.c, apps/app_followme.c,
channels/chan_oss.c, channels/chan_jingle.c, main/channel.c,
channels/chan_phone.c, main/dial.c, channels/chan_misdn.c,
channels/chan_skinny.c, funcs/func_frame_trace.c,
main/features.c, channels/chan_h323.c, main/file.c,
channels/chan_alsa.c, configs/sip.conf.sample,
include/asterisk/frame.h, channels/chan_mgcp.c: Commit framework
for HANGUPCAUSE (replacement for SIP_CAUSE) This is the starting
point for the Asterisk 11: Who Hung Up work and provides a
framework which will allow channel drivers to report the types of
hangup cause information available in SIP_CAUSE without incurring
the overhead of the MASTER_CHANNEL dialplan function. The initial
implementation only includes cause generation for chan_sip and
does not include cause code translation utilities. This change
deprecates SIP_CAUSE and replaces its method of reporting cause
codes with the new framework. This change also deprecates the
'storesipcause' option in sip.conf. Review:
https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221)
2012-05-14 19:27 +0000 [r366401] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Fix broken reinvite glare scenario. To
make a long story short, reinvite glares were broken because
Asterisk would invert the To and From headers when ACKing a 491
response. The reason was because the initreq of the dialog was
being changed to the incoming glared reinvite instead of being
set to the outgoing glared reinvite. This change has three parts
* In handle_incoming, we never will reject an ACK because it has
a to-tag present, even if we think the request may be out of
dialog. * In handle_request_invite, we do not change the initreq
when receiving a reinvite to which we will respond with a 491. *
In handle_request_invite, several superflous settings up
pendinginvite have been removed since this is dones automatically
by transmit_response_reliable Review:
https://reviewboard.asterisk.org/r/1911 ........ Merged revisions
366389 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 366390 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-14 13:42 +0000 [r366351] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* configure, configure.ac, autoconf/ast_pkgconfig.m4 (added): Macro
AST_PKG_CONFIG_CHECK to use chkconfig AST_PKG_CONFIG_CHECK:
Similar to AST_EXT_LIB_CHECK, but simply uses pkg-config data.
This simple version only uses pkg-config(1)'s tests. This commit
also uses the macro to test for GTK2 and GMIME (instead of the
current direct usage of pkg-config). Review:
https://reviewboard.asterisk.org/r/1906/
2012-05-12 00:03 +0000 [r366298] Russell Bryant <russell@russellbryant.com>
* /, addons/format_mp3.c: format_mp3: Fix a possible crash in
mp3_read(). This patch fixes a potential crash in mp3_read() by
not assuming that dbuf has enough data to finish filling up the
output buffer. The patch also makes sure that the dbuf state gets
reset after we know we read everything out of it already. In
passing, this patch includes some other cleanups of this module,
including stripping trailing whitespace, formatting fixes based
on coding guidelines, and removing a number of unused members
from the private state struct. (closes issue ASTERISK-19761)
Reported by: Chris Maciejewsk Tested by: Chris Maciejewsk
........ Merged revisions 366296 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 366297 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-10 23:49 +0000 [r366183-366242] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /: * Made ast_change_name() hold the channels
container lock while changing the channel name. * Eliminate
redundant list not empty check in clone_variables(). ........
Merged revisions 366240 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 366241 from
http://svn.asterisk.org/svn/asterisk/branches/10
* apps/app_dial.c: Tweak app_dial predial documentation.
* apps/app_dial.c, main/channel.c, channels/chan_local.c,
include/asterisk/channel.h: Run predial routine on local;2
channel where you would expect. Before this patch, the predial
routine executes on the ;1 channel of a local channel pair.
Executing predial on the ;1 channel of a local channel pair is of
limited utility. Any channel variables set by the predial routine
executing on the ;1 channel will not be available when the local
channel executes dialplan on the ;2 channel. * Create
ast_pre_call() and an associated pre_call() technology callback
to handle running the predial routine. If a channel technology
does not provide the callback, the predial routine is simply run
on the channel. Review: https://reviewboard.asterisk.org/r/1903/
2012-05-10 20:56 +0000 [r366169] Kinsey Moore <kmoore@digium.com>
* funcs/func_speex.c, main/pbx.c, res/res_calendar_icalendar.c, /,
channels/chan_sip.c, funcs/func_lock.c, channels/chan_agent.c,
channels/sip/reqresp_parser.c, main/devicestate.c,
pbx/dundi-parser.c, channels/chan_iax2.c, channels/iax2-parser.c,
main/config.c, res/res_monitor.c, main/channel.c, main/cdr.c,
res/ael/pval.c, main/data.c, channels/chan_dahdi.c,
main/tcptls.c, main/manager.c, main/features.c, main/app.c,
main/event.c, pbx/pbx_dundi.c, res/res_odbc.c, main/xmldoc.c,
apps/app_voicemail.c: Resolve FORWARD_NULL static analysis
warnings This resolves core findings from ASTERISK-19650 numbers
0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84,
87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers
26, 33, and 29 were already resolved. Those skipped were either
extended/deprecated or in areas of code that shouldn't be
disturbed. (Closes issue ASTERISK-19650) ........ Merged
revisions 366167 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 366168 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-10 18:35 +0000 [r366126] Jonathan Rose <jrose@digium.com>
* main/pbx.c, channels/sig_analog.c, /, channels/chan_sip.c,
funcs/func_lock.c, main/features.c, main/acl.c,
channels/iax2-provision.c, apps/app_queue.c,
channels/chan_iax2.c, res/ael/ael.flex, funcs/func_devstate.c,
main/asterisk.c, main/xmldoc.c, apps/app_voicemail.c: Coverity
Report: Fix issues for error type CHECKED_RETURN for core (issue
ASTERISK-19658) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1905/ ........ Merged
revisions 366094 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 366106 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-10 16:22 +0000 [r366062] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Close the proper tcptls_session when
session creation fails. (issue AST-998) Reported by: Thomas
Arimont Tested by: Thomas Arimont ........ Merged revisions
366052 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 366053 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-10 15:57 +0000 [r366007-366051] Jonathan Rose <jrose@digium.com>
* /, funcs/func_cdr.c, main/features.c, apps/app_disa.c,
apps/app_chanspy.c: Coverity Report: Fix issues for error type
UNINIT in Core supported modules (issue ASTERISK-19652) Reported
by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1909/
........ Merged revisions 366048 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 366049 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, codecs/codec_dahdi.c: Block on frameout if the hardware has
enough samples to complete a frame. Fixes some problems with
skipping audio in elaborate scenarios involving multiple codecs
by making codec_dahdi operate in a more synchronous fashion
similar to codec_g729. This change also fixes the use of file
conversion tools from Asterisk's CLI. This change may cause the
thread responsible for transcoding audio to block briefly (Shaun
Ruffell describes this as 'several milliseconds') while waiting
for the hardware transcoder. (closes issue ASTERISK-19643)
reported by: Shaun Ruffell Patches:
0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch
uploaded by Shaun Ruffell (license 5417) ........ Merged
revisions 365989 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 365990 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-09 19:26 +0000 [r366002] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* Makefile: pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect Allow
menuselect to get its set of CFLAGS and LDFLAGS through the
environment of Make: make BUILD_CFLAGS="whatever"
BUILD_LDFLAGS="whatever" Review:
https://reviewboard.asterisk.org/r/1907/
2012-05-09 17:58 +0000 [r365951] Richard Mudgett <rmudgett@digium.com>
* configs/followme.conf.sample, apps/app_followme.c: Improve
FollowMe accept/decline DTMF string matching. If you hit the
wrong DTMF digit trying to accept/decline a FollowMe call, you
had to wait for the prompt to repeat to try again. * Make
FollowMe compare the last DTMF digits received to the
accept/decline matching strings.
2012-05-09 16:36 +0000 [r365913] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Prevent sip_pvt refleak when an
ast_channel outlasts its corresponding sip_pvt. chan_sip was
coded under the assumption that a SIP dialog with an owner
channel will always be destroyed after the owner channel has been
hung up. However, there are situations where the SIP dialog can
time out and auto destruct before the corresponding channel has
hung up. A typical example of this would be if the 'h' extension
in the dialplan takes a long time to complete. In such cases,
__sip_autodestruct() would complain about the dialog being auto
destroyed with an owner channel still in place. The problem is
that even once the owner channel was hung up, the sip_pvt would
still be linked in its ao2_container because nothing would ever
unlink it. The fix for this is that if __sip_autodestruct() is
called for a sip_pvt that still has an owner channel in place,
the destruction is rescheduled for 10 seconds in the future. This
will continue until the owner channel is finally hung up. (closes
issue ASTERISK-19425) reported by David Cunningham Patches:
ASTERISK-19425.patch uploaded by Mark Michelson (License #5049)
(closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by
Dean Vesvuio ........ Merged revisions 365896 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 365898 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-09 02:35 +0000 [r365766-365856] Richard Mudgett <rmudgett@digium.com>
* configs/followme.conf.sample, UPGRADE.txt, apps/app_followme.c:
Keep answered FollowMe calls until call accepted or last step
times out.
* apps/app_followme.c: Put winning FollowMe outgoing call on hold
if the caller put it on hold. The FollowMe caller call leg is
usually answered and listening to MOH. The caller could put the
call on hold while FollowMe is looking for a winner. The winning
outgoing call is now immediately placed on hold if the caller has
put the call on hold before the winning call was selected.
* apps/app_followme.c: Restructure how the FollowMe outgoing
channel list is handled.
* apps/app_followme.c: Addendum to -r365766. Since it is no longer
allocated.
* apps/app_followme.c: Make FollowMe findmeexec() put the list head
on the stack instead of mallocing it. Why this tiny struct was
malloced instead of the 28k struct in the last change is beyond
me. Just doing my part to help stamp out sillyness.
2012-05-08 21:46 +0000 [r365751] Sean Bright <sean@malleable.com>
* apps/app_externalivr.c: Add interrupt ('I') command to
ExternalIVR. Sending the 'I' command from an external process
will cause the current playlist to be cleared, including stopping
any audio file that is currently playing. This is useful when you
want to interrupt audio playback only when specific DTMF is
entered by the caller.
2012-05-08 21:41 +0000 [r365633-365749] Richard Mudgett <rmudgett@digium.com>
* apps/app_followme.c: Make FollowMe app_exec() not declare a 28k
struct on the stack. Helping to stamp out stack abuse.
* apps/app_followme.c: Simplify findmeexec() parameter passing.
* /, apps/app_followme.c: * Fix FollowMe memory leak on error paths
in app_exec(). * Fix FollowMe leaving recorded caller name file
on error paths in app_exec(). * Use correct buffer dimension
define in struct fm_args.namerecloc[]. This fixes unexpected
namerecloc filename length restriction. ........ Merged revisions
365692 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 365701 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_followme.c: * Fix accept/decline DTMF buffer
overwrite in FollowMe. * Made use MAX_YN_STRING define to make
all accept/decline DTMF buffers the same size. Just using 20
isn't good enough when someone didn't get the memo. * Fix stupid
use of a global variable in FollowMe. (ynlongest) * Fix bit field
declarations in FollowMe. ........ Merged revisions 365631 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 365632 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-08 15:57 +0000 [r365576] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Send more accurate identification
information in dialog-info SIP NOTIFYs. This uses the calling
channel's caller ID and connected line information to populate
the remote and local identities in the dialog-info NOTIFY when an
extension is ringing. There is a bit of an oddity here, and that
is that we seed the remote target with the To header of the
outbound call rather than the from header. This is because it was
reported that seeding with the from header caused hints to be
broken with certain SNOM devices. A comment has been added to the
code to explain this. (closes issue ASTERISK-16735) reported by
Maciej Krajewski patches: local_remote_hint2.diff uploaded by
Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark
Michelson (license #5049) Tested by Niccolo Belli ........ Merged
revisions 365574 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 365575 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-07 20:08 +0000 [r365532] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Change comment to use local channel name
designators in features.c
2012-05-07 18:58 +0000 [r365480] Matthew Jordan <mjordan@digium.com>
* main/pbx.c, apps/app_voicemail.c: Fix channel opaquification
slip-up in r365477 Those channels are opaque now...
2012-05-07 18:51 +0000 [r365479] Richard Mudgett <rmudgett@digium.com>
* /, tests/test_config.c: Fix type punned compiler warning in
test_config.c ........ Merged revisions 365476 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 365478 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-07 18:42 +0000 [r365477] Matthew Jordan <mjordan@digium.com>
* main/pbx.c, /, apps/app_voicemail.c: Support VoiceMail d() option
when extension does not exist in channel's context The VoiceMail
d([c]) option is documented to accept digits for a new extension
in context <c>, if played during the greeting. This option works
fine if the extension being redirected to has an extension with
the same initial digit in the channel's current context. If that
digit did not happen to exist in some extension, a dialplan match
would fail and the user would not be redirected. This patch fixes
it such that if the <c> option is used, the extensions are
matched in that context as opposed to the caller's original
context. (closes issue ASTERISK-18243) Reported by: mjordan
Tested by: mjordan Review:
https://reviewboard.asterisk.org/r/1892 ........ Merged revisions
365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 365475 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-04 22:17 +0000 [r365400] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c, funcs/func_aes.c, main/features.c,
apps/app_followme.c, channels/chan_iax2.c,
channels/sip/config_parser.c, pbx/pbx_config.c,
apps/app_chanspy.c, apps/app_stack.c, main/config.c,
apps/app_voicemail.c: Fix many issues from the NULL_RETURNS
Coverity report Most of the changes here are trivial NULL checks.
There are a couple optimizations to remove the need to check for
NULL and outboundproxy parsing in chan_sip.c was rewritten to
avoid use of strtok. Additionally, a bug was found and fixed with
the parsing of outboundproxy when "outboundproxy=," was set.
(Closes issue ASTERISK-19654) ........ Merged revisions 365398
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 365399 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-04 17:38 +0000 [r365356] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c, /: Fix local channel chains optimizing
themselves out of a call. * Made chan_local.c:check_bridge()
check the return value of ast_channel_masquerade(). In long
chains of local channels, the masquerade occasionally fails to
get setup because there is another masquerade already setup on an
adjacent local channel in the chain. * Made the outgoing local
channel (the ;2 channel) flush one voice or video frame per
optimization attempt. * Made sure that the outgoing local channel
also does not have any frames in its queue before the masquerade.
* Made do the masquerade immediately to minimize the chance that
the outgoing channel queue does not get any new frames added and
thus unconditionally flushed. * Made block indication -1 (Stop
tones) event when the local channel is going to optimize itself
out. When the call is answered, a chain of local channels pass
down a -1 indication for each bridge. This blizzard of -1 events
really slows down the optimization process. (closes issue
ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec
Davis Review: https://reviewboard.asterisk.org/r/1894/ ........
Merged revisions 365313 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 365320 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-04 15:52 +0000 [r365300] Mark Michelson <mmichelson@digium.com>
* res/res_rtp_asterisk.c, /: Fix core FINDING 2, FINDING 3, and
FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report.
These three all are in RTP code that attempts to print the number
of sequence number cycles in an RTCP RR report. The code was
masking out the upper 16 bits and then shifting the number right
by 16 bits. This led to an all zero result in all cases. The fix
is to do the shift without the bit masking. (issue
ASTERISK-19649) ........ Merged revisions 365298 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 365299 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-03 19:36 +0000 [r365248] Michael L. Young <elgueromexicano@gmail.com>
* tests/test_security_events.c: Update security events unit tests
The security events framework API was changed in Asterisk 10 but
the unit tests were not updated at the same time. This patch does
the following: * Adds two more security events that were added to
the API * Add challenge, received_challenge and received_hash in
the inval_password security event unit test (Closes issue
ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael
L. Young Patches: issue-asterisk-19760-trunk.diff uploaded by
Michael L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/1897/
2012-05-03 18:43 +0000 [r365213] Sean Bright <sean@malleable.com>
* CHANGES: Update documentation references in CHANGES to reflect
the correct pages on the wiki. The current CHANGES file refers to
doc/ in many places and those files no longer exist.
2012-05-03 15:05 +0000 [r365161] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooh323.c, /,
addons/ooh323c/src/h323/H323-MESSAGES.h,
addons/ooh323c/src/h323/H323-MESSAGESEnc.c: Fix warning of
Coverity Static analysis, change H225ProtocolIdentifier from
value to pointer per functions that use this. (close issue
ASTERISK-19670) Reported by: Matt Jordan Patches:
ASTERISK-19670.patch (License #5415) ........ Merged revisions
365159 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 365160 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-03 14:47 +0000 [r365158] Sean Bright <sean@malleable.com>
* apps/app_externalivr.c, CHANGES: Add IPv6 support to ExternalIVR.
Review: https://reviewboard.asterisk.org/r/1896/
2012-05-03 14:35 +0000 [r365157] Alexandr Anikin <may@telecom-service.ru>
* /, addons/ooh323c/src/ooq931.c: Fix coverity static analysis
warning, allocate full ie structure instead of without data
buffer (close issue ASTERISK-19674) Reported by: Matt Jordan
Patches: ASTERISK-19674.patch (License #5415) ........ Merged
revisions 365143 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 365155 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-02 17:43 +0000 [r365084] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, /, main/cel.c: Multiple revisions
365006,365068 ........ r365006 | twilson | 2012-05-02 10:49:03
-0500 (Wed, 02 May 2012) | 12 lines Fix a CEL LINKEDID_END race
and local channel linkedids This patch has the ;2 channel inherit
the linkedid of the ;1 channel and fixes the race condition by no
longer scanning the channel list for "other" channels with the
same linkedid. Instead, cel.c has an ao2 container of linkedid
strings and uses the refcount of the string as a counter of how
many channels with the linkedid exist. Not only does this
eliminate the race condition, but it also allows us to look up
the linkedid by the hashed key instead of traversing the entire
channel list. Review: https://reviewboard.asterisk.org/r/1895/
........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02
May 2012) | 11 lines Don't leak a ref if out of memory and can't
link the linkedid If the ao2_link fails, we are most likely out
of memory and bad things are going to happen. Before those bad
things happen, make sure to clean up the linkedid references.
This patch also adds a comment explaining why linkedid can't be
passed to both local channel allocations and combines two ao2_ref
calls into 1. Review: https://reviewboard.asterisk.org/r/1895/
........ Merged revisions 365006,365068 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 365083 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-02 15:59 +0000 [r365011] Jason Parker <jparker@digium.com>
* channels/chan_sip.c: Save the address on which a MESSAGE was
received, so it can be used in MESSAGE() This is useful in cases
where chan_sip may be listening on multiple addresses.
2012-05-02 02:51 +0000 [r364966] Matthew Jordan <mjordan@digium.com>
* /, main/audiohook.c: Only log a failure to get read/write samples
from factories if it didn't happen In audiohook_read_frame_both,
anytime samples are obtained from the read/write factories a
debug statement is logged stating that samples were not obtained
from the factories. This statement used to only occur if
option_debug was turned on and no samples were obtained; in some
refactoring when the option_debug statement was removed, the
"else" clause was removed as well. This patch makes it so that
those debug log statements only occur if the condition leading up
to them actually happened. ........ Merged revisions 364965 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-01 23:23 +0000 [r364915] Mark Michelson <mmichelson@digium.com>
* channels/chan_sip.c: Remove a function that has been marked
unused since Asterisk 1.6.0. The reason I'm removing this is that
Coverity reported a STRAY_SEMICOLON issue here. Since the
function has been unused for so long, I just elected to remove it
altogether. (closes issue ASTERISK-19660)
2012-05-01 23:21 +0000 [r364910] Richard Mudgett <rmudgett@digium.com>
* /, main/astobj2.c: Fixed __ao2_ref() validating user_data twice.
(closes issue ASTERISK-19755) Reported by: Gunther Kelleter
Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther
Kelleter ........ Merged revisions 364902 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364903 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-01 23:11 +0000 [r364901] Mark Michelson <mmichelson@digium.com>
* /, funcs/func_volume.c: Fix Coverity-reported ARRAY_VS_SINGLETON
error. As it turned out, this wasn't a huge deal. We were calling
ast_app_parse_options() for a set of options of which none took
arguments. The proper thing to do for this case is to pass NULL
for the "args" parameter here. We were instead passing a
seemingly-randomly chosen char * from the function. While this
would never get written to, you can rest assured things would
have gotten bad had new options (which took arguments) been added
to func_volume. (closes issue ASTERISK-19656) ........ Merged
revisions 364899 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364900 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-01 22:00 +0000 [r364846] Richard Mudgett <rmudgett@digium.com>
* channels/chan_local.c, /: * Fix error path resouce leak in
local_request(). * Restructure local_request() to reduce
indentation. ........ Merged revisions 364840 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364845 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-01 21:49 +0000 [r364844] Jason Parker <jparker@digium.com>
* main/manager.c, /: Prevent a potential crash when using manager
hooks. Found by me while poking at DPMA-127. ........ Merged
revisions 364841 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364842 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-01 19:10 +0000 [r364788] Kinsey Moore <kmoore@digium.com>
* /, apps/app_confbridge.c: Play conf-placeintoconf message to the
correct channel Correct the code in app_confbridge to play the
conf-placeintoconf message to the marked user entering the bridge
instead of to the conference while the marked user hears silence.
(closes issue ASTERISK-19641) Reported-by: Mark A Walters
........ Merged revisions 364786 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364787 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-05-01 18:29 +0000 [r364785] Jonathan Rose <jrose@digium.com>
* /, main/app.c: Fix bad check in voicemail functions for
ast_inboxcount2_func Check looks for ast_inboxcount_func instead
of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes
issue ASTERISK-19718) Reported by: Corey Farrell Patches:
ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell
(license 5909) ........ Merged revisions 364769 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364777 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-30 19:51 +0000 [r364708] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Revert revision 360862. Revision 360862
was intended to improve identities sent in dialog-info NOTIFY
requests. Some users reported that hint became broken once this
was done. It's not clear exactly what part of the patch has
caused this regression, but broken hints are bad. For now, this
revision is being reverted so that the next releases of Asterisk
do not have bad behavior in them. The original reported issue
will have to be fixed differently in the next version of
Asterisk. (issue ASTERISK-16735) ........ Merged revisions 364706
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 364707 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-30 17:17 +0000 [r364654] Mark Murawki <markm@intellasoft.net>
* /, main/logger.c: Merged revisions 364635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) |
10 lines Sanatize result from bfd_find_nearest_line
(BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file
to null resulting in a crash when strrchr(file) runs (closes
issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark
Murawski ........ ........ Merged revisions 364650 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-30 16:59 +0000 [r364652] Alexandr Anikin <may@telecom-service.ru>
* /, addons/ooh323cDriver.c: Fix use freed pointer in return value
from call thread (issue ASTERISK-19663) Reported by: Matt Jordan
Patches: ASTERISK-19663-ooh323.patch (License #5415) ........
Merged revisions 364649 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364651 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-29 19:50 +0000 [r364580] Matthew Jordan <mjordan@digium.com>
* formats/format_ilbc.c, /, formats/format_sln.c,
formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c,
formats/format_g723.c, formats/format_h263.c,
formats/format_h264.c, formats/format_wav_gsm.c,
formats/format_siren14.c, formats/format_gsm.c,
formats/format_g719.c, formats/format_siren7.c,
formats/format_g729.c: Fix error that caused truncate operations
to fail Another very inappropriate placement of a ')' (again
introduced in r362151) caused the various truncate operations to
attempt to truncate the sound file at a position of '0'. (issue
ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810)
Reported by: colbec ........ Merged revisions 364578 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364579 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-29 02:23 +0000 [r364537] Michael L. Young <elgueromexicano@gmail.com>
* /, apps/confbridge/conf_config_parser.c: Fix configuring custom
sound_leader_has_left in confbridge.conf The configuration option
to specify a custom sound_leader_has_left file for a conference
bridge was not being parsed. This patch fixes it so that a custom
sound file will now be used. (closes issue ASTERISK-19771)
Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young
Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak
(license 6380) Review: https://reviewboard.asterisk.org/r/1884/
........ Merged revisions 364536 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-28 20:24 +0000 [r364500] Joshua Colp <jcolp@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
channels/sip/include/sip.h: Add support for lightweight NAT
keepalive. If enabled using the keepalive option in sip.conf a
small packet will be sent at a regular interval to keep the NAT
mapping open. This is lightweight as the remote side does not
need to parse and handle a SIP message. (closes issue AST-783)
Review: https://reviewboard.asterisk.org/r/1756/
2012-04-28 01:33 +0000 [r364437-364462] Russell Bryant <russell@russellbryant.com>
* main/md5.c: md5: supress some compiler warnings. md5.c: In
function ‘MD5Final’: md5.c:154:2: error: dereferencing
type-punned pointer will break strict-aliasing rules
[-Werror=strict-aliasing] md5.c:155:2: error: dereferencing
type-punned pointer will break strict-aliasing rules
[-Werror=strict-aliasing] There is an md5 unit test and it still
passes.
* configure, include/asterisk/autoconfig.h.in, res/res_corosync.c,
configure.ac: res_corosync: Fix build against corosync 2.0.
* apps/app_minivm.c: app_minivm: Fix a couple compiler warnings.
The warnings were about argv[0] being used uninitialized, which
is correct. Just remove setting username to this value, since
username is set again before it actually gets used.
* main/features.c, CHANGES: features: Add FEATURE() and
FEATUREMAP() functions. Add two new dialplan functions: FEATURE()
and FEATUREMAP(). FEATURE() lets you set some of the
configuration options from the [general] section of features.conf
on a per-channel basis. FEATUREMAP() lets you customize the key
sequence used to activate built-in features, such as blindxfer,
and automon. See the built-in documentation for details. Review:
https://reviewboard.asterisk.org/r/1871/
2012-04-28 00:31 +0000 [r364436] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, CHANGES: PreDial - Ability to run dialplan on
callee and caller channels before Dial. Thanks to Mark Murawski
for the initial patch and feature definition. (closes issue
ASTERISK-19548) Reported by: Mark Murawski Review:
https://reviewboard.asterisk.org/r/1878/ Review:
https://reviewboard.asterisk.org/r/1229/
2012-04-27 22:54 +0000 [r364397] Terry Wilson <twilson@digium.com>
* /, tests/test_config.c (added), main/config.c: Multiple revisions
364365,364369 ........ r364365 | twilson | 2012-04-27 17:31:01
-0500 (Fri, 27 Apr 2012) | 11 lines Fix ast_parse_arg numeric
type range checking and add tests ast_parse_arg wasn't checking
for strto* parse errors or limiting the results by the actual
range of the numeric types. This patch fixes that and adds unit
tests as well. Review: https://reviewboard.asterisk.org/r/1879/
........ Merged revisions 364340 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012)
| 2 lines Add missing test_config.c ........ Merged revisions
364365,364369 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-27 22:11 +0000 [r364343] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Don't attempt to make use of the
dynamic_exclude_static ACL if DNS lookup fails. (closes issue
ASTERISK-18321) Reported by Dan Lukes Patches:
ASTERISK-18321.patch by Mark Michelson (license #5049) ........
Merged revisions 364341 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364342 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-27 19:30 +0000 [r364287] Matthew Jordan <mjordan@digium.com>
* /, include/asterisk/time.h: Prevent overflow in calculation in
ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms
attempts to calculate the difference, in milliseconds, between
two timeval structs, and return the difference in a 64-bit
integer. Unfortunately, it assumes that the long tv_sec/tv_usec
members in the timeval struct are large enough to hold the
calculated values before it returns. On 64-bit machines, this
might be the case, as a long may be 64-bits. On 32-bit machines,
however, a long may be less (32-bits), in which case, the
calculation can overflow. This overflow caused significant
problems in MixMonitor, which uses the method to determine if an
audio factory, which has not presented audio to an audiohook, is
merely late in providing said audio or will never provide audio.
In an overflow situation, the audiohook would incorrectly
determine that an audio factory that will never provide audio is
merely late instead. This led to situations where a MixMonitor
never recorded any audio. Note that this happened most frequently
when that MixMonitor was started by the ConfBridge application
itself, or when the MixMonitor was attached to a Local channel.
(issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben
Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license
#6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark
Murawski Tested by: Michael L. Young Patches:
32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan)
(closes issue ASTERISK-19471) Reported by: feyfre Tested by:
feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review:
https://reviewboard.asterisk.org/r/1889/ ........ Merged
revisions 364277 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364285 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-27 18:59 +0000 [r364260] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Allow SIP pvts involved in Replaces
transfers to fall out of reference sooner Unref the SIP pvt
stored in the refer structure as soon as it is no longer needed
so that the pvt and associated file descriptors can be freed
sooner. This change makes a reference decrement unnecessary in
code that handles SIP BYE/Also transfers which should not touch
the reference anyway. (Closes issue ASTERISK-19579) Reported by:
Maciej Krajewski Tested by: Maciej Krajewski ........ Merged
revisions 364258 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364259 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-27 14:45 +0000 [r364205] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Allow for reloading SRTP crypto keys
within the same SIP dialog As a continuation of the patch in
r356604, which allowed for the reloading of SRTP keys in
re-INVITE transfer scenarios, this patch addresses the more
common case where a new key is requested within the context of a
current SIP dialog. This can occur, for example, when certain
phones request a SIP hold. Previously, once a dialog was
associated with an SRTP object, any subsequent attempt to process
crypto keys in any SDP offer - either the current one or a new
offer in a new SIP request - were ignored. This patch changes
this behavior to only ignore subsequent crypto keys within the
current SDP offer, but allows future SDP offers to change the
keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested
by: Thomas Arimont Review:
https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions
364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 364204 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-27 12:58 +0000 [r364164] Stefan Schmidt <sst@sil.at>
* res/res_calendar_icalendar.c, /, res/res_calendar_caldav.c: fix a
wrong behavior of alarm timezones in caldav and icalendar when an
alarm doesnt use utc. This change uses the same timezone from the
start time. ........ Merged revisions 364163 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-26 21:11 +0000 [r364082-364110] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_directed_pickup.c: Update Pickup application
documentation. (With feeling this time.) ........ Merged
revisions 364108 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364109 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/features.c: Fix DTMF atxfer running h exten after the
wrong bridge ends. When party B does an attended transfer of
party A to party C, the attending bridge between party B and C
should not be running an h exten when the bridge ends. Running an
h exten now sets a softhangup flag to ensure that an AGI will run
in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the
party B channel for the attending bridge between party B and C.
(closes issue AST-870) (closes issue ASTERISK-19717) Reported by:
Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev
Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch
uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario
........ Merged revisions 364060 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364065 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-26 19:33 +0000 [r364048] Terry Wilson <twilson@digium.com>
* /, main/asterisk.c: Add more constness to the end_buf pointer in
the netconsole issue ASTERISK-18308 Review:
https://reviewboard.asterisk.org/r/1876/ ........ Merged
revisions 364046 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 364047 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-26 13:59 +0000 [r363989] Olle Johansson <oej@edvina.net>
* apps/app_queue.c: Code formatting fixes.
2012-04-26 13:31 +0000 [r363988] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Fix reference leaks involving SIP
Replaces transfers The reference held for SIP blind transfers
using the Replaces header in an INVITE was never freed on success
and also failed to be freed in some error conditions. This caused
a file descriptor leak since the RTP structures in use at the
time of the transfer were never freed. This reference leak and
another relating to subscriptions in the same code path have now
been corrected. (closes issue ASTERISK-19579) ........ Merged
revisions 363986 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 363987 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-26 09:48 +0000 [r363936] Alec L Davis <sivad.a@paradise.net.nz>
* /, channels/chan_sip.c: chan_sip: [general] maxforwards, not
checked for a value greater than 255 The peer maxforwards is
checked for both '< 1' and '> 255', but the default 'maxforwards'
in the [general] section is only checked for '< 1' alecdavis
(license 585) Reported by: alecdavis Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1888/ ........ Merged
revisions 363934 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 363935 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-26 03:12 +0000 [r363689-363877] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_directed_pickup.c: Update Pickup application
documentation. (Even better) ........ Merged revisions 363875
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 363876 from
http://svn.asterisk.org/svn/asterisk/branches/10
* apps/app_directed_pickup.c: * Put more information in
pickup_exec() LOG_NOTICE. * Delay duplicating a string on the
stack in pickup_exec().
* /, apps/app_directed_pickup.c: Update Pickup application
documentation. ........ Merged revisions 363788 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 363789 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/chan_dahdi.c, /, channels/sig_pri.c: Make
DAHDISendCallreroutingFacility wait 5 seconds for a reply before
disconnecting the call. Some switches may not handle the
call-deflection/call-rerouting message if the call is
disconnected too soon after being sent. Asteisk was not waiting
for any reply before disconnecting the call. * Added a 5 second
delay before disconnecting the call to wait for a potential
response if the peer does not disconnect first. (closes issue
ASTERISK-19708) Reported by: mehdi Shirazi Patches:
jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: rmudgett ........ Merged revisions 363730
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 363734 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
Clear ISDN channel resetting state if the peer continues to use
it. Some ISDN switches occasionally fail to send a RESTART
ACKNOWLEDGE in response to a RESTART request. * Made the second
SETUP received after sending a RESTART request clear the channel
resetting state as if the peer had sent the expected RESTART
ACKNOWLEDGE before continuing to process the SETUP. The peer may
not be sending the expected RESTART ACKNOWLEDGE. (issue
ASTERISK-19608) (issue AST-844) (issue AST-815) Patches:
jira_ast_815_v1.8.patch (license #5621) patch uploaded by
rmudgett (modified) ........ Merged revisions 363687 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 363688 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-25 13:57 +0000 [r363480-363637] Olle Johansson <oej@edvina.net>
* apps/app_queue.c: Add documentation Thanks Tilghman!
* apps/app_queue.c: Formatting changes only
* apps/app_followme.c, apps/app_queue.c: Use the DEFINED value for
musicclass length. For some reason, features.c has it's own
definition. Should propably be fixed too.
* main/channel.c, configs/asterisk.conf.sample, CHANGES,
include/asterisk/options.h, main/asterisk.c: Make it possible to
change the minimum DTMF duration in asterisk.conf Asterisk has a
setting for the minimum allowed DTMF. If we get shorter DTMF
tones, these will be changed to the minimum on the outbound call
leg. (closes issue ASTERISK-19772) Review:
https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested
by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for
this patch: agave-dtmf-duration-asterisk-conf-1.8
* main/say.c: Formatting fixes Developer guidelines are important.
* main/channel.c: Formatting fixes Found a small amount of curly
brackets in my hotel room here in Denmark. I hereby donate them
to the Asterisk project.
2012-04-25 01:26 +0000 [r363377-363430] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Fix recalled party B feature flags for a
failed DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF
atxfer to C 3) B hangs up 4) C does not answer 5) B is called
back 6) B answers 7) B cannot initiate transfers anymore * Add
dial features datastore to recalled party B channel that is a
copy of the original party B channel's dial features datastore. *
Extracted add_features_datastore() from
add_features_datastores(). * Renamed struct ast_dial_features
features_caller and features_callee members to my_features and
peer_features respectively. These better names eliminate the need
for some explanatory comments. * Simplified code accessing the
struct ast_dial_features datastore. (closes issue ASTERISK-19383)
Reported by: lgfsantos ........ Merged revisions 363428 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 363429 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/features.c: Hangup affected channel in error paths of
bridge_call_thread(). ........ Merged revisions 363375 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 363376 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-24 17:52 +0000 [r363335] Terry Wilson <twilson@digium.com>
* /, main/asterisk.c: OpenBSD doesn't have rawmemchr, use strchr
(closes issue ASTERISK-19758) Reported by: Barry Miller Tested
by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller
(license 5434) ........ Merged revisions 362868 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362869 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-23 17:05 +0000 [r363269] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, apps/app_queue.c: Make app_dial and app_queue
use new macro and gosub calls. * Simplify some code in app_dial
and app_queue by calling ast_app_exec_macro() and
ast_app_exec_sub(). * Fix minor locking issue in app_dial for
post-answer macro/gosub MACRO/GOSUB_RESULT=GOTO: handling.
2012-04-23 16:08 +0000 [r363215] Tilghman Lesher <tilghman@meg.abyt.es>
* /, main/astfd.c: On some platforms, O_RDONLY is not a flag to be
checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX
specification does not mandate how these 3 flags must be
specified, only that one of the three must be specified in every
call. ........ Merged revisions 363209 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 363212 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-23 14:48 +0000 [r363159] Jonathan Rose <jrose@digium.com>
* main/manager.c, /: AST-2012-004: Fix an error that allows AMI
users to run shell commands sans authorization. As detailed in
the advisory, AMI users without write authorization for SYSTEM
class AMI actions were able to run system commands by going
through other AMI commands which did not require that
authorization. Specifically, GetVar and Status allowed users to
do this by setting their variable/s options to the SHELL or EVAL
functions. Also, within 1.8, 10, and trunk there was a similar
flaw with the Originate action that allowed users with originate
permission to run MixMonitor and supply a shell command in the
Data argument. That flaw is fixed in those versions of this
patch. (closes issue ASTERISK-17465) Reported By: David Woolley
Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose
(license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose
(license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose
(license 6182) ........ Merged revisions 363117 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
Merged revisions 363141 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 363156 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-23 14:10 +0000 [r363105-363108] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: AST-2012-006: Fix crash in UPDATE
handling when no channel owner exists If Asterisk receives a SIP
UPDATE request after a call has been terminated and the channel
has been destroyed but before the SIP dialog has been destroyed,
a condition exists where a connected line update would be
attempted on a non-existing channel. This would cause Asterisk to
crash. The patch resolves this by first ensuring that the SIP
dialog has an owning channel before attempting a connected line
update. If an UPDATE request is received and no channel is
associated with the dialog, a 481 response is sent. (closes issue
ASTERISK-19770) Reported by: Thomas Arimont Tested by: Matt
Jordan Patches: ASTERISK-19278-2012-04-16.diff uploaded by Matt
Jordan (license 6283) ........ Merged revisions 363106 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 363107 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_skinny.c: AST-2012-005: Fix remotely exploitable
heap overflow in keypad button handling When handling a keypad
button message event, the received digit is placed into a fixed
length buffer that acts as a queue. When a new message event is
received, the length of that buffer is not checked before placing
the new digit on the end of the queue. The situation exists where
sufficient keypad button message events would occur that would
cause the buffer to be overrun. This patch explicitly checks that
there is sufficient room in the buffer before appending a new
digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant
........ Merged revisions 363100 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
Merged revisions 363102 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 363103 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-21 11:45 +0000 [r363045-363046] Russell Bryant <russell@russellbryant.com>
* res/res_corosync.c: res_corosync: Recover if corosync gets
restarted. If corosync gets restarted while Asterisk is running,
automatically recover.
* res/res_corosync.c: res_corosync: reimplement "corosync show
members" command. Reimplement the "corosync show members" CLI
command using a CPG iterator instead of the cpg_membership_get
API call. This will also show all CPG members, including those in
groups other than 'asterisk', which may be useful at some point
for debugging purposes.
2012-04-21 01:46 +0000 [r362920-362999] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, /: Update app_dial M and U option GOTO return
value documentation. ........ Merged revisions 362997 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362998 from
http://svn.asterisk.org/svn/asterisk/branches/10
* include/asterisk/app.h, main/app.c, apps/app_stack.c: Fix
connected-line/redirecting interception gosubs executing more
than intended. * Redo ast_app_run_sub()/ast_app_exec_sub() to use
a known return point so execution will stop after the routine
returns there. (s@gosub_virtual_context:1) * Create
ast_app_exec_macro() and ast_app_exec_sub() to run the macro and
gosub application respectively with the parameter string already
created.
* main/rtp_engine.c: Move debug message in
ast_rtp_instance_early_bridge_make_compatible(). Move debug
message in ast_rtp_instance_early_bridge_make_compatible() to be
output when what it states has actually happened.
2012-04-20 16:50 +0000 [r362919] Michael L. Young <elgueromexicano@gmail.com>
* /, main/event.c: Add missing payload type to events API The
Security Events Framework API was changed while adding the
generation of security events in chan_sip. A payload type and
name was missed from being added to struct ie_maps. (closes issue
ASTERISK-19759) Reported by: Michael L. Young Patches:
issue-asterisk-19759.diff uploaded by Michael L. Young (license
5026) ........ Merged revisions 362918 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-20 16:23 +0000 [r362867-362888] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, channels/chan_dahdi.c, channels/chan_local.c,
channels/chan_misdn.c, main/rtp_engine.c: Use
ast_channel_lock_both() where it was inlined before. The
CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the
channel lock was originally obtained is overkill where
ast_channel_lock_both() was inlined.
* main/pbx.c: * Add more information to some messages in
__ast_pbx_run(). * Simplify some dialplan priority setting code
in ast_explicit_goto() because of opaquification.
2012-04-20 14:50 +0000 [r362817] Terry Wilson <twilson@digium.com>
* /, apps/app_speech_utils.c: Document Speech* apps hangup on
failure and suggest TryExec The Speech API apps return -1 on
failure, which will hang up the channel. This may not be
desirable behavior for some, but it isn't something that can be
changed without breaking people's dialplans or writing an option
to all of the Speech apps that does what TryExec already does.
This patch documents the hangup behavior of the apps, and
suggests TryExec as the solution. (closes issue AST-813) ........
Merged revisions 362815 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362816 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-20 00:57 +0000 [r362779] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, UPGRADE.txt, include/asterisk/channel.h, CHANGES,
channels/sig_pri.c, funcs/func_callerid.c: Add original party id
and reason support. ISDN ETSI PTP and Q.SIG (And SS7 in future)
have support for reporting who was the original redirecting party
of a call. * Added support for the original redirecting party and
reason to the REDIRECTING function and the system core as well as
to the stubbed locations in sig_pri.c. Review:
https://reviewboard.asterisk.org/r/1829/
2012-04-19 22:01 +0000 [r362731] Walter Doekes <walter+asterisk@wjd.nu>
* funcs/func_version.c, /: Fix documentation for
${VERSION(ASTERISK_VERSION_NUM)}. ........ Merged revisions
362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 362730 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-19 21:14 +0000 [r362682] Michael L. Young <elgueromexicano@gmail.com>
* /, tests/test_linkedlists.c, tests/test_poll.c: Add leading and
trailing backslashes A couple of unit tests did not have have
leading or trailing backslashes when setting their test category
resulting in a warning message being displayed. Added the
backslash where needed. ........ Merged revisions 362680 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362681 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-19 21:01 +0000 [r362679] Richard Mudgett <rmudgett@digium.com>
* /, configs/queues.conf.sample: Update membermacro and membergosub
documentation in queues.conf.sample. ........ Merged revisions
362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 362678 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-19 19:05 +0000 [r362635] Terry Wilson <twilson@digium.com>
* addons/chan_ooh323.c, apps/app_alarmreceiver.c,
channels/iax2-provision.c, res/snmp/agent.c: Convert some
strncpys to ast_copy_string Review:
https://reviewboard.asterisk.org/r/1732/
2012-04-19 16:10 +0000 [r362588] Sean Bright <sean@malleable.com>
* /, apps/app_externalivr.c: Prevent a crash in ExternalIVR when
the 'S' command is sent first. If the first command sent from an
ExternalIVR client is an 'S' command, we were blindly removing
the first element from the play list and deferencing it, even if
it was NULL. This corrects that and also locks appropriately in
one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski
........ Merged revisions 362586 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362587 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-19 14:35 +0000 [r362538] Terry Wilson <twilson@digium.com>
* /, main/asterisk.c: Handle multiple commands per connection via
netconsole Asterisk would accept multiple NULL-delimited CLI
commands via the netconsole socket, but would occasionally miss a
command due to the command not being completely read into the
buffer. This patch ensures that any partial commands get moved to
the front of the read buffer, appended to, and properly sent.
(closes issue ASTERISK-18308) Review:
https://reviewboard.asterisk.org/r/1876/ ........ Merged
revisions 362536 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362537 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-19 02:40 +0000 [r362497] Matthew Jordan <mjordan@digium.com>
* channels/chan_unistim.c, /, main/tdd.c, main/jitterbuf.c,
apps/app_sms.c, main/stdtime/localtime.c, utils/extconf.c,
addons/chan_mobile.c, main/format_pref.c, main/asterisk.c: Fix a
variety of potential buffer overflows * chan_mobile: Fixed an
overrun where the cind_state buffer (an integer array of size 16)
would be overrun due to improper bounds checking. At worst, the
buffer can be overrun by a total of 48 bytes (assuming 4-byte
integers), which would still leave it within the allocated memory
of struct hfp. This would corrupt other elements in that struct
but not necessarily cause any further issues. * app_sms: The
array imsg is of size 250, while the array (ud) that the data is
copied into is of size 160. If the size of the inbound message is
greater then 160, up to 90 bytes could be overrun in ud. This
would corrupt the user data header (array udh) adjacent to ud. *
chan_unistim: A number of invalid memmoves are corrected. These
would move data (which may or may not be valid) into the ends of
these buffers. * asterisk: ast_console_toggle_loglevel does not
check that the console log level being set is less then or equal
to the allowed log levels of 32. * format_pref: In
ast_codec_pref_prepend, if any occurrence of the specified codec
is not found, the value used to index into the array pref->order
would be one greater then the maximum size of the array. *
jitterbuf: If the element being placed into the jitter buffer
lands in the last available slot in the jitter history buffer,
the insertion sort attempts to move the last entry in the buffer
into one slot past the maximum length of the buffer. Note that
this occurred for both the min and max jitter history buffers. *
tdd: If a read from fsk_serial returns a character that is
greater then 32, an attempt to read past one of the statically
defined arrays containing the values that character maps to would
occur. * localtime: struct ast_time and tm are not the same size
- ast_time is larger, although it contains the elements of tm
within it in the same layout. Hence, when using memcpy to copy
the contents of tm into ast_time, the size of tm should be used,
as opposed to the size of ast_time. * extconf: this treats
ast_timing's minmask array as if it had a length of 48, when it
has defined the size of the array as 24. pbx.h defines minmask as
having a size of 48. (issue ASTERISK-19668) Reported by: Matt
Jordan ........ Merged revisions 362485 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362496 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-18 17:03 +0000 [r362432] Michael L. Young <elgueromexicano@gmail.com>
* tests/test_security_events.c: Fix building security events test
The Security Events Framework API changed in trunk to support
IPv6. This broke the building of the security events test which
was based around IPv4. This patches fixes the build by changing
the test to conform to the new changes. (related to issue
ASTERISK-19447) Review: https://reviewboard.asterisk.org/r/1874/
2012-04-18 16:41 +0000 [r362430] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, /, channels/sig_pri.c: Add
ability to ignore layer 1 alarms for BRI PTMP lines. Several
telcos bring the BRI PTMP layer 1 down when the line is idle.
When layer 1 goes down, Asterisk cannot make outgoing calls.
Incoming calls could fail as well because the alarm processing is
handled by a different code path than the Q.931 messages. * Add
the layer1_presence configuration option to ignore layer 1 alarms
when the telco brings layer 1 down. This option can be configured
by span while the similar DAHDI driver teignorered=1 option is
system wide. This option unlike layer2_persistence does not
require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA
ABE-2845 ........ Merged revisions 362428 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362429 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-17 21:23 +0000 [r362365-362380] Matthew Jordan <mjordan@digium.com>
* /, main/format_pref.c: Handle case where an unknown format is
used to get the preferred codec size In ast_codec_pref_getsize,
if an unknown format is passed to the method, no preferred codec
will be selected and a negative number will be used to index into
the format list. The method now logs an unknown format as a
warning, and returns an empty format list. (issue ASTERISK-19655)
Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1863/ ........ Merged
revisions 362377 from
http://svn.asterisk.org/svn/asterisk/branches/10
* res/res_rtp_asterisk.c, /, res/res_agi.c, res/res_musiconhold.c:
Fix places in resources where a negative return value could
impact execution This patch addresses a number of modules in
resources that did not handle the negative return value from
function calls adequately. This includes: * res_agi.c: if the
result of the read function is a negative number, indicating some
failure, the result would instead be treated as the number of
bytes read. This patch now treats negative results in the same
manner as an end of file condition, with the exception that it
also logs the error code indicated by the return. *
res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor
to srcfd, and instead assigns a negative value, that file
descriptor could later be passed to functions that require a
valid file descriptor. If spawn_mp3 fails, we now immediately
retry instead of continuing in the logic. * res_rtp_asterisk.c:
if no codec can be matched between two RTP instances in a peer to
peer bridge, we immediately return instead of attempting to use
the codec payload type as an index to determine the appropriate
negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged
revisions 362362 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362364 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-17 21:10 +0000 [r362363] Jonathan Rose <jrose@digium.com>
* res/res_config_curl.c, res/res_config_pgsql.c,
res/res_config_odbc.c, /: Make use of va_args more appropriate to
form in various res_config modules plus utils. A number of
va_copy operations weren't matched with a corresponding va_end in
res_config_odbc. Also, there was a potential for va_end to be
invoked twice on the same va_arg in utils, which would mean
invoking va_end on an undefined variable... which is bad. va_end
is removed from various functions in config_pgsql and config_curl
since they aren't making their own copy. The invokers of those
functions are responsible for calling va_end on them. (issue
ASTERISK-19451) Reported by: Walter Doekes Review:
https://reviewboard.asterisk.org/r/1848/ ........ Merged
revisions 362354 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362357 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-17 21:08 +0000 [r362358-362361] Matthew Jordan <mjordan@digium.com>
* main/manager.c, /, main/asterisk.c: Fix places in main where a
negative return value could impact execution This patch addresses
a number of modules in main that did not handle the negative
return value from function calls adequately, or were not
sufficiently clear that the conditions leading to improper
handling of the return values could not occur. This includes: *
asterisk.c: A negative return value from the read function would
be used directly as an index into a buffer. We now check for
success of the read function prior to using its result as an
index. * manager.c: Check for failures in mkstemp and lseek when
handling the temporary file created for processing data returned
from a CLI command in action_command. Also check that the result
of an lseek is sanitized prior to using it as the size of a
memory map to allocate. (issue ASTERISK-19655) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........
Merged revisions 362359 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362360 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, funcs/func_env.c: Fix places where a negative return from
ftello could be used as invalid input In a variety of locations
in both reading and writing a file, the result from the C library
function ftello is used as input to other functions. For the
parameters and functions in question, a negative value is invalid
input. This patch checks the return value from the ftello
function to determine if we were able to determine the current
position in the file stream and, if not, fail gracefully. (issue
ASTERISK-19655) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1863/ ........ Merged
revisions 362355 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362356 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-17 18:57 +0000 [r362307] Walter Doekes <walter+asterisk@wjd.nu>
* channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c,
funcs/func_env.c, res/res_phoneprov.c, channels/chan_gtalk.c,
cdr/cdr_pgsql.c, res/res_http_post.c, res/res_musiconhold.c,
res/res_jabber.c, res/res_format_attr_celt.c,
channels/chan_dahdi.c, funcs/func_groupcount.c,
apps/app_osplookup.c, funcs/func_odbc.c, main/ast_expr2f.c,
apps/app_minivm.c, channels/chan_alsa.c, codecs/codec_resample.c,
formats/format_h264.c, res/res_format_attr_silk.c,
res/res_config_ldap.c, main/ast_expr2.fl,
res/res_config_sqlite3.c, channels/chan_sip.c,
channels/vcodecs.c, codecs/codec_g726.c, main/data.c,
res/res_corosync.c, channels/chan_h323.c, codecs/codec_dahdi.c,
funcs/func_callerid.c, main/asterisk.c, res/res_odbc.c: Avoid
cppcheck warnings; removing unused vars and a bit of cleanup.
Patch by: junky Review: https://reviewboard.asterisk.org/r/1743/
2012-04-17 18:29 +0000 [r362306] Matthew Jordan <mjordan@digium.com>
* /, formats/format_sln.c, formats/format_vox.c,
formats/format_wav.c, formats/format_pcm.c,
formats/format_wav_gsm.c, formats/format_siren14.c,
formats/format_gsm.c, formats/format_g719.c,
formats/format_siren7.c: Fix error that caused seek format
operations to set max file size to '1' or '0' A very
inappropriate placement of a ')' (introduced in r362151) caused
the maximum size of a file to be set as the result of a
comparison operation, as opposed to the result of the ftello
operation. This resulted in seeking being restricted to the
beginning of the file, or 1 byte into the file. Thanks to the
Asterisk Test Suite for properly freaking out about this on at
least one test. (issue ASTERISK-19655) Reported by: Matt Jordan
........ Merged revisions 362304 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362305 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-17 15:00 +0000 [r362266] Michael L. Young <elgueromexicano@gmail.com>
* /, channels/chan_sip.c: Turn off warning message when bind
address is set to any. When a bind address is set to an ANY
address (udpbindport=::), a warning message is displayed stating
that "Address remapping activated in sip.conf but we're using
IPv6, which doesn't need it. Please remove 'localnet' and/or
'externaddr' settings." But if one is running dual stack, we
shouldn't be told to turn those settings off. This patch checks
if the bind address is an ANY address or not. The warning message
will now only be displayed if the bind address is NOT an ANY
address and IPv6 is being used. Also, updated the copyright year.
(closes issue ASTERISK-19456) Reported by: Michael L. Young
Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff
uploaded by Michael L. Young (license 5026) ........ Merged
revisions 362253 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362264 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-16 21:58 +0000 [r362203-362206] Matthew Jordan <mjordan@digium.com>
* channels/chan_dahdi.c, /, channels/chan_agent.c: Fix negative
return handling in channel drivers In chan_agent, while handling
a channel indicate, the agent channel driver must obtain a lock
on both the agent channel, as well as the channel the agent
channel is using. To do so, it attempts to lock the other channel
first, then unlock the agent channel which is locked prior to
entry into the indicate handler. If this unlock fails with a
negative return value, which can occur if the object passed to
agent_indicate is an invalid ao2 object or is NULL, the return
value is passed directly to strerror, which can only accept
positive integer values. In chan_dahdi, the return value of
dahdi_get_index is used to directly index into the sub-channel
array. If dahd_get_index returns a negative value, it would use
that value to index into the array, which could cause an invalid
memory access. If dahdi_get_index returns a negative number, we
now default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........
Merged revisions 362204 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362205 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_voicemail.c: Fix handling of negative return code
when storing voicemails in ODBC storage When storing a voicemail
message using an ODBC connection to a database, the voicemail
message is first stored on disk. The sound file associated with
the message is read into memory before being transmitted to the
database. When this occurs, a failure in the C library's lseek
function would cause a negative value to be passed to the mmap as
the size of the memory map to create. This would almost certainly
cause the creation of the memory map to fail, resulting in the
message being lost. (issue ASTERISK-19655) Reported by: Matt
Jordan Review: https://reviewboard.asterisk.org/r/1863 ........
Merged revisions 362201 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362202 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-16 21:20 +0000 [r362200] Michael L. Young <elgueromexicano@gmail.com>
* main/manager.c, main/security_events.c,
channels/sip/security_events.c, CHANGES,
include/asterisk/security_events_defs.h: Add IPv6 address support
to security events framework. The current Security Events
Framework API only supports IPv4 when it comes to generating
security events. This patch does the following: * Changes the
Security Events Framework API to support IPV6 and updates the
components that use this API. * Eliminates an error message that
was being generated since the current implementation was treating
an IPv6 socket address as if it was IPv4. * Some copyright dates
were updated on files touched by this patch. (closes issue
ASTERISK-19447) Reported by: Michael L. Young Tested by: Michael
L. Young Patches: security_events_ipv6v3.diff uploaded by Michael
L. Young (license 5026) Review:
https://reviewboard.asterisk.org/r/1777/
2012-04-16 20:17 +0000 [r362153] Matthew Jordan <mjordan@digium.com>
* formats/format_ilbc.c, /, formats/format_sln.c,
formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c,
formats/format_g723.c, formats/format_h263.c,
formats/format_h264.c, formats/format_wav_gsm.c,
formats/format_siren14.c, formats/format_gsm.c,
formats/format_g719.c, formats/format_siren7.c,
formats/format_g729.c: Check for IO stream failures in various
format's truncate/seek operations For the formats that support
seek and/or truncate operations, many of the C library calls used
to determine or set the current position indicator in the file
stream were not being checked. In some situations, if an error
occurred, a negative value would be returned from the library
call. This could then be interpreted inappropriately as
positional data. This patch checks the return values from these
library calls before using them in subsequent operations. (issue
ASTERISK-19655) Reported by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1863/ ........ Merged
revisions 362151 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362152 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-13 16:12 +0000 [r362081-362085] Jonathan Rose <jrose@digium.com>
* apps/app_forkcdr.c, /: Make ForkCDR e option not set end time of
the newly forked CDR log Prior to this patch, ForkCDR's e option
would immediately set the end time of the forked CDR to that of
the CDR that is being terminated. This resulted in the new CDR's
end time being roughly the same as it's beginning time (which is
in turn roughly the same as the original's end time). (closes
issue ASTERISK-19164) Reported by: Steve Davies Patches:
cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012)
........ Merged revisions 362082 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362084 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_meetme.c: Send relative path named recordings to the
meetme directory instead of sounds Prior to this patch, no effort
was made to parse the path name to determine a proper destination
for recordings of MeetMe's r option. This fixes that. Review:
https://reviewboard.asterisk.org/r/1846/ ........ Merged
revisions 362079 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 362080 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-12 20:08 +0000 [r362043] Paul Belanger <paul.belanger@polybeacon.com>
* main/srv.c: Convert SRV lookup message to debug level This helps
clean up the Asterisk CLI by converting the log message from
verbose to debug
2012-04-12 16:29 +0000 [r361998] Richard Mudgett <rmudgett@digium.com>
* configs/asterisk.conf.sample, UPGRADE.txt, pbx/pbx_config.c,
include/asterisk/options.h, main/asterisk.c: Add option to invoke
the extensions.conf stdexten using the legacy macro method.
ASTERISK-18809 eliminated the legacy macro invocation of the
stdexten in favor of the Gosub method without a means of
backwards compatibility. (issue ASTERISK-18809) (closes issue
ASTERISK-19457) Reported by: Matt Jordan Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1855/
2012-04-12 16:25 +0000 [r361968-361987] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_iax2.c: Make trunkfreq take effect when set
Previously, setting trunkfreq had no effect on initial load or on
reload and only ever used the default value. This causes
trunkfreq to be used appropriately on initial load and reload.
(closes issue ASTERISK-19521) Patch-by: Jaco Kroon ........
Merged revisions 361972 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361981 from
http://svn.asterisk.org/svn/asterisk/branches/10
* Makefile, build_tools/cflags.xml, /,
build_tools/menuselect-deps.in, codecs/gsm/src/k6opt.s,
configure, codecs/gsm/Makefile, configure.ac, Makefile.rules,
makeopts.in, codecs/lpc10/Makefile: Simplify build system
architecture optimization This change to the build system rips
out any usage of PROC along with architecture-specific
optimizations in favor of using -march=native where it is
supported. This fixes broken builds on 64bit Intel systems and
results in better optimized code on systems running GCC 4.2+.
Review: https://reviewboard.asterisk.org/r/1852/ (closes issue
ASTERISK-19462) ........ Merged revisions 361955 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361956 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-11 17:20 +0000 [r361909] Jonathan Rose <jrose@digium.com>
* /, configs/queues.conf.sample, apps/app_queue.c: Change default
value of 'ignorebusy' on Queue members so that behavior is more
like 1.8 Prior to this patch, in order to restore that behavior,
a function would have to be used on the QueueMember to make the
ringinuse option do anything, which is pretty unreasonable.
(closes issue ASTERISK-19536) reported by: Philippe Lindheimer
Review: https://reviewboard.asterisk.org/r/1860/ ........ Merged
revisions 361907 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-10 21:50 +0000 [r361856] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Prevent invalid access of free'd memory
if DAHDI channel during an MWI event In the MWI processing loop,
when a valid event occurs the temporary caller ID information is
deallocated. If a new DAHDI channel is successfully created, the
event is passed up to the analog_ss_thread without error and the
loop exits. If, however, the DAHDI channel is not created, then
the caller ID struct has been free'd, and the gains reset to
their previous level. This will almost certainly cause an invalid
access to the free'd memory, either in subsequent calls to
callerid_free or calls to callerid_feed. * Rework the -r361705
patch to better manage the cs and mtd allocated resources. *
Fixed use of mwimonitoractive flag to be correct if the
mwi_thread() fails to start. ........ Merged revisions 361854
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 361855 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-10 19:58 +0000 [r361659-361805] Matthew Jordan <mjordan@digium.com>
* /, main/http.c: Fix crash caused by unloading or reloading of
res_http_post When unlinking itself from the registered HTTP
URIs, res_http_post could inadvertently free all URIs registered
with the HTTP server. This patch modifies the unregister method
to only free the URI that is actually being unregistered, as
opposed to all of them. ........ Merged revisions 361803 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361804 from
http://svn.asterisk.org/svn/asterisk/branches/10
* funcs/func_curl.c, /: Allow func_curl to exit gracefully if list
allocation fails during write If the global_curl_info data
structure could not be allocated, the datastore associated with
the operation would be free'd, but the function would not return.
This would later dereference the datastore, almost certainly
causing Asterisk to crash. With this patch, if the data structure
is not allocated the method will return an error code, and not
attempt any further operation. ........ Merged revisions 361753
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 361754 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/chan_dahdi.c, /: Prevent invalid access of free'd memory
if DAHDI channel during an MWI event In the MWI processing loop,
when a valid event occurs the temporary caller ID information is
deallocated. If a new DAHDI channel is successfully created, the
event is passed up to the analog_ss_thread without error and the
loop exits. If, however, the DAHDI channel is not created, then
the caller ID struct has been free'd, and the gains reset to
their previous level. This will almost certainly cause an invalid
access to the free'd memory, either in subsequent calls to
callerid_free or calls to callerid_feed. This patch makes it so
that we only free the caller ID structure if a DAHDI channel is
successfully created, and we bump the gains back up if we fail to
make a DAHDI channel. ........ Merged revisions 361705 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361706 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, funcs/func_global.c: Change SHARED function to use a safe
traversal when modifying a variable When the SHARED function
modifies a variable, it removes it from its list of variables and
reinserts the new value at the head of the list of variables.
Doing this inside a standard list traversal can be dangerous, as
the standard list traversal does not account for the list being
changed. While the code in question should not cause a use after
free violation due to its breaking out of the loop after freeing
the variable, it could lead to a maintenance issue if the loop
was modified. This also fixes a violation reported by a static
analysis tool, which also makes this code easier to maintain in
the future. ........ Merged revisions 361657 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361658 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-06 22:00 +0000 [r361561-361608] Matthew Jordan <mjordan@digium.com>
* /, res/res_calendar_ews.c: Fix memory leak in res_calendar_ews
when event email address node is empty If the XML calendar data
returned by a Microsoft Exchange Web Service specifies an XML
Event E-Mail Address ("EmailAddress"), and no e-mail address is
provided, a condition existed where an ast_calendar_attendee
struct would be allocated but not appended to the list of
attendees. Because of that, the memory associated with the
attendee would never be freed. This patch frees the memory if no
e-mail address is provided. ........ Merged revisions 361606 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361607 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_meetme.c: Fix memory leak when using MeetMeAdmin 'e'
option with user specified A memory leak/reference counting leak
occurs if the MeetMeAdmin 'e' command (eject last user that
joined) is used in conjunction with a specified user. Regardless
of the command being executed, if a user is specified for the
command, MeetMeAdmin will look up that user. Because the 'e'
option kicks the last user that joined, as opposed to the one
specified, the reference to the user specified by the command
would be leaked when the user variable was assigned to the last
user that joined. ........ Merged revisions 361558 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361560 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-06 19:58 +0000 [r361523] Richard Mudgett <rmudgett@digium.com>
* /, main/message.c: Don't add an empty MESSAGE_DATA(key) header if
it doesn't already exist. Doing Set(MESSAGE_DATA(key)=) would add
an empty key header if the key header did not already exist. If
it already existed it would delete it. * Made msg_set_var_full()
exit early if the named variable did not already exist and the
value to set is empty. ........ Merged revisions 361522 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-06 18:19 +0000 [r361476] Kinsey Moore <kmoore@digium.com>
* channels/chan_unistim.c, main/pbx.c, /, channels/chan_sip.c,
funcs/func_strings.c, formats/format_ogg_vorbis.c,
channels/console_video.c, apps/app_ices.c, channels/chan_gtalk.c,
channels/chan_iax2.c, res/res_config_sqlite.c, res/res_srtp.c,
main/cdr.c, main/tcptls.c, channels/console_gui.c,
funcs/func_channel.c, apps/app_sms.c, addons/chan_mobile.c,
apps/app_chanspy.c, main/xmldoc.c, channels/chan_mgcp.c,
res/res_config_sqlite3.c, res/res_clioriginate.c,
apps/app_voicemail.c: Add missing newlines to CLI logging
........ Merged revisions 361471 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361472 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-06 16:33 +0000 [r361429] Paul Belanger <paul.belanger@polybeacon.com>
* bridges/bridge_builtin_features.c, /, funcs/func_sysinfo.c,
bridges/bridge_multiplexed.c: Multiple revisions 361403,361412
........ r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri,
06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ r361412
| pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2
lines Fix typo in svn:keywords ........ Merged revisions
361403,361412 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361422 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-06 15:50 +0000 [r361382] Russell Bryant <russell@russellbryant.com>
* /, configs/rpt.conf.sample (removed),
configs/usbradio.conf.sample (removed), apps/rpt_flow.pdf
(removed): Remove a few more files related to chan_usbradio and
app_rpt. ........ Merged revisions 361380 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361381 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-06 14:02 +0000 [r361334] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Fix a typo in the warning messages for an
ignored media stream Added a '\n' to the warning messages when we
ignore a media stream due to the port number being '0'. (closes
issue ASTERISK-19646) Reported by: Badalian Vyacheslav ........
Merged revisions 361332 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361333 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-06 13:32 +0000 [r361331] Kinsey Moore <kmoore@digium.com>
* apps/app_dial.c, /: Remove unnecessary error message in
app_dial.c The error message for failure to stop autoservice
after a gosub or macro call during a dial was removed for macro
while Asterisk 1.4 was still being actively developed. The
corresponding gosub error message was never removed. (closes
issue ASTERISK-19551) ........ Merged revisions 361329 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361330 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-05 17:22 +0000 [r361092-361279] Jonathan Rose <jrose@digium.com>
* /, apps/app_meetme.c: Fix MusicOnHold in MeetMe so that it always
uses the class if it's been defined There were a few instances of
restarting music on hold in meetme that would cause Asterisk to
revert to the default class of music on hold for no adequate
reason. Review: https://reviewboard.asterisk.org/r/1844/ ........
Merged revisions 361269 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361270 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, addons/ooh323cDriver.c: Fix some stuff involving calls to
memcpy and memset The important parts of the patch were already
applied through other updates. (closes issue ASTERISK-19445)
Reported by: Makoto Dei Patches: memset-memcpy-length.patch
uploaded by Makoto Dei (license 5027) ........ Merged revisions
361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 361211 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, funcs/func_devstate.c: Make 'help devstate change' display
properly (get rid of excess comma) (closes issue ASTERISK-19444)
Reported by: Makoto Dei Patches:
devstate-change-usage-truncate.patch uploaded by Makoto Dei
(license 5027) ........ Merged revisions 361201 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361208 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/channel.c, pbx/pbx_loopback.c, addons/chan_ooh323.c, /,
channels/chan_sip.c, main/app.c, pbx/pbx_realtime.c,
apps/app_externalivr.c, channels/chan_iax2.c,
res/res_fax_spandsp.c, apps/app_milliwatt.c: Replace GNU
old-style field designator extensions to fix clang warnings
(issue ASTERISK-19540) Reported by: Makoto Dei Patches:
clang-gnu-designator.patch uploaded by Makoto Dei (license 5027)
........ Also add from the patch the portion in res_fax_spandsp
that didn't apply to 1.8 Merged revisions 361142 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue
ASTERISK-19540) ........ Merged revisions 361143 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_meetme.c: Make the MeetMeAdmin N command (mute all
nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported
by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/
........ Merged revisions 361090 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361091 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-03 20:14 +0000 [r361042] Kinsey Moore <kmoore@digium.com>
* /, apps/app_transfer.c: Fix the display of documentation for
Transfer This came up while fixing documentation generation for
many other cases where the argument separator was not being
displayed properly. Now that it is displayed properly, it shows
up in the wrong place for Transfer since the '/' is only required
if Tech is present. (related to issue ASTERISK-18168) ........
Merged revisions 361040 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 361041 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-04-03 20:03 +0000 [r361038-361039] Mark Murawki <markm@intellasoft.net>
* include/asterisk/manager.h: Fix dev-mode compiler warning about
gnu_printf (related to ASTERISK-19575)
* main/channel.c, main/manager.c, main/utils.c,
include/asterisk/channel.h, include/asterisk/strings.h, CHANGES,
include/asterisk/manager.h: Allow the Hangup manager action to
match channels by regex * Hangup now can take a regular
expression as the Channel option. If you want to hangup multiple
channels, use /regex/ as the Channel option. Existing behavior to
hanging up a single channel is unchanged, but if you pass a
regex, the manager will send you a list of channels back that
were hung up. (closes issue ASTERISK-19575) Reported by: Mark
Murawski Tested by: Mark Murawski
2012-04-02 22:27 +0000 [r360994] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Stop sending out RTCP if RTP is inactive
This change prevents Asterisk from sending RTCP receiver reports
during a remote bridge since it is no longer receiving media and
should not be reporting anything. (related to ASTERISK-19366)
........ Merged revisions 360987 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360993 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-30 21:38 +0000 [r360935] Richard Mudgett <rmudgett@digium.com>
* /, main/logger.c: Fix logger deadlock on Asterisk shutdown. The
logger_thread() had an exit path that failed to release the
logmsgs list lock. * Make logger_thread() exit path unlock the
logmsgs list lock. * Made ast_log() not queue any messages to the
logmsgs list if the close_logger_thread flag is set. (issue
ASTERISK-19463) Reported by: Matt Jordan ........ Merged
revisions 360933 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360934 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-29 23:36 +0000 [r360872-360886] Mark Michelson <mmichelson@digium.com>
* /, main/features.c: Fix potential race condition during call
pickup. Prior to this patch, a connected line update was queued
during call pickup and then an answer frame was queued. The
original caller would presumably then have his connected line
updated and then the call would be answered. In actuality, the
answer frame was not how the call ended up being answered.
Rather, an odd section in app_dial that checks if the called
channel's state is up. The result is that the order of the
connected line update and the answer were variable. In most
cases, this wasn't actually a bad thing. However, if the 'I'
option was passed to dial, the connected line update would be
inhibited. The fix is to queued the connected line after the
answer frame is queued. This way the race in app_dial is between
two conditions resulting in an answer. This way the connected
line update occurs after the answer every time. (closes issue
ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas
Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by
Mark Michelson (license 5049) ........ Merged revisions 360884
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 360885 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c: Improve accuracy of identifying
information sent in dialog-info SIP NOTIFY requests. This change
makes use of connected party information in addition to caller ID
in order to populate local and remote XML elements in the
dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by:
Maciej Krajewski Tested by: Maciej Krajewski Patches:
local_remote_hint2.diff uploaded by Mark Michelson (license 5049)
........ Merged revisions 360862 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360863 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-29 21:57 +0000 [r360827] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/astobj2.h, main/astobj2.c: Misc changes to make
astobj2 enhancement diffs easier to follow. * Rename astobj2 API
parameter funcname to func. * Rename astobj2 API iterator
parameter to iter. * Update some documentation for OBJ_MULTIPLE.
2012-03-29 20:01 +0000 [r360785-360787] Jonathan Rose <jrose@digium.com>
* include/asterisk/logger.h, main/dial.c, main/pbx.c,
include/asterisk/bridging.h, main/features.c, main/logger.c,
CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample:
Introducing the log message unique call identifiers feature Log
messages will now display a call number that they are tied to
(ordered for calls based on when they started). This feature is
made to be minimally invasive without requiring changes to many
of the existing log messages. These IDs won't show up for verbose
messages on CLI (but they will in log files) This is currently in
phase II of production, see more about this feature on the wiki
--
https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging
Review: https://reviewboard.asterisk.org/r/1823/
* include/asterisk/logger.h, main/dial.c, main/pbx.c, /,
include/asterisk/bridging.h, main/features.c, main/logger.c,
CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample:
undoing 360785 due to merging mistake
* include/asterisk/logger.h, main/dial.c, main/pbx.c, /,
include/asterisk/bridging.h, main/features.c, main/logger.c,
CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample:
Introducing the log message unique call identifiers feature Log
messages will now display a call number that they are tied to
(ordered for calls based on when they started). This feature is
made to be minimally invasive without requiring changes to many
of the existing log messages. These IDs won't show up for verbose
messages on CLI (but they will in log files) This is currently in
phase II of production, see more about this feature on the wiki
--
https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging
Review: https://reviewboard.asterisk.org/r/1823/
2012-03-28 19:39 +0000 [r360724] Terry Wilson <twilson@digium.com>
* channels/chan_jingle.c, addons/chan_ooh323.c,
cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c,
channels/chan_gtalk.c, apps/confbridge/conf_config_parser.c: Fix
setting CDR variables in the hangup extension A previous CDR fix
for setting CDR variables during a bridge via custom dialplan
features broke setting CDR variables in the hangup extension.
This patch fixes the issue. Review:
https://reviewboard.asterisk.org/r/1794/ ........ Merged
revisions 358978 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358989 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-27 18:44 +0000 [r360673] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Make a debug message regarding
subscription changes more accurate. I was getting confused during
some testing why Asterisk was saying that a subscription was
being added when it was clearly being removed. This fixes that
confusion. ........ Merged revisions 360625 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360672 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-27 17:13 +0000 [r360626-360627] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c:
Add global ao2 array container. Global ao2 objects must always
exist after initialization because there is no access control to
obtain another reference to the global object. It is expected
that module configuration could use these new API calls to
replace an active configuration parameter object with an updated
configuration parameter object. With these new API calls, the
global object could be replaced, removed, or referenced without
the risk of someone using a stale global object pointer. Review:
https://reviewboard.asterisk.org/r/1824/
* main/astobj2.c: Attempt to be more helpful when using a bad ao2
object pointer.
2012-03-27 14:43 +0000 [r360576] Jonathan Rose <jrose@digium.com>
* /, configure: Updates config with bootstrap where I changed
configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon
Clark ........ Merged revisions 360574 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360575 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-26 21:22 +0000 [r360536] Paul Belanger <paul.belanger@polybeacon.com>
* main/dnsmgr.c, /: Convert ast_verb() to ast_debug() and increase
log level Rather then flood the CLI with verbose messages, we've
changed the level to debug. This will help keep the CLI clean.
2012-03-26 19:49 +0000 [r360490] Jonathan Rose <jrose@digium.com>
* /, configure.ac: Fix BETTER_BACKTRACES library detection for
Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon
Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman
Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch
uploaded by Bryon Clark (license 6157) ........ Merged revisions
360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 360489 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-24 23:49 +0000 [r360359-360415] Russell Bryant <russell@russellbryant.com>
* funcs/func_curl.c, /: func_curl: Fix leak of an ast_str in error
handling code path. ........ Merged revisions 360413 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360414 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/chan_iax2.c: chan_iax2: Use OBJ_NODATA to be a bit more
explicit. This is just a minor code cleanup change. These uses of
ao2_callback() would never return anything since the callbacks
always returned 0. However, be more explicit that no returned
results are wanted by specifying OBJ_NODATA.
* /, apps/app_page.c: app_page: Fix a memory leak on every Page().
dial_list is a dynamically allocated array that is allocated at
the beginning of Page() based on how many devices will be dialed.
This was never being freed. ........ Merged revisions 360363 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360364 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_jack.c: app_jack: fix datastore memory leak in error
handling path. ........ Merged revisions 360360 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360361 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/ast_expr2.h, res/ael/ael.tab.c, main/ast_expr2.y,
main/ast_expr2f.c, res/ael/ael_lex.c, res/ael/ael.tab.h,
main/ast_expr2.c: Multiple revisions 360356-360357 ........
r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012)
| 6 lines expression parser: Fix (theoretical) memory leak. Fix a
memory leak that is very unlikely to actually happen. If a
malloc() succeeded, but the following strdup() failed, the memory
from the original malloc() would be leaked. ........ r360357 |
russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines
Rebuild parsers. This is needed to include the last fix to
main/ast_expr2.y. The changes look much bigger as this
regeneration of the code was done with newer versions of flex and
bison. ........ Merged revisions 360356-360357 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360358 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-24 00:40 +0000 [r360264-360311] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /, channels/sig_pri.c: Make number not available
presentation also set screening to network provided. Q.951
indicates that when the presentation indicator is "Number not
available due to interworking" for a number then the screening
indicator field should be "Network provided". * Made
ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE
when the presentation is "Number not available due to
interworking". This fix makes Asterisk consistent and it also
makes it consistent with earlier branches as far as this
presentation value is concerned. * Made pri_to_ast_presentation()
and ast_to_pri_presentation() conversions handle the "Number not
available due to interworking" case better in sig_pri.c. This
change is possible because the minimum required libpri version
(v1.4.11) has the necessary defines in libpri.h. ........ Merged
revisions 360309 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360310 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c: Add missing initialization of
update_redirecting in chan_sip.c ........ Merged revisions 360262
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 360263 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-22 21:25 +0000 [r360227] Jonathan Rose <jrose@digium.com>
* apps/app_dial.c, include/asterisk/utils.h, main/features.c,
main/utils.c, CHANGES, apps/app_queue.c: Adds F option to Bridge
application Similar to dial and queue F option. (Closes issue
ASTERISK-19282) Reported by: To Patches: bridge_f-v3.diff
uploaded by To (license 6347) Review:
https://reviewboard.asterisk.org/r/1825/
2012-03-22 19:51 +0000 [r360190] Kinsey Moore <kmoore@digium.com>
* main/udptl.c, main/stdtime/test.c, main/autoservice.c,
main/rtp_engine.c, main/frame.c, main/fskmodem_float.c,
main/sha1.c, main/say.c, main/ecdisa.h, main/utils.c,
main/devicestate.c, main/taskprocessor.c, main/indications.c,
main/enum.c, main/config.c, main/loader.c, main/term.c,
main/cli.c, main/io.c, main/ulaw.c, main/channel.c, main/dial.c,
main/manager.c, main/tdd.c, main/strcompat.c, main/plc.c,
main/features.c, main/logger.c, main/fskmodem_int.c, main/app.c,
main/stdtime/localtime.c, main/image.c, main/dns.c,
main/message.c, main/md5.c, main/sched.c, main/lock.c,
main/pbx.c, main/dnsmgr.c, main/slinfactory.c, main/translate.c,
main/jitterbuf.c, main/cel.c, main/chanvars.c, main/netsock.c,
main/srv.c, main/privacy.c, main/fixedjitterbuf.c, main/file.c,
main/callerid.c, main/event.c, main/astmm.c, main/audiohook.c,
main/cygload.c, main/fixedjitterbuf.h, main/asterisk.c,
main/xmldoc.c, main/dsp.c, main/timing.c: Kill off red blobs in
most of main/* Everything still compiled after making these
changes, so I assume these whitespace-only changes didn't break
anything (and shouldn't have).
2012-03-21 14:55 +0000 [r360140] Jonathan Rose <jrose@digium.com>
* /, contrib/scripts/install_prereq: Update install_prereq script
to include missing GSM library for debian amd move SQLite3.
(closes issue ASTERISK-19367) Reported by: Andrew Latham Patches:
debian_install_prereq.diff uploaded by Andrew Latham (license
5985) ........ Merged revisions 360138 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360139 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-21 14:47 +0000 [r360137] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* /, configure, configure.ac: Also detect gmime 2.6 Also detect
gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen
(License #5035) <tzafrir.cohen@xorcom.com> ........ Merged
revisions 360087 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360098 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-21 13:31 +0000 [r360089] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Ensure Asterisk sends a BYE when pending
on the final response to a re-INVITE When Asterisk detects a
hangup and cannot send a BYE due to a pending INVITE, it sets the
pendingbye flag and waits for the final response to that INVITE.
When the response is received, it transmits the BYE. If, however,
that INVITE request is a pending re-INVITE, it needs to first
send a CANCEL request to terminate the pending re-INVITE. In that
circumstance, Asterisk was, in some scenarios, clearing the
pendingbye flag after processing the CANCEL request and not
checking for a pending BYE when receiving the final 487 response
to the INVITE. This patch ensures that if the pendingbye flag is
set, it is honored regardless of the nature of the INVITE request
currently in flight. (closes issue ASTERISK-19365) Reported by:
Thomas Arimont Tested by: Thomas Arimont Patches:
bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license
6283) Review: https://reviewboard.asterisk.org/r/1807 ........
Merged revisions 360086 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360088 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-20 20:42 +0000 [r360036] Kinsey Moore <kmoore@digium.com>
* /, apps/app_echo.c: Prevent Echo() from relaying control, null,
and modem frames Echo()'s description states that it echoes
audio, video, and DTMF except for # while it actually echoes any
frame that it receives other than DTMF #. This was causing frame
storms in the test suite in some circumstances where Echo() was
attached to both ends of a pair of local channels and control
frames were being periodically generated. Echo()'s behavior and
description have been modifed so that it only echoes media and
non-# DTMF frames. ........ Merged revisions 360033 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 360034 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-20 18:17 +0000 [r359983] Sean Bright <sean@malleable.com>
* /, UPGRADE.txt, channels/chan_iax2.c, include/asterisk/manager.h:
chan_iax2: Correct spelling of 'Port' header in IAX2 PeerStatus
AMI Events The PeerStatus event for IAX2 channels currently
includes a header named Post which should have been Port. Post
was removed and the AMI version has been updated to 1.3. ........
Merged revisions 359982 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-20 17:31 +0000 [r359942-359981] Richard Mudgett <rmudgett@digium.com>
* main/data.c, main/pbx.c, main/manager.c, /, main/features.c,
include/asterisk/manager.h, main/db.c: Allow AMI action callback
to be reentrant. Fix AMI module reload deadlock regression from
ASTERISK-18479 when it tried to fix the race between calling an
AMI action callback and unregistering that action. Refixes
ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2
object guaranteed that there were no active callbacks that
mattered when ast_manager_unregister() was called. Unfortunately,
this causes the deadlock situation. The patch stops locking the
ao2 object to allow multiple threads to invoke the callback
re-entrantly. There is no way to guarantee a module unload will
not crash because of an active callback. The code attempts to
minimize the chance with the registered flag and the maximum 5
second delay before ast_manager_unregister() returns. The trunk
version of the patch changes the API to fix the race condition
correctly to prevent the module code from unloading from memory
while an action callback is active. * Don't hold the lock while
calling the AMI action callback. (closes issue ASTERISK-19487)
Reported by: Philippe Lindheimer Review:
https://reviewboard.asterisk.org/r/1818/ Review:
https://reviewboard.asterisk.org/r/1820/ ........ Merged
revisions 359979 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359980 from
http://svn.asterisk.org/svn/asterisk/branches/10
* res/res_mutestream.c: Convert MuteAudio documentation to XML. *
Added missing error exits with cause in manager_mutestream(). *
Cleaned up manager_mutestream() and func_mute_write(). * Some
whitespace and comment cleanup.
2012-03-16 21:00 +0000 [r359905] Jonathan Rose <jrose@digium.com>
* /, apps/app_chanspy.c: Prevent chanspy from binding to zombie
channels This patch addresses a bug with chanspy on local
channels which roughly 50% of the time would create a situation
where chanspy can latch onto a zombie channel, keeping the zombie
alive forever and causing the channel doing the spying to never
be able to hang up. (closes issue ASTERISK-19493) Reported by:
lvl Review: https://reviewboard.asterisk.org/r/1819/ ........
Merged revisions 359892 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359898 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-16 20:37 +0000 [r359904] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/app.h, main/app.c: Simplify some code in
ast_app_run_sub(). * Remove unnnecessary const from const char *
const var declaration in the ast_app_run_macro() and
ast_app_run_sub() prototypes. The second const is unnecessary.
2012-03-16 15:38 +0000 [r359857] Mark Michelson <mmichelson@digium.com>
* apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES:
Revert the pre-dial addition. The code may be just fine, but it
had not received a "ship it!" on review board yet.
2012-03-16 08:27 +0000 [r359811] Alec L Davis <sivad.a@paradise.net.nz>
* /, channels/sip/include/sip.h: Missed lastinvite CSeq int to
uint32_t change from Review:
https://reviewboard.asterisk.org/r/1699/ ........ Merged
revisions 359809 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359810 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-15 20:11 +0000 [r359772] Mark Murawki <markm@intellasoft.net>
* main/pbx.c: Fix warning from commit r359705 (predial options for
app_dial)
2012-03-15 19:11 +0000 [r359708] Matthew Jordan <mjordan@digium.com>
* /, main/utils.c: Fix remotely exploitable stack overflow in HTTP
manager There exists a remotely exploitable stack buffer overflow
in HTTP digest authentication handling in Asterisk. The
particular method in question is only utilized by HTTP AMI. When
parsing the digest information, the length of the string is not
checked when it is copied into temporary buffers allocated on the
stack. This patch fixes this behavior by parsing out pre-defined
key/value pairs and avoiding unnecessary copies to the stack.
(closes issue ASTERISK-19542) Reported by: Russell Bryant Tested
by: Matt Jordan ........ Merged revisions 359706 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359707 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-15 18:58 +0000 [r359705] Mark Murawki <markm@intellasoft.net>
* apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES: Add
options PreDial options 'b' and 'B' to app_dial * Added 'b' and
'B' options to Dial. These options will allow you to run
last-minute dialplan on the caller and callee channels while the
Dial application is executing, but before the call is started.
For example you can use the 'b' option to run dialplan on the
callee channel to get the name of the newly created channel right
away. Review: https://reviewboard.asterisk.org/r/1229/ (closes
issue: ASTERISK-19548) Reported by: Mark Murawski Tested by: Mark
Murawski, Stefan Schmidt
2012-03-15 18:55 +0000 [r359704] Matthew Jordan <mjordan@digium.com>
* /, apps/app_milliwatt.c: Fix remotely exploitable stack overrun
in Milliwatt Milliwatt is vulnerable to a remotely exploitable
stack overrun when using the 'o' option. This occurs due to the
milliwatt_generate function not accounting for
AST_FRIENDLY_OFFSET when calculating the maximum number of
samples it can put in the output buffer. This patch resolves this
issue by taking into account AST_FRIENDLY_OFFSET when determining
the maximum number of samples allowed. Note that at no point is
remote code execution possible. The data that is written into the
buffer is the pre-defined Milliwatt data, and not custom data.
(closes issue ASTERISK-19541) Reported by: Russell Bryant Tested
by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by
Russell Bryant (license 6283) Note that this patch was written by
Russell, even though Matt uploaded it ........ Merged revisions
359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
........ Merged revisions 359656 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359694 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-15 18:34 +0000 [r359651] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_sip.c: Remove unused variable ‘srch’ Missed on the
previous commit
2012-03-15 18:32 +0000 [r359644] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, /, apps/app_queue.c: Add missing connected line
macro calls to initial dial for Dial and Queue apps. The
connected line interception macros do not get executed when the
outgoing channel is initially created and that channel's
caller-id is implicitly imported into the incoming channel's
connected line data. If you are using the interception macros,
you would expect that they get run for every change to a
channel's connected line information outside of normal dialplan
execution. Review: https://reviewboard.asterisk.org/r/1817/
........ Merged revisions 359609 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359620 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-15 17:36 +0000 [r359607] Paul Belanger <paul.belanger@polybeacon.com>
* channels/chan_sip.c: Remove some dead code found in
_sip_show_peers() Review:
https://reviewboard.asterisk.org/r/1696/
2012-03-15 00:54 +0000 [r359456-359560] Russell Bryant <russell@russellbryant.com>
* /, channels/chan_iax2.c: chan_iax2: Fix use of uninitialized
sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in
try_transfer() so that the code isn't (potentially) trying to
read from it while uninitialized. ........ Merged revisions
359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 359559 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_gtalk.c: chan_gtalk: Fix potential use of
uninitialized variable. Avoid potential use of idroster in
gtalk_alloc() before it has been initialized. ........ Merged
revisions 359508 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359509 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_chanisavail.c: app_chanisavail: Fix use of
uninitialized variable. Ensure that status is set before it is
used by resetting it during each loop iteration. This could have
resulted in incorrect results from this app. ........ Merged
revisions 359486 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359491 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/udptl.c, /: udptl: Ensure fec[] in udptl_build_packet() is
initialized. Scan results indicated that this array could be used
uninitialized. At a quick look, it looks correct. In any case,
initializing it is a Good Thing (tm). ........ Merged revisions
359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 359458 from
http://svn.asterisk.org/svn/asterisk/branches/10
* include/asterisk/app.h, /: app.h: Always initialize
AST_DECLARE_APP_ARGS(). This patch ensures that the struct
defined by AST_DECLARE_APP_ARGS() is always fully initialized.
I'm not sure if this fixes any real bugs, but it silences a bunch
of warnings from coverity, and is generally a good thing to do
anyway. ........ Merged revisions 359452 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359454 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-14 22:38 +0000 [r359455] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /, channels/chan_agent.c,
include/asterisk/channel.h: Fix deadlock potential with some
ast_indicate/ast_indicate_data calls. Calling
ast_indicate()/ast_indicate_data() with the channel lock held can
result in a deadlock with a local channel because of how local
channels need to avoid deadlock. ........ Merged revisions 359451
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 359453 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-14 18:56 +0000 [r359406] Matthew Jordan <mjordan@digium.com>
* tests/test_jitterbuf.c (added): Add tests for main/jitterbuf.c
This patch adds unit tests for main/jitterbuf.c. This includes
checking for the following: * Nominal insertion and retrieval of
frames * Insertion and retrieval of frames where the frames are
inserted out of order with respect to the previous frame *
Insertion and retrieval of frames where some number of frames
that would occur in the expected sequence are instead dropped *
Insertion and retrieval of frames with an arrival time that does
not occur at the same rate as the surrounding frames *
Resynchronization of the jitter buffer when an inserted frame
breaks the resynchronization threshold * Overfilling of the
jitter buffer For each of the tests, both JB_TYPE_VOICE and
JB_TYPE_CONTROL permutations exist. Review:
https://reviewboard.asterisk.org/r/1815 (issue: ASTERISK-18964)
Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan
2012-03-14 18:12 +0000 [r359360] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/channel_internal.h: Three copies of the file
contents in channel_internal.h are a bit excessive.
2012-03-14 17:48 +0000 [r359359] Matthew Jordan <mjordan@digium.com>
* /, main/jitterbuf.c: Fix incorrect jitter buffer overflow due to
missed resynchronizations When a change in time occurs, such that
the timestamps associated with frames being placed into an
adaptive jitter buffer (implemented in jitterbuf.c) are
significantly different then the previously inserted frames, the
jitter buffer checks to see if it needs to be resynched to the
new time frame. If three consecutive packets break the threshold,
the jitter buffer resynchs itself to the new timestamps. This
currently only occurs when history is calculated, and hence only
on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other
hand, are never passed to the history calculations. Because of
this, if the jump in time is greater then the maximum allowed
length of the jitter buffer, the JB_TYPE_CONTROL frames are
dropped and no resynchronization occurs. Alterntively, if the
overfill logic is not triggered, the JB_TYPE_CONTROL frame will
be placed into the buffer, but with a time reference that is not
applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger
the overflow logic until reads from the jitter buffer reach the
errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL
frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames
are unlikely to occur in multiples, it perform the
resynchronization on any JB_TYPE_CONTROL frame that breaks the
resynch threshold. Note that this only impacts chan_iax2, as
other consumers of the adaptive jitter buffer use the abstract
jitter buffer API, which does not use JB_TYPE_CONTROL frames.
Review: https://reviewboard.asterisk.org/r/1814/ (closes issue
ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt
Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw
(license 5722) ........ Merged revisions 359356 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359358 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-14 17:39 +0000 [r359357] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, main/channel.c, /: Fix Dial m and r options and
forked calls generating warnings for voice frames. When connected
line support was added, the wait_for_answer() variable single
changed its meaning slightly. Unfortunately, the places where
single was used did not necessarily get updated to reflect that
change. Also audio/video frames were sent to all forked calls
when the endpoints were never made compatible. * Don't pass
audio/video media frames when the channels have not been made
compatible. * Added handling of AST_CONTROL_SRCCHANGE to
app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD
because that frame can also pass a requested MOH class. (closes
issue ASTERISK-16901) Reported by: Chris Gentle (closes issue
ASTERISK-17541) Reported by: clint Review:
https://reviewboard.asterisk.org/r/1805/ ........ Merged
revisions 359344 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359355 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-14 14:40 +0000 [r359306] Matthew Jordan <mjordan@digium.com>
* include/asterisk/astobj2.h: Force non-inlining of
ao2_iterator_destroy when TEST_FRAMEWORK is enabled In r357272,
astobj2 was changed to automatically enable REF_DEBUG when the
TEST_FRAMEWORK flag was enabled. Unfortunately, some compilers
(gcc 4.5.1 at least) will attempt to inline ao2_iterator_destroy
in handle_astobj2_test. This by itself is not a problem;
unfortunately, the compiler believes that there is a code path
wherein an object allocated on the stack will be free'd. As
warnings are treated as errors, this prevents compilation of
astobj2. This patch works around that by adding the noinline
attribue to ao2_iterator_destroy, but only if the TEST_FRAMEWORK
flag is enabled. Preventing inlining is only needed for the test
method defined in astobj2, which is also only enabled if
TEST_FRAMEWORK is enabled.
2012-03-14 10:56 +0000 [r359052-359261] Russell Bryant <russell@russellbryant.com>
* include/asterisk/logger.h, /, main/logger.c: Fix bogus
reads/writes of console log levels in asterisk.c This patch
updates the NUMLOGLEVELS define in logger.h to 32, to match the
fact that logger.c implements 32 log levels (because of the
custom log level stuff). asterisk.c uses this define to size an
array of levels per remote console. This array is modified in
ast_console_toggle_loglevel(), which is called by the "logger set
level" CLI command. While the documentation for the CLI command
doesn't make it terribly obvious, you can use this CLI command to
toggle a custom log level on a remote console, as well. However,
doing so led to an invalid array index in asterisk.c. This array
is read from any time a log message is written to a console. So,
all custom log level messages resulted in a bogus read if a
remote console was connected. ........ Merged revisions 359259
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 359260 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid
reads/writes due to incorrect sizeof(). These few places in the
code used sizeof() on h_addr in struct hostent. This is
sizeof(char *). The correct way to get the size of this address
is to use h_length. This error would result in reads/writes of 8
bytes instead of 4 on 64-bit machines. ........ Merged revisions
359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 359212 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/sched.c: Fix inaccurate sizeof() in sched.c. This code
just needed sizeof(int), not sizeof(int *). ........ Merged
revisions 359157 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359162 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, utils/astman.c: Fix incorrect sizeof() in astman. ........
Merged revisions 359116 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359117 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, res/res_crypto.c: Fix incorrect usage of sizeof() in
res_crypto. In this case, just remove the memset(). There was a
redundant memset that is done correctly just 2 lines later.
........ Merged revisions 359110 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359114 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, res/res_adsi.c: Fix broken usage of sizeof() in res_adsi.
........ Merged revisions 359088 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359091 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/features.c: Fix incorrect sizeof() usage in features.c.
This didn't actually result in a bug anywhere, luckily. The only
place where the result of these memcpys was used is in app_dial,
and the only field that it read out of ast_call_feature was the
first one, which is an int, so these memcpys always copied just
enough to avoid a problem. ........ Merged revisions 359069 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359072 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final().
........ Merged revisions 359059 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359060 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/pbx.c, /: Don't use a buffer after it goes out of scope. 's'
is set to 'workspace'. Make sure 'workspace' doesn't go out of
scope while the reference to it via 's' is still used. ........
Merged revisions 359056 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359057 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/chan_usbradio.c (removed), /, channels/xpmr (removed),
build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, configure.ac, makeopts.in,
apps/app_rpt.c (removed): Remove chan_usbradio and app_rpt. These
modules are being maintained outside of the tree and have been
for a long time now, so it doesn't make sense to keep them here.
Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged
revisions 359050 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 359051 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-13 21:24 +0000 [r359011] Terry Wilson <twilson@digium.com>
* include/asterisk/channel_internal.h (added): Add missing
channel_internal.h ...again.
2012-03-13 21:18 +0000 [r358997] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, channels/sig_pri.c: Add ability
for chan_dahdi ISDN to block connected line updates per span.
Added new chan_dahdi.conf colp_send option parameter to block
connected line updates per span. (closes issue ASTERISK-17025)
Reported by: Michael Smith
2012-03-13 20:43 +0000 [r358907-358993] Terry Wilson <twilson@digium.com>
* /, main/features.c: Fix setting CDR variables in the hangup
extension A previous CDR fix for setting CDR variables during a
bridge via custom dialplan features broke setting CDR variables
in the hangup extension. This patch fixes the issue. Review:
https://reviewboard.asterisk.org/r/1794/ ........ Merged
revisions 358978 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358989 from
http://svn.asterisk.org/svn/asterisk/branches/10
* include/asterisk/devicestate.h, /, channels/chan_sip.c,
tests/test_devicestate.c, main/devicestate.c: Make hints for
invalid SIP devices return Unavail, not idle This patch
drastically simplifies the device state aggegation code. The old
method was not only overly complex, but also made it impossible
to return AST_DEVICE_INVALID from the aggregation code. The unit
test update is as a result of fixing that bug. The SIP change
stems from a bug introduced by removing a DNS lookup for
hostname-based SIP channels. (closes issue ASTERISK-16702)
Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged
revisions 358943 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358944 from
http://svn.asterisk.org/svn/asterisk/branches/10
* apps/app_voicemail.c: Fix IMAP storage compilation after
opaquification changes (closes issue ASTERISK-19513)
* channels/chan_unistim.c, main/autoservice.c,
channels/chan_vpb.cc, channels/chan_local.c, main/rtp_engine.c,
res/res_musiconhold.c, bridges/bridge_multiplexed.c,
apps/app_followme.c, main/indications.c, main/cli.c,
main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c,
channels/sig_analog.c, main/manager.c, main/features.c,
apps/app_dumpchan.c, res/res_agi.c, main/app.c,
apps/app_confbridge.c, apps/app_externalivr.c, main/bridging.c,
apps/app_parkandannounce.c, apps/app_dial.c, main/pbx.c,
channels/chan_sip.c, channels/chan_bridge.c,
main/channel_internal_api.c, channels/chan_agent.c,
apps/app_disa.c, include/asterisk/channel.h,
apps/app_talkdetect.c, apps/app_queue.c, apps/app_speech_utils.c,
apps/app_channelredirect.c, main/file.c, res/snmp/agent.c,
apps/app_macro.c, apps/app_stack.c, apps/app_chanspy.c,
apps/app_mixmonitor.c: Finalize ast_channel opaquification
Review: https://reviewboard.asterisk.org/r/1786/
2012-03-13 17:01 +0000 [r358858-358861] Richard Mudgett <rmudgett@digium.com>
* main/channel.c: Fix crash caused by opaquification change
-r356042. The set_format() function was more subtle in how it
modified the struct ast_channel readtrans/writetrans values. *
Fixed ast_activate_generator() conversion correctly. (closes
issue ASTERISK-19434) Reported by: Birger Harzenetter Tested by:
rmudgett
* main/format.c: Use struct copy instead of memcpy().
2012-03-13 08:06 +0000 [r358812] Tilghman Lesher <tilghman@meg.abyt.es>
* res/ael/pval.c, funcs/func_dialplan.c, /, tests/test_gosub.c,
utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c: Enable
macros in 1.8 to find the next highest "h" extension in a
context, like in 1.4. This change restores functionality that was
present in 1.4, when AEL macros were implemented with the Macro
dialplan application. Macros are fraught with functionality
issues, because they consume a large portion of the underlying
application stack. This limits the ability of AEL users to call
many layers of subroutines, an issue which Gosub does not have
(originally tested to 100,000 levels deep). Therefore, starting
in 1.6.0, AEL macros were implemented with Gosub. However, there
were some implicit behaviors of Macro, which were not replicated
at the same time as with the transition to Gosub, one of which is
documented in the related issue. In particular, the "h" extension
is designed to execute not in the Macro context, but in the
topmost calling context. Due to legacy issues with a misapplied
bugfix many years ago, when a macro exited in 1.4, it looks in
all calling contexts, bubbling up from the deepest level until it
finds an "h" extension. Since AEL hides the complexity of the
underlying dialplan logic from the AEL programmer, it's
reasonable to assume that this behavior should not change in the
transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we
break working AEL configurations in the transition to Asterisk
1.8 LTS. This fix is the result, which implements a search for
the "h" extension in all calling Gosub contexts. Fixes
ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff
(License #5003) by Tilghman Lesher (with slight modifications for
1.8) Tested by: Johan Wilfer Review:
https://reviewboard.asterisk.org/r/1776/ ........ Merged
revisions 358810 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358811 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-12 17:01 +0000 [r358766] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c, contrib/unistimLang/ru.po (added),
contrib/unistimLang/ru.po.utf8 (added),
configs/unistim.conf.sample, UPGRADE.txt, CHANGES,
contrib/unistimLang/en.po (added), contrib/unistimLang (added):
Massive changes in chan_unistim channel driver. Include many
fixes in channel driver operation and add additional
functionality: * Added ability to use multiple lines on phone, so
for one device in configuration multiple lines can be defined, it
allows to have multiple calls on one phone, callwaiting and
switching between calls. * Added ability for translation
on-screen menu to multiple languages. Tested on Russian
languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO
8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by
'language' and on-screen menu of phone * Other described in
CHANGES file Testing done by issue tracker users: ibercom,
scsiborg, idarwin, TeknoJuce, c0rnoTa. Tested on production
system by Jonn Taylor (jonnt) using phone models: Nortel i2004,
1120E and 1140E. (closes issue ASTERISK-16890) Review:
https://reviewboard.asterisk.org/r/1243/
2012-03-10 20:06 +0000 [r358730] Joshua Colp <jcolp@digium.com>
* configs/confbridge.conf.sample, main/dial.c, apps/app_page.c,
apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
include/asterisk/dial.h, CHANGES,
apps/confbridge/conf_config_parser.c: Transition app_page to
using app_confbridge internally for the conference bridge portion
of paging. This also adds a new 'announcement' option to
ConfBridge user profiles. Review:
https://reviewboard.asterisk.org/r/1754/
2012-03-08 17:48 +0000 [r358646-358691] Sean Bright <sean@malleable.com>
* apps/app_dial.c, apps/app_directory.c, apps/app_queue.c: Resolve
a few more cases of variable shadowing.
* channels/chan_phone.c, channels/chan_skinny.c,
channels/chan_agent.c, pbx/pbx_lua.c, pbx/pbx_dundi.c,
channels/chan_gtalk.c, pbx/pbx_config.c, channels/chan_oss.c,
apps/confbridge/conf_config_parser.c: Eliminate a bunch of shadow
warnings.
* include/asterisk/linkedlists.h: Add some underscores in a few of
our llist macros to reduce name collisions.
2012-03-08 16:59 +0000 [r358645] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: Make transfer not ignore port information
with SIP. Attempting to transfer with SIP to an address like
1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from
the host string and ignored. This simply keeps chan_sip from
cutting off the port number during these kinds of transfers.
(closes issue ASTERISK-19321) Reported by: Federico Alves Review:
https://reviewboard.asterisk.org/r/1790/diff/#index_header
........ Merged revisions 358643 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358644 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-08 16:21 +0000 [r358609-358622] Sean Bright <sean@malleable.com>
* Makefile, configure, configure.ac, makeopts.in: Add
--enable-dev-mode=strict to configure. Passing -Wshadow to gcc
enables shadow warnings. From the gcc manual: Warn whenever a
local variable or type declaration shadows another variable,
parameter, type, or class member (in C++), or whenever a built-in
function is shadowed. Asterisk will not currently compile with
this option set, but a number of bugs have been discovered by
enabling this flag on specific files. The long-term goal is to
eliminate all of the suspect code that causes this warning to be
emitted.
* Makefile: Whitespace only change to the Makefile
2012-03-07 21:28 +0000 [r358576] Terry Wilson <twilson@digium.com>
* cel/cel_odbc.c, configs/cel_odbc.conf.sample: Handle numeric
columns for eventtype properly in cel_odbc Patch also implements
correct handling of datetime2 and datetimeoffset new datatypes in
SQL Server 2008 and 2008 R2. (closes issue ASTERISK-17548)
Review: https://reviewboard.asterisk.org/r/1160/ Review:
https://reviewboard.asterisk.org/r/1804/
2012-03-07 18:33 +0000 [r358532] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_ss7.c: Change directly setting _softhangup in
sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue
ASTERISK-19372) ........ Merged revisions 358530 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358531 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-07 16:16 +0000 [r358486] Sean Bright <sean@malleable.com>
* /, codecs/codec_dahdi.c: Return g729 and g723.1 frames with the
number of samples set properly. If the wctc4xxp returns more than
a single packet, we need to update the number of samples in the
returned frame accordingly. Acked-by: Shaun Ruffell
<sruffell@digium.com> ........ Merged revisions 358484 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358485 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-07 15:19 +0000 [r358437-358444] Terry Wilson <twilson@digium.com>
* /, configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in
cdr_adaptive_odbc.conf.sample ........ Merged revisions 358438
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 358441 from
http://svn.asterisk.org/svn/asterisk/branches/10
* cel/cel_odbc.c, /, cdr/cdr_adaptive_odbc.c: Add detection for
ODBC WCHAR fields Without detecting these types, cel_odbc blows
up when the character set for the table is utf8. This also wraps
cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR
#ifdef seen in other parts of the code. ........ Merged revisions
358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 358436 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-06 17:47 +0000 [r358262-358379] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Fix ring cadance setup for outgoing
calls on FXS ports. * Fix referencing the wrong variable in
chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for
compiling with -Wshadow and finding this bug. ........ Merged
revisions 358377 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358378 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES:
Add dialtone_detect option for analog incoming calls. For analog
lines, enables Asterisk to use dialtone detection per channel if
an incoming call was hung up before it was answered. If dialtone
is detected, the call is hung up. no: Disabled. (Default) yes:
Look for dialtone for 10000 ms after answer. <number>: Look for
dialtone for the specified number of ms after answer. always:
Look for dialtone for the entire call. Dialtone may return if the
far end hangs up first. dialtone_detect=yes dialtone_detect=5000
dialtone_detect=always (closes issue ASTERISK-19316) Reported by:
Jeremy Pepper Patch by: Jeremy Pepper Tested by: rmudgett,Jeremy
Pepper Review: https://reviewboard.asterisk.org/r/1737/
* /, channels/sig_ss7.c: Drop SS7 call if not connected yet when
INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should
clear a failed call as soon as possible. * Made SS7 hangup a call
immediately if it has not connected yet for
INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate
inband tone. (closes issue ASTERISK-19372) Reported by: Igor
Nikolaev ........ Merged revisions 358278 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358284 from
http://svn.asterisk.org/svn/asterisk/branches/10
* include/asterisk/channel.h: Make usage of
DECLARE_STRINGFIELD_SETTERS_FOR() not look so odd.
* channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c:
Setup DSP when SS7 call is connected or early media is available.
Outgoing SS7 calls fail to detect incoming DTMF so any bridged
channel that requires out-of-band DTMF will not work. * Added
sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
The new call converts conditionaled out unconverted code and
shows that the code really did something useful. * Improved some
chan_dahdi DTMF debug messages to help track DTMF handling.
(closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........
Merged revisions 358260 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358261 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-05 19:06 +0000 [r358216] Jonathan Rose <jrose@digium.com>
* main/manager.c, /: Eliminate double close of file descriptor in
manager.c The process_output function in manager.c attempted to
call fclose and close immediately afterwards. Since fclose
implies close, this resulted in a potential double free on file
descriptors. This patch changes that behavior and also adds error
checking to fclose and close depending on which was deemed
necessary. Also error messages. Thanks to Rosen Iliev for
pointing out the location of the problem. (closes issue
ASTERISK-18453) Reported By: Jaco Kroon Review:
https://reviewboard.asterisk.org/r/1793/ ........ Merged
revisions 358214 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358215 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-05 16:44 +0000 [r358164] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Defer sending the connected line reinvite
if a reinvite is already in progress. (issue ASTERISK-19355)
Reported by: tomaso (closes issue AST-825) ........ Merged
revisions 358162 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358163 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-05 16:00 +0000 [r358117] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx
on Replaces errors Asterisk was not setting pendinginvite in the
upper half of handle_request_invite such that the 4xx was
retransmitted repeatedly even though an ack was received for
every retransmission. (closes issue ASTERISK-19303) Reported by:
Jon Tsiros Patches: fix-19303.patch uploaded by Jeremiah Gowdy
(license 6358) ........ Merged revisions 358115 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358116 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-05 11:20 +0000 [r358082] Sean Bright <sean@malleable.com>
* configs/iax.conf.sample: Tab to spaces and text change.
2012-03-02 23:29 +0000 [r357999-358038] Terry Wilson <twilson@digium.com>
* channels/chan_usbradio.c, /, channels/xpmr/xpmr.c: Fix
unused-but-set-variable warnings All of these were pretty
obviously unused. Some were unused because the code that used
them was #if 0'd. In those cases, I just commented out the
unused-but-set variables. ........ Merged revisions 358029 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358033 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /: Correct some set-but-unused variable warnings in the mISDN
library. (from kpfleming's commit to trunk r356292) ........
Merged revisions 358011 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 358017 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/xpmr/xpmr.c: Make chan_usbradio compile under dev
mode x=++x and x=x=1? Really? ........ Merged revisions 357986
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 357987 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-02 21:06 +0000 [r357942] Kinsey Moore <kmoore@digium.com>
* /, main/ccss.c, tests/test_event.c, main/event.c,
include/asterisk/strings.h: Fix case-sensitivity for
device-specific event subscriptions and CCSS This change fixes
case-sensitivity for device-specific subscriptions such that the
technology identifier is case-insensitive while the remainder of
the device string is still case-sensitive. This should also
preserve the original case of the device string as passed in to
the event system. CCSS is the only feature affected as it is the
only consumer of device-specific event subscriptions. The second
part of this patch addresses similar case-sensitivity issues
within CCSS itself that prevented it from functioning correctly
after the fix to the events system. This adds a unit test to
verify that the event system works as expected. (closes issue
ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/
........ Merged revisions 357940 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357941 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-02 18:38 +0000 [r357896] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /, channels/sig_pri.c: Remove ISDN hold
restriction for non-bridged calls. The check if an ISDN call is
bridged before it could be placed on hold is not necessary and is
overly restrictive. The check was originally done to prevent
problems with call transfers in case a user tried to transfer a
call connected to an application to another call connected to an
application. The ISDN transfer code has not required this
restriction for quite some time because ECT could transfer any
two active calls to each other. * Remove ISDN hold restriction
for calls connected to applications. * Made
ast_waitfordigit_full() ignore AST_CONTROL_HOLD and
AST_CONTROL_UNHOLD instead of generating a warning message.
(closes issue ASTERISK-19388) Reported by: Birger Harzenetter
Tested by: rmudgett ........ Merged revisions 357894 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357895 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-02 16:57 +0000 [r357861] Jonathan Rose <jrose@digium.com>
* apps/app_queue.c: Adds a transfer callee on hangup option (like
with Dial option F) to queues. This should (and does in my
testing) act just like the Dial option of the same name. This
allows a queue member to be transfered to the next priority (no
args), or to a context/extension/priority similar to goto (with
args context^extension^priority) when a caller hangs up on them.
(closes issue ASTERISK-19283) Reported by: To Patches:
queue_f-v3.diff uploaded by To (license 6347) Review:
https://reviewboard.asterisk.org/r/1785/
2012-03-02 16:26 +0000 [r357834] Richard Mudgett <rmudgett@digium.com>
* apps/app_chanspy.c: Remove bad usage of goto in ChanSpy
next_channel().
2012-03-02 16:19 +0000 [r357821] Sean Bright <sean@malleable.com>
* configs/iax.conf.sample: Beef up the IAX2 sample configuration a
bit and fix some formatting issues.
2012-03-02 16:03 +0000 [r357814-357815] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_chanspy.c: Fix channel reference leak in ChanSpy. *
Fix next_channel() channel reference leak in ChanSpy. (closes
issue ASTERISK-19461) Reported by: Irontec Patches:
app_chanspy_iteartor_next_unref.patch (license #6213) patch
uploaded by Irontec (issue ASTERISK-17515) ........ Merged
revisions 357809 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357810 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/chan_usbradio.c: Fix compile error from latest channel
opaquification change.
2012-03-02 16:00 +0000 [r357813] Sean Bright <sean@malleable.com>
* /, channels/chan_iax2.c: The default value for mohinterpret is
the empty string, so when resetting to default values don't
explicitly set the value to "default." ........ Merged revisions
357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 357812 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-03-02 01:33 +0000 [r357774-357775] Mark Michelson <mmichelson@digium.com>
* main/channel.c, /: Fix race condition that can cause important
control frames (such as a hangup) to be missed. This takes two
actions. 1. Move the reading of the alertpipe in __ast_read() to
immediately before the removal of frames from the readq. This
means we won't do something silly like read from the alertpipe,
then ignore the fact that there's a frame to get from the readq
since channel's fdno is the AST_TIMING_FD. 2. When
ast_settimeout() sets the rate to 0 and the timingfunc to NULL,
if the channel's fdno is the AST_TIMING_FD, then set the fdno to
-1. This is because if the rate is 0 and the timingfunc is NULL,
it means that the channel's timing fd is being invalidated, so
any pending reads should not occur. This may actually solve more
issues than the referenced one below, but it's not known at this
time for sure. (closes issue ASTERISK-19223) reported by
Frank-Michael Wittig Review:
https://reviewboard.asterisk.org/r/1779 ........ Merged revisions
357761 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 357762 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/chan_dahdi.c: Fix compilation error due to typo during
channel opaquification.
s/ast_channel_fd_set/ast_channel_internal_fd_set/g
2012-03-01 22:09 +0000 [r357721] Terry Wilson <twilson@digium.com>
* channels/chan_unistim.c, apps/app_dahdibarge.c,
main/autoservice.c, addons/chan_ooh323.c, channels/chan_vpb.cc,
apps/app_meetme.c, channels/console_video.c,
channels/chan_gtalk.c, channels/chan_iax2.c, main/cli.c,
main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c,
channels/sig_analog.c, channels/chan_skinny.c, main/features.c,
apps/app_dumpchan.c, channels/sig_ss7.c, channels/chan_mgcp.c,
main/pbx.c, channels/chan_sip.c, main/channel_internal_api.c,
channels/chan_agent.c, apps/app_dahdiras.c,
include/asterisk/channel.h, apps/app_queue.c, channels/sig_pri.c,
channels/chan_jingle.c, channels/chan_misdn.c, apps/app_flash.c,
funcs/func_channel.c, apps/app_directed_pickup.c, main/file.c,
channels/chan_h323.c, res/snmp/agent.c, main/dsp.c: Opaquify
ast_channel typedefs, fd arrays, and softhangup flag Review:
https://reviewboard.asterisk.org/r/1784/
2012-03-01 14:22 +0000 [r357673] Kinsey Moore <kmoore@digium.com>
* /, main/acl.c: Prevent outbound SIP NOTIFY packets from
displaying a port of 0 In the change from 1.6.2 to 1.8,
ast_sockaddr was introduced which changed the behavior of
ast_find_ourip such that port number was wiped out. This caused
the port in internip (which is used for Contact and Call-ID on
NOTIFYs) to be 0. This change causes ast_find_ourip to be
port-preserving again. (closes issue ASTERISK-19430) ........
Merged revisions 357665 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357667 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-29 20:41 +0000 [r357621] Walter Doekes <walter+asterisk@wjd.nu>
* /, main/utils.c, include/asterisk/stringfields.h: Update
stringfield documentation for removed second va_list in favor of
va_copy. In r320946, the second va_list that was passed to
ast_string_field_build_va and friends, was removed. This patch
updates the documentation to reflect that. ........ Merged
revisions 357620 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-29 20:31 +0000 [r357610] Sean Bright <sean@malleable.com>
* res/res_agi.c, CHANGES: Add IPv6 support to FastAGI. Review:
https://reviewboard.asterisk.org/r/1774/ Reviewed by: Simon
Perreault, Mark Michelson
2012-02-29 19:48 +0000 [r357577] Walter Doekes <walter+asterisk@wjd.nu>
* apps/app_dial.c, /: Fix copying of CDR(accountcode) to local
channels. In r203638, during the addition of the Channel Event
Logging, in mid-2009, this got broken in trunk and ended up in
asterisk 1.8 and higher. This fixes so the CDR(accountcode) from
the calling channel is available to dialed channels again as well
as showing up properly in the CDR's. (closes issue
ASTERISK-19384) Reported by: jamicque Patches: accountcode.patch
(License #6033) by jamicque Review:
https://reviewboard.asterisk.org/r/1775/ Reviewed by: Richard
Mudgett ........ Merged revisions 357575 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357576 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-29 16:52 +0000 [r357542] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, addons/chan_ooh323.c,
funcs/func_strings.c, channels/console_video.c,
apps/app_alarmreceiver.c, channels/chan_iax2.c, main/cli.c,
channels/chan_dahdi.c, channels/sig_analog.c,
channels/chan_skinny.c, apps/app_dumpchan.c, main/features.c,
apps/app_amd.c, channels/sig_ss7.c, apps/app_dial.c, main/pbx.c,
include/asterisk/utils.h, funcs/func_timeout.c,
apps/app_privacy.c, apps/app_fax.c, channels/chan_agent.c,
apps/app_disa.c, include/asterisk/channel.h,
apps/app_talkdetect.c, main/cel.c, channels/chan_misdn.c,
apps/app_macro.c, apps/app_zapateller.c, apps/app_mixmonitor.c,
apps/app_voicemail.c, channels/chan_unistim.c,
tests/test_substitution.c, channels/chan_vpb.cc,
apps/app_meetme.c, main/ccss.c, apps/app_readexten.c,
channels/chan_gtalk.c, main/autochan.c, apps/app_followme.c,
main/cdr.c, main/channel.c, main/dial.c, channels/chan_phone.c,
apps/app_osplookup.c, apps/app_setcallerid.c, main/manager.c,
bridges/bridge_builtin_features.c, apps/app_minivm.c,
res/res_agi.c, main/app.c, apps/app_confbridge.c, apps/app_rpt.c,
main/message.c, channels/chan_mgcp.c, apps/app_parkandannounce.c,
apps/app_while.c, funcs/func_dialplan.c, channels/chan_sip.c,
res/res_fax.c, main/channel_internal_api.c, pbx/pbx_lua.c,
channels/chan_console.c, channels/sig_pri.c, apps/app_queue.c,
channels/chan_oss.c, channels/chan_jingle.c,
channels/chan_usbradio.c, funcs/func_blacklist.c,
main/abstract_jb.c, channels/chan_h323.c, main/file.c,
res/snmp/agent.c, apps/app_sms.c, apps/app_stack.c,
funcs/func_callerid.c: Opaquify ast_channel structs and lists
Review: https://reviewboard.asterisk.org/r/1773/
2012-02-28 22:31 +0000 [r357460-357503] Jonathan Rose <jrose@digium.com>
* /, configs/sip.conf.sample, UPGRADE-1.8.txt: Adding transport=udp
to sample sip.conf - Also changes version of Asterisk 1.8 in
UPGRADE (issue ASTERISK-19352) Reported by: jamicque Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by
Michael L. Young (license 5026) ........ Merged revisions 357490
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 357497 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, cdr/cdr_adaptive_odbc.c: Add additional character type types
to supported data types for cdr_adaptive_odbc The reporter was
uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so
this patch adds those along with some other character types to
the list of types cdr_adaptive_odbc will work using the varchar
conditions. The problem wasn't really UTF8 characters as much as
it was a failure to respond to the exact type that was
declared/in use on that database. (closes issue ASTERISK-19334)
Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch
uploaded by Igor Nikolaev (license 6236) ........ Merged
revisions 357455 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357458 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-28 21:26 +0000 [r357436] Tilghman Lesher <tilghman@meg.abyt.es>
* /, apps/app_stack.c: Correctly reset the dialplan priority. When
the stack frame is allocated, we save the address to which we
should return, when the Gosub returns. However, if we just want
to restore the priority, then we need to subtract 1 before
setting it. Otherwise, when a Gosub goes to a nonexistent
address, it will skip a priority in the dialplan. This is because
when we return from an application, the PBX increments the
priority for us. ........ Merged revisions 357416 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357421 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-28 21:01 +0000 [r357409] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_pri.c: Use more reasonable cause code when
rejecting incoming call waiting calls. (closes issue
ASTERISK-19397) Reported by: Birger Harzenetter Patches:
nochannel-cause.patch (license #5870) patch uploaded by Birger
Harzenetter ........ Merged revisions 357407 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357408 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-28 20:43 +0000 [r357406] Jonathan Rose <jrose@digium.com>
* /, UPGRADE-10.txt: revision 357386 -- oops, accidentally made it
10.3 to 10.4 instead of 10.2 to 10.3 (issue ASTERISK-19352)
reported by: jamicque ........ Merged revisions 357405 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-28 20:34 +0000 [r357404] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, res/res_musiconhold.c, apps/app_queue.c: Fix
REF_DEBUG compile errors.
2012-02-28 20:33 +0000 [r357358-357403] Jonathan Rose <jrose@digium.com>
* /, UPGRADE-10.txt, UPGRADE-1.8.txt: Moves UPGRADE.txt notes from
r357356 to a new section specific to 1.8.12 (issue
ASTERISK-19352) reported by: jamicque ........ Merged revisions
357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 357400 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, UPGRADE-1.8.txt: Adds UPGRADE.txt notes to r357266 indicating
changes to transport option (issue ASTERISK-19352) Reported by:
jamicque ........ Merged revisions 357356 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357357 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-28 19:55 +0000 [r357355] Sean Bright <sean@malleable.com>
* include/asterisk/netsock2.h: Documentation update. There is no
AST_SOCKADDR_UNSPEC.
2012-02-28 19:37 +0000 [r357354] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_page.c: Remove dupliate 'i' option table entry in
app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei
Patches: app_page-duplicate-i-option.patch (license #5027) patch
uploaded by Makoto Dei ........ Merged revisions 357352 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357353 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-28 18:52 +0000 [r357319] Mark Michelson <mmichelson@digium.com>
* /, channels/sip/security_events.c: Add a security event for the
case where fake authentication challenge is sent. ........ Merged
revisions 357318 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-28 18:46 +0000 [r357317] Richard Mudgett <rmudgett@digium.com>
* main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h:
Convert struct ast_tcptls_session_instance to finally use the ao2
object lock.
2012-02-28 18:23 +0000 [r357288] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: Changes transport option in sip.conf so
that using multiple instances doesn't stack. Prior to this patch,
Using "transport=" multiple times would cause them to add to one
another like allow/deny. This patch changes that behavior to
simply use the transport option specified last. Also, if no
transport option is applied now, the default will automatically
be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by
Michael L. Young (license 5026)
issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes
(license 5674) Review:
https://reviewboard.asterisk.org/r/1745/diff/#index_header
........ Merged revisions 357266 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357271 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-28 18:15 +0000 [r357272] Richard Mudgett <rmudgett@digium.com>
* main/format.c, main/format_cap.c, include/asterisk/astobj2.h,
include/asterisk/lock.h, main/astobj2.c: Astobj2 locking
enhancement. Add the ability to specify what kind of locking an
ao2 object has when it is allocated. The locking could be one of:
MUTEX, RWLOCK, or none. New API: ao2_t_alloc_options()
ao2_alloc_options() ao2_t_container_alloc_options()
ao2_container_alloc_options() ao2_rdlock() ao2_wrlock()
ao2_tryrdlock() ao2_trywrlock() The OBJ_NOLOCK and
AO2_ITERATOR_DONTLOCK flags have a slight meaning change. They no
longer mean that the object is protected by an external
mechanism. They mean the lock associated with the object has
already been manually obtained by one of the ao2_lock calls. This
change is necessary for RWLOCK support since they are not
reentrant. Also an operation on an ao2 container may require
promoting a read lock to a write lock by releasing the already
held read lock to re-acquire as a write lock. Replaced API calls:
ao2_t_link_nolock() ao2_link_nolock() ao2_t_unlink_nolock()
ao2_unlink_nolock() with the respective ao2_t_link_flags()
ao2_link_flags() ao2_t_unlink_flags() ao2_unlink_flags() API
calls to be more flexible and to allow an anticipated enhancement
to control linking duplicate objects into a container. The
changes to format.c and format_cap.c are taking advantange of the
new ao2 locking options to simplify the use of the format
capabilities containers. Review:
https://reviewboard.asterisk.org/r/1554/
2012-02-28 14:47 +0000 [r357178-357214] Kevin P. Fleming <kpfleming@digium.com>
* /, Makefile.rules: Make COMPILE_DOUBLE magic actually work. The
build system has some special magic to ensure that if Asterisk is
built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the
source is still compiled with the optimizer enabled (even though
the result will be thrown away), because the compiler is able to
find a great deal of coding errors and bugs as a result of
running its optimizers. Unfortunately at some point this mode got
broken, and the 'throwaway' compile of the code was no longer
done with the optimizer enabled. This patch corrects that
problem. ........ Merged revisions 357212 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 357213 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/astobj2.c: Trailing whitespace cleanup.
2012-02-28 00:42 +0000 [r357096-357145] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c:
Add ability to clone ao2 containers. Occasionally there is a need
to put all objects in one container also into another container.
Some reasons you might need to do this: 1) You need to
reconfigure a container. You would do this by creating a new
container with the new configuration and ao2_container_dup the
old container into it. Then replace the old container with the
new. Then destroy the old container. 2) You need the contents of
a container to remain stable while operating on all of the
objects. You would do this by creating a cloned container of the
original with ao2_container_clone. The cloned container is a
snapshot of the objects at the time of the cloning. When done,
just destroy the cloned container. Review:
https://reviewboard.asterisk.org/r/1746/
* main/channel.c: Fix ast_channel allocation init setting priority
to -1 instead of 1. * Fix opaquification conversion error.
(closes issue ASTERISK-19424) Reported by: Jeremy Pepper Patches:
asterisk-19424-initialize_priority_regression.diff (license
#5026) patch uploaded by Michael L. Young
* main/channel.c, /: Fix callerid of Originated calls. Thanks to
Matt Riddell for tracking this down. (closes issue
ASTERISK-19385) Reported by: ornix ........ Merged revisions
357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 357095 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-27 19:55 +0000 [r357051] Jonathan Rose <jrose@digium.com>
* include/asterisk/res_odbc.h, res/res_odbc.c: Converts locking for
odbc containers from ast_mutex_lock to ao2_locks.
2012-02-27 17:03 +0000 [r357014] Sean Bright <sean@malleable.com>
* channels/chan_iax2.c, main/netsock.c: Address comments from Mark
Michelson
2012-02-27 16:50 +0000 [r357013] Kinsey Moore <kmoore@digium.com>
* apps/app_dial.c, main/channel.c, include/asterisk/app.h,
main/dial.c, main/rtp_engine.c, main/ccss.c, main/features.c,
UPGRADE.txt, main/app.c, include/asterisk/channel.h,
configs/ccss.conf.sample, apps/app_followme.c, apps/app_queue.c,
include/asterisk/ccss.h: Deprecated macro usage for connected
line, redirecting, and CCSS This commit adds GoSub alternatives
to connected line, redirecting, and CCSS macro hooks so that
macro can finally be deprecated. This also adds deprecation
warnings for those features when used and in documentation.
Review: https://reviewboard.asterisk.org/r/1760/ (closes issue
SWP-4256)
2012-02-27 16:31 +0000 [r357005] Sean Bright <sean@malleable.com>
* include/asterisk/netsock.h, channels/chan_iax2.c, main/netsock.c:
Convert netsock.h over to use ast_sockaddrs rather than
sockaddr_in and update chan_iax2 to pass in the correct types.
chan_iax2 is the only consumer for the various ast_netsock_*
functions in trunk at this point, so this feels like a safe
change to make.
2012-02-27 16:24 +0000 [r356987] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
channels/sip/include/sip.h: Adds an option to sip.conf that
prevents diversion headers from being added. send_diversion=no
will prevent Diversion headers from being added to SIP requests.
This doesn't prevent Diversion from being added with dialplan
such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported
by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/
2012-02-27 16:12 +0000 [r356966] Sean Bright <sean@malleable.com>
* channels/chan_iax2.c: There isn't much point in saving off and
restoring a value that we never use again.
2012-02-27 16:08 +0000 [r356965] Terry Wilson <twilson@digium.com>
* /, main/features.c: Copy CDR variables when set during a bridge
This patch makes sure amaflags, accountcode, and userfield get
copied to the bridge CDR when set during a bridge (like via a
custom feature). (closes issue ASTERISK-16990) Review:
https://reviewboard.asterisk.org/r/1721/ ........ Merged
revisions 356963 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 356964 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-27 15:35 +0000 [r356962] Jonathan Rose <jrose@digium.com>
* /, res/res_odbc.c: Remove possible segfaults from res_odbc by
adding locks around usage of odbc handle (closes issue
ASTERISK-19011) Reported by: Walter Doekes Patches:
issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch
uploaded by Walter Doekes (license 5674) review:
https://reviewboard.asterisk.org/r/1719/ review:
https://reviewboard.asterisk.org/r/1622/ ........ Merged
revisions 356917 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 356961 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-27 14:57 +0000 [r356881-356916] Sean Bright <sean@malleable.com>
* include/asterisk/netsock.h, main/netsock.c: Make
ast_netsock_set_qos() delegate to ast_set_qos().
* include/asterisk/netsock.h: Correct typo in deprecation comment.
* channels/chan_unistim.c, main/udptl.c, channels/chan_skinny.c,
include/asterisk/netsock.h, pbx/pbx_dundi.c,
channels/chan_mgcp.c: Prefer ast_set_qos() over
ast_netsock_set_qos()
* main/netsock.c: Remove trailing whitespace
2012-02-26 18:25 +0000 [r356848] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c: Add
support change gatekeeper mode or ip per ooh323 reload command
(issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches:
change_gk_on_reload-1.patch (License #5415)
2012-02-25 17:22 +0000 [r356799] Matthew Jordan <mjordan@digium.com>
* /, apps/app_voicemail.c: Fix crash in app_voicemail during
close_mailbox In r354890, a memory leak in app_voicemail was
fixed by properly disposing of the allocated heard/deleted
pointers. However, there are situations, particularly when no
messages are found in a folder, where these pointers are not
allocated and not NULL. In that case, an invalid free would be
attempted, which could crash app_voicemail. As there are a number
of code paths where this could occur, this patch uses the number
of messages detected in the folder before it attempts to free the
pointers. This resolves the crash detected in the Asterisk Test
Suite's check_voicemail_nominal test. ........ Merged revisions
356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 356798 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-24 23:40 +0000 [r356697-356765] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/astobj2.h: astobj2.h comment tweaks.
* include/asterisk/astobj2.h, main/astobj2.c: astobj2.h
documentation updates.
* /, channels/chan_sip.c, include/asterisk/tcptls.h,
channels/sip/include/sip.h: Fix worker thread resource leak in
SIP TCP/TLS. The SIP TCP/TLS worker threads were created joinable
but noone could join them if they died on their own. * Fix the
SIP TCP/TLS worker threads to not be created joinable. *
_sip_tcp_helper_thread() only needs one parameter since the pvt
parameter is only passed in as NULL and never used. (closes issue
ASTERISK-19203) Reported by: Steve Davies Review:
https://reviewboard.asterisk.org/r/1714/ ........ Merged
revisions 356677 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 356690 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-24 17:43 +0000 [r356606-356652] Matthew Jordan <mjordan@digium.com>
* /, res/res_srtp.c: Remove srtp_shutdown from res_srtp The patch
for ASTERISK-19253 included properly shutting down the libsrtp
library in the case of module unload. Unfortunately, not all
distributions have the srtp_shutdown call. As such, this patch
removes calling srtp_shutdown. ........ Merged revisions 356650
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 356651 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/sip/sdp_crypto.c, include/asterisk/res_srtp.h,
main/rtp_engine.c, /, include/asterisk/rtp_engine.h,
res/res_srtp.c: Allow SRTP policies to be reloaded Currently,
when using res_srtp, once the SRTP policy has been added to the
current session the policy is locked into place. Any attempt to
replace an existing policy, which would be needed if the remote
endpoint negotiated a new cryptographic key, is instead rejected
in res_srtp. This happens in particular in transfer scenarios,
where the endpoint that Asterisk is communicating with changes
but uses the same RTP session. This patch modifies res_srtp to
allow remote and local policies to be reloaded in the underlying
SRTP library. From the perspective of users of the SRTP API, the
only change is that the adding of remote and local policies are
now added in a single method call, whereas they previously were
added separately. This was changed to account for the differences
in handling remote and local policies in libsrtp. Review:
https://reviewboard.asterisk.org/r/1741/ (closes issue
ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas
Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt
Jordan (license 6283) (with some small modifications for this
check-in) ........ Merged revisions 356604 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 356605 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-24 00:32 +0000 [r356573] Terry Wilson <twilson@digium.com>
* channels/chan_unistim.c, channels/chan_local.c,
addons/chan_ooh323.c, channels/chan_multicast_rtp.c,
channels/chan_vpb.cc, main/rtp_engine.c, apps/app_meetme.c,
apps/app_dictate.c, apps/app_record.c, apps/app_test.c,
bridges/bridge_softmix.c, channels/chan_gtalk.c, apps/app_ices.c,
res/res_musiconhold.c, channels/chan_iax2.c,
bridges/bridge_multiplexed.c, main/indications.c, main/cli.c,
main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c,
channels/chan_skinny.c, res/res_agi.c, main/features.c,
apps/app_mp3.c, apps/app_dumpchan.c, main/app.c, apps/app_amd.c,
channels/chan_alsa.c, apps/app_confbridge.c,
addons/chan_mobile.c, main/bridging.c, channels/chan_mgcp.c,
apps/app_nbscat.c, main/pbx.c, channels/chan_sip.c,
res/res_fax.c, apps/app_festival.c, channels/chan_bridge.c,
main/channel_internal_api.c, apps/app_fax.c,
apps/app_waitforsilence.c, res/res_adsi.c, channels/chan_agent.c,
bridges/bridge_simple.c, include/asterisk/channel.h,
channels/chan_console.c, apps/app_talkdetect.c,
channels/chan_oss.c, apps/app_speech_utils.c,
channels/chan_usbradio.c, channels/chan_jingle.c,
channels/chan_misdn.c, funcs/func_channel.c, main/file.c,
channels/chan_nbs.c, apps/app_chanspy.c, apps/app_voicemail.c,
res/res_calendar.c: Opaquification for ast_format structs in
struct ast_channel Review:
https://reviewboard.asterisk.org/r/1770/
2012-02-23 20:14 +0000 [r356523] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c, main/features.c: Fix blind transfer
parking issues if the dialed extension is not recognized as a
parking extension. Custom parking extensions may not be coded
such that the first and only extension priority is the Park
application. These custom parking extensions will not be
recognized as parking extensions. When a call is blind
transferred to an extension that is not recognized as a parking
extension, the normal blind transfer code causes the transferred
channel to start executing dialplan. Calls that get parked in
this manner do not know the original channel name that parked the
call so the original parker could never be called back if the
parked call is not retrieved before the timeout time. The parking
space is also announced to the call being parked as a side effect
of not knowing the original parking channel. * Fix handling of
BLINDTRANSFER channel variable for call parking. * Fixed SIP
blind transfer using the wrong dialplan context variable to check
for the parking extension. (closes issue ASTERISK-19322) Reported
by: aragon Tested by: rmudgett, jparker Review:
https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........
Merged revisions 356521 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 356522 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-23 15:49 +0000 [r356477] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Fix ACK routing for non-2xx responses.
When we send an ACK for a 2xx response to an INVITE, we are
supposed to use the learned route set. However, when we receive a
non-2xx final response to an INVITE, we are supposed to send the
ACK to the same place we initially sent the INVITE. We had been
doing this up until the changes went in that would build a route
set from provisional responses. That introduced a regression
where we would use the learned route set under all circumstances.
With this change, we now will set the destination of our ACK
based on the invitestate. If it is INV_COMPLETED then that means
that we have received a non-2xx final response (INV_TERMINATED
indicates a 2xx response was received). If it is INV_CANCELLED,
then that means the call is being canceled, which means that we
should be ACKing a 487 response. The other change introduced here
is setting the invitestate to INV_CONFIRMED when we send an ACK
*after* the reqprep instead of before. This way, we can tell in
reqprep more easily what the invitestate is prior to sending the
ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer
patches: ASTERISK-19389v2.patch uploaded by Mark Michelson
(license #5049) (with some slight modifications prior to commit)
........ Merged revisions 356475 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 356476 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-23 03:27 +0000 [r356429] Paul Belanger <paul.belanger@polybeacon.com>
* /, apps/app_rpt.c: Multiple revisions 356290,356335,356337
........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed,
22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable
compiler error (gcc 4.6.2) Review:
https://reviewboard.asterisk.org/r/1763/ ........ r356335 |
pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2
lines Add back strsep() function for previous commit ........
r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb
2012) | 2 lines Missed one strsep() function ........ Merged
revisions 356290,356335,356337 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 356428 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-23 01:53 +0000 [r356397] Terry Wilson <twilson@digium.com>
* tests/test_substitution.c, tests/test_utils.c: Fix some tests
that didn't get opaquification changes Review:
https://reviewboard.asterisk.org/r/1766/
2012-02-23 00:56 +0000 [r356366] Richard Mudgett <rmudgett@digium.com>
* main/channel_internal_api.c: Revert some apparently accidental
spacing changes.
2012-02-22 21:22 +0000 [r356314] Terry Wilson <twilson@digium.com>
* /, include/asterisk/calendar.h, main/loader.c,
res/res_calendar.c: Track module use count for res_calendar If
the res_calendar module was followed immediately by one of the
calendar tech modules and "core stop gracefully" was run,
Asterisk would crash. This patch adds use count tracking for
res_calendar so that it is unloaded after the tech modules when
shutting down gracefully. It is now not possible to unload all
the of the calendar modules via "module unload res_calednar.so",
but it is still possible to unload them all via "module unload -h
res_calendar.so". Review:
https://reviewboard.asterisk.org/r/1752/ ........ Merged
revisions 356291 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 356297 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-22 21:10 +0000 [r356292] Kevin P. Fleming <kpfleming@digium.com>
* channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c:
Correct some set-but-unused variable warnings in the mISDN
library.
2012-02-22 17:34 +0000 [r356259] Terry Wilson <twilson@digium.com>
* channels/chan_misdn.c: Fix chan_misdn after the lastest
opaquification changes It now compiles, but there are some
unrelated warnings for set but unused variables.
2012-02-22 14:54 +0000 [r356216] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Merged revisions 356215 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r356215 | mjordan | 2012-02-22 08:53:53 -0600
(Wed, 22 Feb 2012) | 32 lines Merged revisions 356214 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012)
| 27 lines Fix potential buffer overrun and memory leak when
executing "sip show peers" The "sip show peers" command uses a
fix sized array to sort the current peers in the peers
ao2_container. The size of the array is based on the current
number of peers in the container. However, once the size of the
array is determined, the number of peers in the container can
change, as the peers container is not locked. This could cause a
buffer overrun when populating the array, if peers were added to
the container after the array was created. Additionally, a memory
leak of the allocated array would occur if a user caused the
_show_peers method to return CLI_SHOWUSAGE. We now create a
snapshot of the current peers using an ao2_callback with the
OBJ_MULTIPLE flag. This size of the array is set to the number of
peers that the iterator will iterate over; hence, if peers are
added or removed from the peers container it will not affect the
execution of the "sip show peers" command. Review:
https://reviewboard.asterisk.org/r/1738/ (closes issue
ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas
Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey
Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan
(license 6283) ........ ................
2012-02-22 00:35 +0000 [r356152-356183] Terry Wilson <twilson@digium.com>
* main/channel.c, main/channel_internal_api.c,
include/asterisk/channel.h: Rename
ast_channel_emulate_dtmf_digit* funcs The accessors names for the
"emulate_dtmf_digit" field on the ast_channel are misleading.
Change them to ast_channel_dtmf_digit_to_emulate*.
* main/channel.c, main/framehook.c, res/res_monitor.c: Fix some
opaquification-related compiler warnings (closes issue
ASTERISK-19419) PseudoReview - seanbright on IRC
2012-02-21 11:17 +0000 [r356111] Sean Bright <sean@malleable.com>
* /, channels/chan_iax2.c: Make 'iax2 show callnumber usage' output
make sense when an IP is passed in. ........ Merged revisions
356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 356108 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-21 04:31 +0000 [r356075] Kinsey Moore <kmoore@digium.com>
* /, main/ccss.c: Add missing newline to ccss state change
notification Move along, nothing to see here... ........ Merged
revisions 356074 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-20 23:43 +0000 [r356042] Terry Wilson <twilson@digium.com>
* main/udptl.c, apps/app_dahdibarge.c, addons/chan_ooh323.c,
cdr/cdr_sqlite3_custom.c, channels/chan_local.c,
main/rtp_engine.c, apps/app_playtones.c, apps/app_record.c,
apps/app_sayunixtime.c, apps/app_test.c, main/devicestate.c,
apps/app_alarmreceiver.c, apps/app_chanisavail.c,
apps/app_ices.c, channels/chan_iax2.c,
bridges/bridge_multiplexed.c, main/cli.c, channels/chan_dahdi.c,
channels/sig_analog.c, main/framehook.c, channels/chan_skinny.c,
main/features.c, apps/app_dumpchan.c, pbx/pbx_realtime.c,
channels/chan_alsa.c, apps/app_externalivr.c, main/bridging.c,
channels/sig_ss7.c, apps/app_milliwatt.c, cdr/cdr_manager.c,
apps/app_dial.c, main/pbx.c, funcs/func_timeout.c,
apps/app_privacy.c, channels/chan_bridge.c, apps/app_echo.c,
apps/app_softhangup.c, apps/app_fax.c, apps/app_dahdiras.c,
channels/chan_agent.c, apps/app_disa.c, bridges/bridge_simple.c,
include/asterisk/channel.h, apps/app_talkdetect.c,
apps/app_transfer.c, main/cel.c, res/res_monitor.c,
apps/app_playback.c, apps/app_speech_utils.c,
channels/chan_misdn.c, apps/app_sendtext.c, funcs/func_channel.c,
funcs/func_cdr.c, channels/sip/dialplan_functions.c,
apps/app_macro.c, apps/app_zapateller.c, main/audiohook.c,
apps/app_chanspy.c, apps/app_voicemail.c, apps/app_cdr.c,
res/res_calendar.c, channels/chan_unistim.c,
channels/chan_multicast_rtp.c, channels/chan_vpb.cc,
apps/app_meetme.c, main/ccss.c, apps/app_dictate.c,
apps/app_authenticate.c, apps/app_readexten.c,
channels/chan_gtalk.c, res/res_musiconhold.c,
apps/app_followme.c, main/channel.c, main/cdr.c,
channels/chan_phone.c, main/dial.c, main/manager.c,
apps/app_osplookup.c, bridges/bridge_builtin_features.c,
res/res_agi.c, apps/app_minivm.c, main/app.c,
apps/app_confbridge.c, main/image.c, apps/app_directory.c,
main/message.c, apps/app_ivrdemo.c, addons/chan_mobile.c,
apps/app_rpt.c, cdr/cdr_custom.c, apps/app_parkandannounce.c,
channels/chan_mgcp.c, apps/app_while.c, res/res_rtp_asterisk.c,
apps/app_read.c, channels/chan_sip.c, apps/app_festival.c,
res/res_fax.c, cdr/cdr_syslog.c, apps/app_waitforsilence.c,
main/channel_internal_api.c, res/res_adsi.c, pbx/pbx_lua.c,
funcs/func_jitterbuffer.c, channels/chan_console.c,
apps/app_queue.c, channels/sig_pri.c, channels/chan_oss.c,
channels/chan_jingle.c, channels/chan_usbradio.c,
apps/app_channelredirect.c, apps/app_forkcdr.c, apps/app_flash.c,
main/abstract_jb.c, main/file.c, channels/chan_h323.c,
include/asterisk/sched.h, res/snmp/agent.c, apps/app_sms.c,
channels/chan_nbs.c, funcs/func_callerid.c, apps/app_verbose.c,
apps/app_stack.c: ast_channel opaquification of pointers and
integral types Review: https://reviewboard.asterisk.org/r/1753/
2012-02-20 18:40 +0000 [r355903-355999] Sean Bright <sean@malleable.com>
* /, channels/chan_iax2.c: Remove spurious warning when
'qualifyfreqnotok' is set successfully. (closes issue
ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright
Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John
Covert (license 5512) ........ Merged revisions 355997 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355998 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/chan_dahdi.c, /: This was a LOG_NOTICE, so roll it back.
........ Merged revisions 355952 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355953 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/chan_dahdi.c, /: Change some debug messages from
LOG_DEBUG to ast_debug. ........ Merged revisions 355949 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355950 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_iax2.c: Add some boilerplate documentation for
IAXVAR and IAXPEER. ........ Merged revisions 355904 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355905 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_iax2.c: Set the length of the ast_sockaddr, so
that we can set it's port later. Without this, the call to
ast_sockaddr_set_port a few lines later is a noop. ........
Merged revisions 355901 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355902 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-18 08:02 +0000 [r355852] Alec L Davis <sivad.a@paradise.net.nz>
* channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c,
channels/sig_ss7.h, /, channels/sig_analog.h, channels/sig_pri.c,
channels/sig_ss7.c: push 'outgoing' flag from sig_XXX up to
chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept
in sync, particulary FXS ast_hangup didn't clear the 'outgoing'
flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync.
Now provides a callback for all the low level sig_XXX modules.
(issue ASTERISK-19316) alecdavis (license 585) Reported by:
Jeremy Pepper Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1747/ ........ Merged
revisions 355850 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355851 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-17 22:03 +0000 [r355795] Sean Bright <sean@malleable.com>
* configs/iax.conf.sample, /, channels/chan_iax2.c: Don't allow
trunkfreq to be greater than 1000ms. ........ Merged revisions
355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 355794 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-17 19:56 +0000 [r355749] Tilghman Lesher <tilghman@meg.abyt.es>
* main/asterisk.c: Non-verbose output should always go to the
remote console, regardless of the previous level.
2012-02-17 19:35 +0000 [r355748] Sean Bright <sean@malleable.com>
* /, channels/chan_iax2.c: Pass the correct value to
ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq
variable to determine how often to send trunk packets, but this
value is in milliseconds while ast_timer_set_rate() expects the
rate argument to be ticks per second. So we divide 1000 by
trunkfreq and pass that in instead. With a default of 20ms, this
change makes IAX2 send trunk packets every 20ms instead of every
50ms. Tracked down by myself and Bob Wienholt. ........ Merged
revisions 355746 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355747 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-17 19:22 +0000 [r355745] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Fix regressions with regards to route-set
creation on early dialogs. This fixes two main issues: 1.
Asterisk would send a CANCEL to the route created by the
provisional response instead of using the same destination it did
in the initial INVITE. 2. If a new route set arrives in a 200 OK
than was in the 1XX response (perfectly possible if our outbound
INVITE gets forked), then the route set in the 200 OK needs to
overwrite the route set in the 1XX response. (closes issue
ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten
Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson
(license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt
(license 6034) Review: https://reviewboard.asterisk.org/r/1749
........ Merged revisions 355732 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355733 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-16 22:00 +0000 [r355667] Paul Belanger <paul.belanger@polybeacon.com>
* apps/app_rpt.c: Fix channel opaquification for app_rpt
2012-02-16 20:03 +0000 [r355624] Sean Bright <sean@malleable.com>
* /, main/audiohook.c: Revert a change to
audio_audiohook_write_list that had no affect. When I made this
change initially, I was under the false impression that the
audiohooks structure remained on the channel after all of the
hooks had been detached. This is not the case, ast ast_read takes
care of removing the audiohooks structure if the lists are empty.
........ Merged revisions 355622 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355623 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-16 19:51 +0000 [r355576-355621] Richard Mudgett <rmudgett@digium.com>
* /, configure, include/asterisk/autoconfig.h.in,
autoconf/ast_c_declare_check.m4 (added), configure.ac,
formats/format_ogg_vorbis.c: Fix compile problem when old version
of libvorbisfile v1.1.2 is used. The principle difference between
libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition
of the predefined callbacks OV_CALLBACKS_xxx in
vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the
configure script to detect if libvorbisfile.h declares
OV_CALLBACKS_NOCLOSE. * Copied the declaration of
OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile.
(closes issue ASTERISK-19370) Reported by: Jonn Taylor ........
Merged revisions 355608 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355620 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, res/res_monitor.c: Fix AMI Monitor action without File header
converting channel name into filename. * Fix potential Solaris
crash if Monitor application has a urlbase and no fname_base
option. ........ Merged revisions 355574 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355575 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-15 19:29 +0000 [r355450-355531] Sean Bright <sean@malleable.com>
* /, channels/chan_iax2.c: When IAX2 debugging is enabled, make
sure to log 'apathetic' messages too. ........ Merged revisions
355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 355530 from
http://svn.asterisk.org/svn/asterisk/branches/10
* build_tools/cflags.xml, channels/chan_iax2.c: Remove IAX_OLD_FIND
from chan_iax2.
* /, channels/chan_iax2.c: Use TRUNK_CALL_START as originally
intended. Back in r646, TRUNK_CALL_START was added and defined as
0x4000. That same value was also hard-coded in one part of the
IAX2 code instead of using the #define. TRUNK_CALL_START has
changed over the years (for dealing with LOW_MEMORY), but the
hard-coded usage was never updated to match. This patch fixes
that. ........ Merged revisions 355448 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355449 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-14 20:27 +0000 [r355413] Tilghman Lesher <tilghman@meg.abyt.es>
* utils/refcounter.c, main/pbx.c, funcs/func_timeout.c,
include/asterisk/autoconfig.h.in, utils/hashtest.c, UPGRADE.txt,
CHANGES, main/config.c, configs/logger.conf.sample,
main/loader.c, include/asterisk/logger.h, main/manager.c,
main/logger.c, utils/ael_main.c, utils/hashtest2.c,
codecs/codec_dahdi.c, main/stdtime/localtime.c, main/asterisk.c,
addons/res_config_mysql.c: Re-commit the verbose branch. This
change permits each verbose destination (consoles, logger) to
have its own concept of what the verbosity level is. The big
feature here is that the logger will now be able to capture a
particular verbosity level without condemning each console to
need to suffer that level of verbosity. Additionally, a stray
'core set verbose' will no longer change what will go to the log.
Review: https://reviewboard.asterisk.org/r/1599/
2012-02-14 19:29 +0000 [r355321-355376] Richard Mudgett <rmudgett@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
formats/format_ogg_vorbis.c: Fix voicemail problems when using
ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file
format because it did not implement the seek and tell format
callbacks among other problems. Since we were already using the
libvorbis and libvorbisenc libraries we can use libvorbisfile as
it is also part of the vorbis library package. * Made use the
libvorbisfile to handle the ogg/vorbis file stream. The
format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
(closes issue ASTERISK-16926) Reported by: sque Patches:
ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded
by sque ........ Merged revisions 355365 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355375 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock
in cel_sqlite_custom reload. (closes issue ASTERISK-19356)
Reported by: Alex Villacis Lasso Patches:
asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch
(license #5617) patch uploaded by Alex Villacis Lasso Review:
https://reviewboard.asterisk.org/r/1740/ ........ Merged
revisions 355319 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355320 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-14 16:28 +0000 [r355274] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Properly invert the return of a strncmp
call. This was causing identification that should have been made
private to be public. (closes issue AST-814) reported by Patrick
Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson
(license 5430) ........ Merged revisions 355268 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355271 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-14 15:58 +0000 [r355230] Jason Parker <jparker@digium.com>
* /, configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3
CDRs by default in sample configs. ........ Merged revisions
355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 355229 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-14 13:35 +0000 [r355184] Sean Bright <sean@malleable.com>
* /, channels/chan_iax2.c: Clear the high order bit from the
destination call number before sending. send_apathetic_reply
takes the incoming frame's source call number as the destination
call number for the outgoing frame. If the incoming frame was a
full frame, then the high order bit of the source call number is
set and will be interpreted as a retransmit when sent back out as
the destination call number. ........ Merged revisions 355182
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 355183 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-14 09:58 +0000 [r355138] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c, /: call manager_event only if there is not
null channel structure (Closes issue ASTERISK-19298) Reported by:
robinfood Patches: issue19298.patch uploaded by may213 (License
#5415) ........ Merged revisions 355136 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355137 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-14 00:43 +0000 [r355102] Russell Bryant <russell@russellbryant.com>
* res/res_agi.c, CHANGES: res_agi: Add AGIEXITONHANGUP variable.
This patch adds a variable AGIEXITONHANGUP for res_agi. If this
variable is set to "yes" on a channel, AGI() will exit
immediately once a channel hangup has been detected. This was the
behavior of AGI() in Asterisk 1.4 and earlier and is still
desired by some people. Review:
https://reviewboard.asterisk.org/r/1734/
2012-02-13 22:04 +0000 [r355055-355058] Richard Mudgett <rmudgett@digium.com>
* pbx/pbx_spool.c, /: Fix occasional incorrectly delayed call-file
execution. Since the dir timestamp is available at one second
resolution, we cannot know if it was updated within the same
second after we scanned it. Therefore, we will force another scan
if the dir was just modified. * Changed to force another scan if
the directory was just modified. (closes issue ASTERISK-19081)
Reported by: Knut Bakke Review:
https://reviewboard.asterisk.org/r/1688/ ........ Merged
revisions 355056 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 355057 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/chan_misdn.c: Fix compile error from most recent
ast_channel opaquification installment.
2012-02-13 19:56 +0000 [r355011] Joshua Colp <jcolp@digium.com>
* /, pbx/pbx_config.c: Only allow one 'dialplan reload' to execute
at a time as otherwise they would share the same common local
context list. (closes issue AST-758) ........ Merged revisions
355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 355010 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-13 17:27 +0000 [r354968] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, addons/chan_ooh323.c,
channels/chan_iax2.c, main/cli.c, channels/chan_dahdi.c,
channels/sig_analog.c, channels/chan_skinny.c, main/features.c,
apps/app_dumpchan.c, pbx/pbx_realtime.c, channels/chan_alsa.c,
apps/app_dial.c, main/pbx.c, apps/app_fax.c,
channels/chan_agent.c, include/asterisk/channel.h,
apps/app_talkdetect.c, main/cel.c, channels/chan_misdn.c,
funcs/func_channel.c, apps/app_macro.c, apps/app_chanspy.c,
res/res_calendar.c, apps/app_voicemail.c,
channels/chan_unistim.c, tests/test_substitution.c,
channels/chan_vpb.cc, apps/app_meetme.c, main/ccss.c,
apps/app_readexten.c, channels/chan_gtalk.c, main/cdr.c,
main/channel.c, main/dial.c, channels/chan_phone.c,
main/manager.c, apps/app_osplookup.c,
bridges/bridge_builtin_features.c, res/res_agi.c,
apps/app_minivm.c, apps/app_confbridge.c, apps/app_directory.c,
addons/chan_mobile.c, apps/app_rpt.c, apps/app_parkandannounce.c,
channels/chan_mgcp.c, apps/app_while.c, funcs/func_dialplan.c,
channels/chan_sip.c, res/res_fax.c, main/channel_internal_api.c,
pbx/pbx_lua.c, channels/sig_pri.c, apps/app_queue.c,
channels/chan_oss.c, channels/chan_jingle.c,
apps/app_directed_pickup.c, main/file.c, channels/chan_h323.c,
res/snmp/agent.c, pbx/pbx_dundi.c, channels/chan_nbs.c,
apps/app_stack.c, apps/app_verbose.c: Opaquify char * and char[]
in ast_channel Review: https://reviewboard.asterisk.org/r/1733/
2012-02-13 17:25 +0000 [r354964] Richard Mudgett <rmudgett@digium.com>
* res/res_config_pgsql.c, /, configs/extconfig.conf.sample: Fix
reconnecting to pgsql database after connection loss. There can
only be one database connection in res_config_pgsql just like
res_config_sqlite. If the connection is lost, the connection may
not get reestablished to the same database if the res_pgsql.conf
and extconfig.conf files are inconsistent. * Made only use the
configured database from res_pgsql.conf. * Fixed potential buffer
overwrite of last[] in config_pgsql(). (closes issue
ASTERISK-16982) Reported by: german aracil boned Review:
https://reviewboard.asterisk.org/r/1731/ ........ Merged
revisions 354953 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354959 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-13 16:42 +0000 [r354939] Joshua Colp <jcolp@digium.com>
* /, apps/app_confbridge.c: Don't try to play sound files that do
not exist. (closes issue ASTERISK-19188) Reported by: slesru
........ Merged revisions 354938 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-10 22:44 +0000 [r354903] Jason Parker <jparker@digium.com>
* /, apps/app_voicemail.c: Fix a voicemail memory leak with
heard/deleted messages. open_mailbox() was changed quite a long
time ago to allocate this memory. close_mailbox() should have
been changed to be responsible for freeing it. ........ Merged
revisions 354889 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354890 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-10 18:08 +0000 [r354837] Richard Mudgett <rmudgett@digium.com>
* main/manager.c, /: Fix AMI Redirect ExtraChannel not redirecting
to the same exten and context. The astman_get_header() never
returns NULL so the check by the code for NULL would never fail.
(closes issue ASTERISK-16974) Reported by: Nuno Borges Patches:
0018325.patch (license #6116) patch uploaded by Nuno Borges
(modified) ........ Merged revisions 354835 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354836 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-10 14:51 +0000 [r354799] Matthew Jordan <mjordan@digium.com>
* apps/app_voicemail.c: Fix IMAP app_voicemail compilation issue
introduced in r354429 This simply fixes the compilation issue
introduced in r354429 by re-adding the 'quote' variable. (closes
issue ASTERISK-19337) Reported by: John Taylor
2012-02-09 22:06 +0000 [r354751] Terry Wilson <twilson@digium.com>
* /, funcs/func_cdr.c: Note that CDRs are immutable once a bridge
is torn down CDRs cannot be modified after a bridge is torn down,
(e.g. after Dial() returns) even though the CDR() function may be
called. Since modifying the CDR code to change this behavior
could very easily break all kinds of things, this patch just
documents this limitation. (closes issues ASTERISK-16923) Review:
https://reviewboard.asterisk.org/r/1720/ ........ Merged
revisions 354749 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354750 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-09 20:52 +0000 [r354657-354704] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Fix parsing of SIP headers where compact
and non-compact headers are mixed Change parsing of SIP headers
so that compactness of the header no longer influences which
header will be chosen. Previously, a non-compact header would be
chosen instead of a preceeding compact-form header. (closes issue
ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/
........ Merged revisions 354702 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354703 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/config.c: Make the config parser remove escaping
backslashes The config parser in Asterisk does not currently
remove a backslash that is used to escape a semicolon which would
otherwise be interpreted as the start of a comment. The change
here causes that backslash to be removed, but does not create a
real escape system in the config parser. The biggest complication
with a real escape system would be breaking existing configs
everywhere (parsing \\ as \ and breaking on escaped non-semicolon
characters) even though it would be the "right" way to do things.
(closes issue ASTERISK-17121) Review:
https://reviewboard.asterisk.org/r/1724/ ........ Merged
revisions 354655 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354656 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-09 18:14 +0000 [r354597] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c, channels/sip/include/config_parser.h,
channels/sip/utils.c (added), configs/sip.conf.sample, CHANGES,
channels/sip/config_parser.c, channels/sip/include/sip.h,
channels/sip/include/sip_utils.h: Add auto_force_rport and
auto_comedia NAT options This patch adds the auto_force_rport and
auto_comedia NAT options. It also converts the nat= setting to a
list of comma-separated combinable options: no, force_rport,
comedia, auto_force_rport, and auto_comedia. nat=yes remains as
an undocumented option equal to "force_rport,comedia". The first
instance of 'yes' or 'no' in the list stops parsing and overrides
any previously set options. If an auto_* option is specified with
its non-auto_ counterpart, the auto setting takes precedence.
This patch builds upon the patch posted to ASTERISK-17860 by JIRA
user pedro-garcia. (closes issue ASTERISK-17860) Review:
https://reviewboard.asterisk.org/r/1698/
2012-02-09 17:17 +0000 [r354552] Mark Michelson <mmichelson@digium.com>
* /, res/res_fax.c: Adding reload support to res_fax.so (closes
issue ASTERISK-16712) reported by Frank DiGennaro Review:
https://reviewboard.asterisk.org/r/1713 ........ Merged revisions
354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 354546 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-09 17:09 +0000 [r354544-354549] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Clean-up of minor formatting issues in
r354542/3/4 rmudgett pointed out some formatting issues in the
check-in for ASTERISK-19290. This cleans those up. Review:
https://reviewboards.asterisk.org/r/1722/ ........ Merged
revisions 354547 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354548 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c: Fix SIP INFO DTMF handling for
non-numeric codes In ASTERISK-18924, SIP INFO DTMF handlingw as
changed to account for both lowercase alphatbetic DTMF events, as
well as uppercase alphabetic DTMF events. When this occurred, the
comparison of the character buffer containing the event code was
changed such that the buffer was first compared again '0' and '9'
to determine if it was numeric. Unfortunately, since the first
character in the buffer will typically be '1' in the case of
non-numeric event codes (10-16), this caused those codes to be
converted to a DTMF event of '1'. This patch fixes that, and
cleans up handling of both application/dtmf-relay and
application/dtmf content types. Review:
https://reviewboard.asterisk.org/r/1722/ (closes issue
ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........
Merged revisions 354542 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354543 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-09 03:09 +0000 [r354497-354498] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/chan_misdn.c: Fix some compile
problems from the 'cppcheck' patch.
* /, apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce.
Well, thats embarrasing. I forgot to initialize the caller_id
storage. (closes issue ASTERISK-19311) Reported by: tootai Tested
by: rmudgett ........ Merged revisions 354495 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354496 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-09 02:28 +0000 [r354494] Russell Bryant <russell@russellbryant.com>
* main/channel.c, /: Remove some unnecessary locking from
ast_hangup(). This patch removes some unnecessary locking of the
channels container in ast_hangup(). The reason this came up is
that this lock can very quickly block the entire system. If any
of the channel cleanup code decides to block, it causes a problem
for the whole system. For example, when audiohooks get destroyed,
if that blocks for a while waiting on the mixmonitor thread to
exit because it's busy blocking on some I/O, it causes a problem
for many other threads in the meantime. Review:
https://reviewboard.asterisk.org/r/1712/ ........ Merged
revisions 354492 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354493 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-08 21:29 +0000 [r354459] Kevin P. Fleming <kpfleming@digium.com>
* res/res_ais.c (removed), contrib/scripts/install_prereq: Revision
354046 added res_corosync as a replacement for res_ais, but
didn't actually remove res_ais. This commit removes it. In
addition, the 'install_prereq' script has been updated to no
longer install AIS dependency packages, and instead install
Corosync packages instead.
2012-02-08 21:28 +0000 [r354458] Terry Wilson <twilson@digium.com>
* channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql,
CHANGES, channels/sip/include/sip.h: Add callbackextension
matching & realtime callbackextensions This patch is based on the
one by David Vossel, developer extrodinaire, at
https://reviewboard.asterisk.org/r/344/. If multiple peers are
defined with the same host/port, but differing
callbackextensions, it chooses the peer with the matching
callbackextension. Since callbackextension creates an outbound
registration with the callbackextension as the Contact address,
matching an incoming request by that (in addition to the
host/port) makes a lot of sense. This patch also adds support for
callbackextension to realtime by querying all peers with
callbackextensions on reload and adding registrations for them.
(closes issue ASTERISK-13456) Review:
https://reviewboard.asterisk.org/r/344/ Review:
https://reviewboard.asterisk.org/r/1717/
2012-02-08 21:25 +0000 [r354450] Kevin P. Fleming <kpfleming@digium.com>
* channels/chan_dahdi.c: Restore some variables removed by the
'cppcheck' patch that were actually needed.
2012-02-08 20:49 +0000 [r354429] Walter Doekes <walter+asterisk@wjd.nu>
* apps/app_dial.c, main/udptl.c, main/pbx.c, addons/chan_ooh323.c,
funcs/func_env.c, funcs/func_strings.c, utils/astman.c,
main/acl.c, apps/app_disa.c, apps/app_alarmreceiver.c,
apps/app_queue.c, channels/chan_iax2.c,
addons/ooh323c/src/memheap.c, channels/chan_usbradio.c,
channels/chan_dahdi.c, apps/app_osplookup.c,
channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_odbc.c,
main/ast_expr2f.c, apps/app_minivm.c, formats/format_h263.c,
addons/chan_mobile.c, apps/app_chanspy.c, main/ast_expr2.fl,
apps/app_voicemail.c: Avoid cppcheck warnings; removing unused
vars and a bit of cleanup. Patch by: Clod Patry Review:
https://reviewboard.asterisk.org/r/1651
2012-02-08 15:28 +0000 [r354395] Kinsey Moore <kmoore@digium.com>
* CHANGES: Add CHANGES documentation for the "pri set debug"
bitmask change (related to ASTERISK-17159)
2012-02-07 21:33 +0000 [r354360] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql:
Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing
instead of "" 2. Don't set ipaddr or port to the string "(null)"
when they are empty 3. Add missing required fields, set default
for lastms to 0, and modify the length of the ipaddr field to 45
in the Postgresql realtime.sql file. (closes issue
ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/
........ Merged revisions 354348 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354349 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-07 18:07 +0000 [r354312-354314] Sean Bright <sean@malleable.com>
* contrib/scripts/live_ast: Continuation of last patch - since
LIVE_AST_LD_PATH_EXTRA will now never be empty, don't check for
it, instead of check if LD_LIBRARY_PATH is already set and if so,
append LIVE_AST_LD_PATH_EXTRA properly.
* contrib/scripts/live_ast: Include live/usr/lib in the shared
library search path to that we pick up libasteriskssl.so at run
time when using live_ast.
* contrib/scripts/live_ast: Whitespace only (remove trailing
spaces)
2012-02-07 15:29 +0000 [r354275] Jonathan Rose <jrose@digium.com>
* /, cdr/cdr_pgsql.c: Fix column duplication bug in module reload
for cdr_pgsql. Prior to this patch, attempts to reload
cdr_pgsql.so would cause the column list to keep its current data
and then add a second copy during the reload. This would cause
attempts to log the CDR to the database to fail. This patch also
cleans up some unnecessary null checks for ast_free and deals
with a few potential locking problems. (closes issue
ASTERISK-19216) Reported by: Jacek Konieczny Review:
https://reviewboard.asterisk.org/r/1711/ ........ Merged
revisions 354263 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354270 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-06 23:15 +0000 [r354174-354218] Richard Mudgett <rmudgett@digium.com>
* /, pbx/pbx_config.c: Improved documentation of CLI "dialplan add
extension" command. * Documented dialplan add extension
<exten>,<priority>,<app(<app-data>)> format. * Allow acceptance
of command without the app-data value. There are many
applications that do no need any parameters so it is silly to
require that field for all commands. * Fixed a couple
ast_malloc/ast_free mismatches with ast_add_extension2() calls.
(closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested
by: rmudgett ........ Merged revisions 354216 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354217 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/sig_pri.h: Restore alternate SIG_PRI_DEBUG_DEFAULT
meaning.
2012-02-06 20:18 +0000 [r354165] Kinsey Moore <kmoore@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c: Allow more control
over the output of pri debug This changes the debuglevel of 'pri
set debug' to a bit mask allowing the user to independently
select bits of output: 1 libpri internals including state machine
2 Decoded Q.931 messages 4 Decoded Q.921 headers 8 raw hex dump
of the full frames Additionally, this ensures that the meaning of
"on" does not change and intrudces intense and hex to simplify
usage. (closes issue ASTERISK-17159) Original-patch-by: wimpy
2012-02-06 17:33 +0000 [r354120] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Add missing headers to AMI UnParkedCall event
to uniquely identify the call. The AMI UnParkedCall event was
missing the Parkinglot and Uniqueid headers that the AMI
ParkedCall event contains. (closes issue ASTERISK-19240) Reported
by: Michael Yara ........ Merged revisions 354116 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354119 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-06 16:38 +0000 [r354084] Joshua Colp <jcolp@digium.com>
* apps/app_meetme.c, UPGRADE.txt: Make the 'c' option to MeetMe
work even if the 'q' option is used. (closes issue
ASTERISK-17053) Reported by: justdave
2012-02-05 10:58 +0000 [r354046] Russell Bryant <russell@russellbryant.com>
* build_tools/menuselect-deps.in, configure,
include/asterisk/autoconfig.h.in, res/res_corosync.c (added),
configure.ac, configs/res_corosync.conf.sample (added), res/ais
(removed), UPGRADE.txt, configs/ais.conf.sample (removed),
CHANGES, makeopts.in: Replace res_ais with a new module,
res_corosync. This patch removes res_ais and introduces a new
module, res_corosync. The OpenAIS project is deprecated and is
now just a wrapper around Corosync. This module provides the same
functionality using the same core infrastructure, but without the
use of the deprecated components. Technically res_ais could have
been used with an AIS implementation other than OpenAIS, but that
is the only one I know of that was ever used. Review:
https://reviewboard.asterisk.org/r/1700/
2012-02-03 21:33 +0000 [r354001] Jonathan Rose <jrose@digium.com>
* /, channels/chan_agent.c: Fixes deadlocks occuring in chan_agent
due to r335976 Bad locking order was added to chan_agent to
prevent segfaults from having no locking in a patch by irroot.
This patch addresses the bad locking order by releasing locks
before getting the right locking order to stop deadlocks from
occuring when doing multiple interactions with agents. (closes
issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review:
https://reviewboard.asterisk.org/r/1708/ ........ Merged
revisions 353999 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 354000 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-03 16:50 +0000 [r353964] Kinsey Moore <kmoore@digium.com>
* UPGRADE.txt, cdr/cdr_adaptive_odbc.c,
configs/cdr_adaptive_odbc.conf.sample: Support schema selection
in cdr_adaptive_odbc Asterisk now supports using ODBC with
databases where a single schema must be selected. Previously,
INSERTs would fail because they did not take into account extra
fields cause by having multiple schemas. This also corrects some
SQL resource leaks. (closes issue ASTERISK-17106) Patch-by:
Alexander Frolkin Patch-by: Tilgnman Lesher
2012-02-03 16:23 +0000 [r353963] Jonathan Rose <jrose@digium.com>
* /, res/res_fax.c: Fixes a segfault occuring when performing
attended transfer with FAXOPT(gateway)=yes (closes issue
ASTERISK-19184) Reported by: Alexandr ........ Merged revisions
353962 from http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-02 22:28 +0000 [r353917] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Ensure entering T.38 passthrough does not
cause an infinite loop After R340970 Asterisk was still polling
the RTCP file descriptor after RTCP is shut down and removed. If
the descriptor happened to have data ready when the removal
occured then Asterisk would go into an infinite loop trying to
read data that it can never actually access. This change disables
the audio RTCP file descriptor for the duration of the T.38
transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan
Vrban ........ Merged revisions 353915 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 353916 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-02 20:18 +0000 [r353872] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c:
Restore the 'w' modifier support for ISDN spans.
Dial(DAHDI/g0/1234w888) This feature also causes the sending
complete ie to be sent for switch types that do not automatically
send the ie. (EuroISDN/ETSI) The main difference between dialing
Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the
sending of the sending complete ie. (closes issue ASTERISK-19176)
Reported by: rmudgett Tested by: rmudgett ........ Merged
revisions 353867 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 353868 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-02 18:55 +0000 [r353821] Mark Michelson <mmichelson@digium.com>
* main/manager.c, /, main/http.c, configs/manager.conf.sample,
include/asterisk/manager.h, configs/http.conf.sample: Fix TLS
port binding behavior as well as reload behavior: * Removes
references to tlsbindport from http.conf.sample and
manager.conf.sample * Properly bind to port specified in
tlsbindaddr, using the default port if specified. * On a reload,
properly close socket if the service has been disabled. A note
has been added to UPGRADE.txt to indicate how ports must be set
for TLS. (closes issue ASTERISK-16959) reported by Olaf
Holthausen (closes issue ASTERISK-19201) reported by Chris
Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas
Review: https://reviewboard.asterisk.org/r/1709 ........ Merged
revisions 353770 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 353820 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-02 17:07 +0000 [r353725-353772] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: Fix sip show peers port output, align
columns, and fix ami port output. A previous patch I committed
from ASTERISK-16930 unexpectedly changed some output for the AMI
action "sippeers" which this patch changes back. Also, this
aligns the output for the cli command "sip show peers" and fixes
another issue that patch introduced by using
ast_sockaddr_stringify calls multiple times without immediately
using the pointer. I also went ahead and did a little janitorial
work to clean up whitespace in _sip_show_peers. (issue
ASTERISK-16930) (closes issue ASTERISK-19281) Reported by:
Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by
Walter Doekes (license 5674) ........ Merged revisions 353769
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 353771 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers
for various functions in chan_sip There are a number of cleaner
looking wrappers for ast_sockaddr_stringify_fmt available which
are slightly more readable than using a direct call to
ast_sockaddr_stringify_fmt. This patch switches a number of those
calls in chan_sip to use those wrappers and is generally
harmless. (Closes issue ASTERISK-16930) Reported by: Michael L.
Young Patches: chan_sip-broken-registration-1.8.diff uploaded by
Michael L. Young (license 5026) ........ Merged revisions 353720
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 353721 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-01 19:53 +0000 [r353647-353685] Richard Mudgett <rmudgett@digium.com>
* channels/chan_unistim.c, channels/chan_multicast_rtp.c,
channels/chan_local.c, addons/chan_ooh323.c,
channels/chan_vpb.cc, channels/chan_gtalk.c,
channels/chan_iax2.c, main/channel.c, channels/chan_phone.c,
channels/chan_dahdi.c, channels/sig_analog.c, main/manager.c,
pbx/pbx_spool.c, channels/chan_skinny.c, main/features.c,
channels/sig_analog.h, channels/chan_alsa.c,
apps/app_confbridge.c, addons/chan_mobile.c, channels/sig_ss7.c,
channels/chan_mgcp.c, main/pbx.c, channels/sig_ss7.h,
channels/chan_sip.c, channels/chan_bridge.c,
channels/chan_agent.c, include/asterisk/channel.h,
channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c,
channels/chan_usbradio.c, channels/chan_jingle.c,
channels/sig_pri.h, channels/chan_misdn.c, channels/chan_h323.c,
channels/chan_nbs.c, include/asterisk/pbx.h: Constify some more
channel driver technology callback parameters. Review:
https://reviewboard.asterisk.org/r/1707/
* cel/cel_pgsql.c, configs/cel_sqlite3_custom.conf.sample,
cel/cel_odbc.c, configs/cel.conf.sample, cel/cel_manager.c,
cel/cel_tds.c, configs/cel_pgsql.conf.sample,
configs/cel_odbc.conf.sample, main/cel.c,
configs/cel_custom.conf.sample: Remove inconsistency in CEL
eventtype for user defined events. The CEL eventtype field for
ODBC and PGSQL backends should be USER_DEFINED instead of the
user defined event name supplied by the CELGenUserEvent
application. If the field is output as a number, the user defined
name does not have a value and is always output as 21 for
USER_DEFINED and the userdeftype field would be required to
supply the user defined name. The following CEL backends
(cel_odbc, cel_pgsql, cel_custom, cel_manager, and
cel_sqlite3_custom) can be independently configured to remove
this inconsistency. * Allows cel_manager, cel_custom, and
cel_sqlite3_custom to behave the same way. (closes issue
ASTERISK-17189) Reported by: Bryant Zimmerman Review:
https://reviewboard.asterisk.org/r/1669/
* main/channel.c, include/asterisk/channel.h: Fix ExtenSpy and
simplify the channel search functions. When ast_channel name was
opaquified, the channel search functions did not get converted
correctly. As a result ExtenSpy which uses a channel iterator
search by exten@context could never find anything. * Updated the
doxygen documentation for the search functions in channel.h.
Review: https://reviewboard.asterisk.org/r/1702/
2012-02-01 15:59 +0000 [r353600] Sean Bright <sean@malleable.com>
* /, include/asterisk/audiohook.h: Resolve an overlap in the
ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and
AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused
unintended side effects. This patch moves
AST_AUDIOHOOK_TRIGGER_WRITE, and updates
AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention.
This will affect existing modules that use these flags, so be
sure to recompile as necessary. (closes issue ASTERISK-19246)
Reported by: feyfre ........ Merged revisions 353598 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 353599 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-01 15:07 +0000 [r353552] Matthew Jordan <mjordan@digium.com>
* /, contrib/init.d/etc_default_asterisk: Added clarification for
the VERBOSITY setting to etc_default_asterisk Clarified that
using the VERBOSITY setting in etc_default_asterisk is the same
as using the -v command line switch, which causes Asterisk to
launch in console mode. (closes issue ASTERISK-17030) Reported
by: Jonas ........ Merged revisions 353550 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 353551 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-02-01 00:08 +0000 [r353504] Terry Wilson <twilson@digium.com>
* /, res/res_calendar.c: Allow res_calendar to be unloaded The
calendaring tech modules depend on res_calendar and initially
res_calendar just bumped the use count so that it couldn't be
unloaded. res_calendar can potentially create many threads and
I've seen issues where the Asterisk shutdown has failed where it
looked like these threads could be the culprit. This patch adds
unload support for res_calendar. Unloading res_calendar will also
unload the dependant tech modules as well. (closes issue
ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/
........ Merged revisions 353502 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 353503 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-31 17:26 +0000 [r353466] Richard Mudgett <rmudgett@digium.com>
* main/manager.c, /, include/asterisk/channel.h: Fix memory leak in
error paths for action_originate(). * Fix memory leak of vars in
error paths for action_originate(). * Moved struct
fast_originate_helper tech and data members to stringfields. *
Simplified ActionID header handling for fast_originate(). * Added
doxygen note to ast_request() and ast_call() and the associated
channel callbacks that the data/addr parameters should be treated
as const char *. Review: https://reviewboard.asterisk.org/r/1690/
........ Merged revisions 353454 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 353463 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-30 23:58 +0000 [r353418] Terry Wilson <twilson@digium.com>
* main/dnsmgr.c, /, channels/chan_sip.c, include/asterisk/dnsmgr.h:
Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr
currently takes a pointer to an ast_sockaddr and updates it
anytime an address resolves to something different. There are a
couple of issues with this. First, the ast_sockaddr is usually
the address of an ast_sockaddr inside a refcounted struct and we
never bump the refcount of those structs when using dnsmgr. This
makes it possible that a refresh could happen after the
destructor for that object is called (despite ast_dnsmgr_release
being called in that destructor). Second, the module using dnsmgr
cannot be aware of an address changing without polling for it in
the code. If an action needs to be taken on address update (like
re-linking a SIP peer in the peers_by_ip table), then polling for
this change negates many of the benefits of having dnsmgr in the
first place. This patch adds a function to the dnsmgr API that
calls an update callback instead of blindly updating the address
itself. It also moves calls to ast_dnsmgr_release outside of the
destructor functions and into cleanup functions that are called
when we no longer need the objects and increments the refcount of
the objects using dnsmgr since those objects are stored on the
ast_dnsmgr_entry struct. A helper function for returning the
proper default SIP port (non-tls vs tls) is also added and used.
This patch also incorporates changes from a patch posted by Timo
Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue
ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/
........ Merged revisions 353371 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 353397 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-30 22:44 +0000 [r353347-353370] Alec L Davis <sivad.a@paradise.net.nz>
* /, channels/chan_sip.c: Merged revisions 353369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r353369 | alecdavis | 2012-01-31 11:42:28 +1300
(Tue, 31 Jan 2012) | 9 lines Merged revisions 353368 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31
Jan 2012) | 2 lines prevent debug messsges displaying -ve Cseq
numbers. Missed in R353320 ........ ................
* channels/sip/include/dialog.h, /, channels/chan_sip.c,
channels/sip/include/sip.h: Merged revisions 353321 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r353321 | alecdavis | 2012-01-31 11:16:22 +1300
(Tue, 31 Jan 2012) | 25 lines Merged revisions 353320 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan
2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number
value MUST be expressible as a 32-bit unsigned integer * fix: use
%u instead of %d when dealing with CSeq numbers - to remove
possibility of -ve numbers. * fix: change all uses of seqno and
friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t.
Summary of CSeq numbers. An initial CSeq number must be less than
2^31 A CSeq number can increase in value up to 2^32-1 An
incrementing CSeq number must not wrap around to 0. Tested with
Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1699/ ........
................
2012-01-30 21:34 +0000 [r353262-353319] Kevin P. Fleming <kpfleming@digium.com>
* Makefile: Correct serious flaw in the top-level Makefile.
* include/asterisk.h, /, main/Makefile, main/libasteriskssl.c
(added), configure.ac, Makefile.moddir_rules, main/ssl.c
(removed), addons, CHANGES, include/asterisk/optional_api.h,
Makefile, build_tools/mkpkgconfig, configure, main, makeopts.in,
build_tools/make_defaults_h, main/libasteriskssl.exports.in
(added): Address OpenSSL initialization issues when using
third-party libraries. When Asterisk is used with various
third-party libraries (CURL, PostgresSQL, many others) that have
the ability themselves to use OpenSSL, it is possible for
conflicts to arise in how the OpenSSL libraries are initialized
and shutdown. This patch addresses these conflicts by 'wrapping'
the important functions from the OpenSSL libraries in a new
shared library that is part of Asterisk itself, and is loaded in
such a way as to ensure that *all* calls to these functions will
be dispatched through the Asterisk wrapper functions, not the
native functions. This new library is optional, but enabled by
default. See the CHANGES file for documentation on how to disable
it. Along the way, this patch also makes a few other minor
changes: * Changes MODULES_DIR to ASTMODDIR throughout the build
system, in order to more closely match what is used during
run-time configuration. * Corrects some errors in the configure
script where AC_CHECK_TOOLS was used instead of AC_PATH_PROG. *
Adds a new variable for linker flags in the build system
(DYLINK), used for producing true shared libraries (as opposed to
the dynamically loadable modules that the build system produces
for 'regular' Asterisk modules). * Moves the Makefile bits that
handle installation and uninstallation of the main Asterisk
binary into main/Makefile from the top-level Makefile. * Moves a
couple of useful preprocessor macros from optional_api.h to
asterisk.h. Review: https://reviewboard.asterisk.org/r/1006/
* /, channels/chan_sip.c: Clarify log WARNING message when
port-zero SDP 'm' lines received. Previously, if an m-line in an
SDP offer or answer had a port number of zero, that line was
skipped, and resulted in an 'Unsupported SDP media type...'
warning message. This was misleading, as the media type was not
unsupported, but was ignored because the m-line indicated that
the media stream had been rejected (in an answer) or was not
going to be used (in an offer). ........ Merged revisions 353260
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 353261 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-29 22:33 +0000 [r353224] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Allow softkey reject while device onhook.
Fixes up softkey endcall. Previous code was a copy of onhook, now
allows for endcall softkey to be used while device is still
onhook.
2012-01-29 02:45 +0000 [r353177] Russell Bryant <russell@russellbryant.com>
* /, main/netsock.c: Find even more network interfaces. The
previous change made the code look for emN and pciN in addition
to what it did originally, which was search for ethN. However, it
needed to be looking for pciN#N, so that's what it does now. This
also moves the memset() to be before every ioctl(). ........
Merged revisions 353175 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 353176 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-28 14:52 +0000 [r353128] Kevin P. Fleming <kpfleming@digium.com>
* main/rtp_engine.c, /: Add 'L16-256' MIME subtype alias for
slin16. Asterisk has supported the 'L16' MIME subtype for 16kHz
signed linear (PCM) audio for quite some time, but some endpoints
refer to it as 'L16-256'. This commit adds this as an alias for
the existing format. ........ Merged revisions 353126 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 353127 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-28 04:31 +0000 [r353079] Russell Bryant <russell@russellbryant.com>
* /, main/netsock.c: Update ast_set_default_eid() to find more
network interfaces. As of Fedora 15, ethN is not the name of
ethernet interfaces. The names are emN or pciN. Update some code
that searched for interfaces named ethN to look for the new
names, as well. For more information about why this change was
made, see this page: http://domsch.com/blog/?p=455 ........
Merged revisions 353077 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 353078 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-27 21:38 +0000 [r352996-353040] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_queue.c: Audit of ao2_iterator_init() usage for v10.
Missed one. ........ Merged revisions 353039 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, tests/test_format_api.c: Audit of ao2_iterator_init() usage
for v10. Fix double format_cap iterator cleanup. ........ Merged
revisions 352992 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-27 19:26 +0000 [r352981] Jonathan Rose <jrose@digium.com>
* /, res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor
with no valid channel not close AMI session. I also went ahead
and took a little time to make sure that the manager value
AMI_SUCCESS was used instead of just return 0 being thrown around
everywhere since that's how we handle this stuff these days.
(closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches:
res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey
(license 5766) ........ Merged revisions 352959 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352965 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-27 18:47 +0000 [r352957] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c, /, channels/chan_sip.c,
include/asterisk/indications.h, res/snmp/agent.c,
main/taskprocessor.c, apps/app_queue.c, channels/chan_iax2.c,
apps/app_chanspy.c, main/indications.c, res/res_odbc.c,
res/res_srtp.c: Audit of ao2_iterator_init() usage for v1.8.
Fixes numerous reference leaks and missing ao2_iterator_destroy()
calls as a result. Review:
https://reviewboard.asterisk.org/r/1697/ ........ Merged
revisions 352955 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352956 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-27 15:57 +0000 [r352916] Terry Wilson <twilson@digium.com>
* res/res_calendar_exchange.c, res/res_calendar_caldav.c,
res/res_calendar.c: Add aresult variable for CALENDAR_WRITE This
patch adds a CALENDAR_SUCCESS=1/0 variable that is set to show
whether or not CALENDAR_WRITE has passed. This patch also adds
some debugging for caldav PUT responses and no longer treats
responses with no body as an error (as a PUT gets a 201 Created
with no body). (closes issue ASTERISK-16903) Reported by: Clod
Patry Tested by: Terry Wilson Patches: calendarstatus.diff
uploaded by Clod Patry (License #5138), slightly modified by
Terry Wilson Review: https://reviewboard.asterisk.org/r/1692/ -
This line, and those below, will be ignored-- M
res/res_calendar.c M res/res_calendar_exchange.c M
res/res_calendar_caldav.c
2012-01-27 00:11 +0000 [r352864] Alec L Davis <sivad.a@paradise.net.nz>
* /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
revisions 352863 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r352863 | alecdavis | 2012-01-27 13:08:03 +1300
(Fri, 27 Jan 2012) | 19 lines Merged revisions 352862 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan
2012) | 12 lines rfc4235 - Section 4.1: Versions MUST be
representable using a non-negative 32 bit integer. If a BLF
subscription exists for long enough, using %d may print negative
version numbers. Unlikely, as 2^32 at 1 update per second is ~137
years, or half that before the versions number started going
negative. Tested with Asterisk 1.8.8.2 with Grandstream phones.
alecdavis (license 585) Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1694/ ........
................
2012-01-26 20:44 +0000 [r352821] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c, /: Fix outbound DTMF for inband mode (tell
asterisk core to generate DTMF sounds). (Closes issue
ASTERISK-19233) Reported by: Matt Behrens Patches:
chan_ooh323.c.patch uploaded by Matt Behrens (License #6346)
........ Merged revisions 352807 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352817 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-26 19:09 +0000 [r352757] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: Copy amaflags to sip_pvt from peer during
create_addr_from_peer For whatever reason, we don't have a single
function for copying data like this from SIP peers to the SIP
pvt. This patch adds the copying of amaflags to the sip_pvt, but
it would probably be worth discussing this function along with
the others that essentially just copy some amount of data from a
peer to a private. (Closes issue ASTERISK-19029) Reported by:
Matt Lehner ........ Merged revisions 352755 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352756 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-26 06:36 +0000 [r352706] Alec L Davis <sivad.a@paradise.net.nz>
* /, channels/chan_sip.c: Merged revisions 352705 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r352705 | alecdavis | 2012-01-26 19:33:11 +1300
(Thu, 26 Jan 2012) | 27 lines Merged revisions 352704 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan
2012) | 20 lines Cleanup dialog-info+xml Notify dialog Make
similar to other Notify messages. sample output: <?xml
version="1.0"?> <dialog-info
xmlns="urn:ietf:params:xml:ns:dialog-info" version="715"
state="full" entity="sip:8523@192.168.x.xx"> <dialog id="8523">
<state>terminated</state> </dialog> </dialog-info> Tested with
Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585)
Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1693/ ........
................
2012-01-25 22:25 +0000 [r352659] Paul Belanger <paul.belanger@polybeacon.com>
* /, apps/app_voicemail.c: Fix -Werror=unused-but-set-variable
compiler error (gcc 4.6.2) ........ Merged revisions 352643 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352651 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-25 21:31 +0000 [r352626] Kevin P. Fleming <kpfleming@digium.com>
* Makefile, include/asterisk/version.h (added), main/test.c,
build_tools/make_version_h (removed), include/asterisk: Remove
"asterisk/version.h" in favor of "asterisk/ast_version.h". A long
time ago, in a land far far away, we added
"asterisk/ast_version.h", which provides the ast_get_version()
and ast_get_version_num() functions. These were added so that
modules that needed the version information for the Asterisk
instance they were loaded in could actually get it (as opposed
the version that they were compiled against). We changed
everything in the tree to use the new mechanism (although later
main/test.c was added using the old method). However, the old
mechanism was never removed, and as a result, new code is still
trying to use it. This commit removes asterisk/version.h and
replaces it with a header that will generate a compile-time error
if you try to use it (the error message tells you which header
you should use instead). It also removes the Makefile and
build_tools bits that generated the file, and it updates
main/test.c to use the 'proper' method of getting the Asterisk
version information. This is an API change and thus is being
committed for trunk only, but it's a fairly minor one and
definitely improves the situation for out-of-tree modules.
2012-01-25 17:33 +0000 [r352565] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Remove some extraneous debugging from
registry memleak fix ........ Merged revisions 352551 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352556 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-25 17:23 +0000 [r352538] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c, CHANGES, main/message.c,
channels/sip/include/sip.h: Fixes for sending SIP MESSAGE outside
of calls. * Fix authenticate MESSAGE losing custom headers added
by the MESSAGE_DATA function in the authorization attempt. * Pass
up better From header contents for SIP to use. Now is in the
"display-name" <URI> format expected by MessageSend. (Note that
this is a behavior change that could concievably affect some
people.) * Block user from adding standard headers that are added
automatically. (To, From,...) * Allow the user to override the
Content-Type header contents sent by MessageSend. * Decrement
Max-Forwards header if the user transferred it from an incoming
message. * Expand SIP short header names so the dialplan and
other code only has to deal with the full names. * Documents what
SIP expects in the MessageSend(from) parameter. (closes issue
ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917)
Reported by: Shaun Clark Review:
https://reviewboard.asterisk.org/r/1683/ ........ Merged
revisions 352520 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-25 17:02 +0000 [r352519] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Clean up some SIP registry-related memory
leaks 1) Be sure and free at unload the epa_backend we allocate
at startup 2) Do the same sip_registry cleanup at unload we do at
reload Review: https://reviewboard.asterisk.org/r/1689/ ........
Merged revisions 352514 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352515 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-25 16:54 +0000 [r352517] Kevin P. Fleming <kpfleming@digium.com>
* main/format.c, /, main/format_cap.c, main/format_pref.c:
Eliminate unnecessary rebuilds of main/format*.c. These files
have no need to include "asterisk/version.h", and doing so forces
them to be rebuilt each time a Subversion checkout moves between
'modified' and 'unmodified' states. ........ Merged revisions
352516 from http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-25 16:42 +0000 [r352513] Jonathan Rose <jrose@digium.com>
* /, configs/sip.conf.sample: Redocuments sip types peer, user,
friend in sip.conf.sample There was faulty information in the
sample config describing user as a synonym for friend so it has
been changed to better elaborate on the differences between the
three entity types. (closes issue ASTERISK-15537) Reported by:
yarique ........ Merged revisions 352511 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352512 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-25 01:21 +0000 [r352475] Terry Wilson <twilson@digium.com>
* channels/chan_vpb.cc: Fix channel opaquification of stringfields
for chan_vpb
2012-01-24 22:28 +0000 [r352431] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Don't do a DNS lookup on an outbound
REGISTER host if there is an outbound proxy configured. (closes
issue ASTERISK-16550) reported by: Olle Johansson ........ Merged
revisions 352424 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352430 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-24 20:37 +0000 [r352377] Jonathan Rose <jrose@digium.com>
* /, sounds/Makefile: Set core sounds version to 1.4.22. Now that
we have the right license for the Russian 1.4.22 sounds as well
as the sounds for the Australian English 1.4.22 sounds, we can
finally set the sounds to use 1.4.22! (closes issue
ASTERISK-18978) Reported by: Cameron Twomey Patches:
confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002
uploaded by Cameron Twomey ........ Merged revisions 352367 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352373 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-24 20:12 +0000 [r352348] Terry Wilson <twilson@digium.com>
* channels/chan_local.c, addons/chan_ooh323.c, main/say.c,
apps/app_record.c, apps/app_sayunixtime.c, channels/chan_iax2.c,
main/cli.c, channels/chan_dahdi.c, channels/sig_analog.c,
channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c,
channels/chan_alsa.c, pbx/pbx_realtime.c, apps/app_externalivr.c,
apps/app_dial.c, main/pbx.c, apps/app_page.c,
channels/chan_bridge.c, apps/app_privacy.c,
channels/chan_agent.c, apps/app_disa.c,
include/asterisk/channel.h, main/aoc.c, apps/app_talkdetect.c,
main/cel.c, res/res_monitor.c, apps/app_playback.c,
apps/app_speech_utils.c, channels/chan_misdn.c,
funcs/func_channel.c, apps/app_chanspy.c, apps/app_voicemail.c,
channels/chan_unistim.c, channels/chan_multicast_rtp.c,
apps/app_meetme.c, apps/app_dictate.c, apps/app_authenticate.c,
apps/app_readexten.c, apps/app_userevent.c,
res/res_musiconhold.c, channels/chan_gtalk.c,
apps/app_followme.c, main/cdr.c, main/channel.c,
channels/chan_phone.c, main/dial.c, main/manager.c,
apps/app_minivm.c, res/res_agi.c, main/app.c,
apps/app_confbridge.c, main/image.c, apps/app_directory.c,
addons/chan_mobile.c, apps/app_rpt.c, channels/chan_mgcp.c,
apps/app_parkandannounce.c, channels/chan_sip.c, res/res_fax.c,
main/channel_internal_api.c, channels/chan_console.c,
channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c,
funcs/func_global.c, channels/chan_jingle.c,
channels/chan_usbradio.c, channels/chan_h323.c, main/file.c,
res/snmp/agent.c, channels/chan_nbs.c, apps/app_stack.c,
addons/app_saycountpl.c: Opaquify channel stringfields Continue
channel opaque-ification by wrapping all of the stringfields.
Eventually, we will restrict what can actually set these
variables, but the purpose for now is to hide the implementation
and keep people from adding code that directly accesses the
channel structure. Semantic changes will follow afterward.
Review: https://reviewboard.asterisk.org/r/1661/
2012-01-24 17:04 +0000 [r352293] Richard Mudgett <rmudgett@digium.com>
* /, funcs/func_odbc.c: Fix locking issues with channel datastores
in func_odbc.c. * Fixed a potential memory leak when an existing
datastore is manually destroyed by inline code instead of calling
ast_datastore_free(). (closes issue ASTERISK-17948) Reported by:
Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/
........ Merged revisions 352291 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352292 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-23 20:31 +0000 [r352229-352232] Mark Michelson <mmichelson@digium.com>
* /, main/features.c: Fix grammar of comment. ........ Merged
revisions 352230 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352231 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/features.c: Fix blind transfers from failing if an 'h'
extension is present. This prevents the 'h' extension from being
run on the transferee channel when it is transferred via a native
transfer mechanism such as SIP REFER. (closes ASTERISK-19173)
Reported by: Ross Beer Tested by: Kristjan Vrban Patches:
ASTERISK-19173 by Mark Michelson (license 5049) Review:
https://reviewboard.asterisk.org/r/1685 ........ Merged revisions
352199 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 352228 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-23 19:22 +0000 [r352166] Matthew Jordan <mjordan@digium.com>
* /, res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17,
V27, V29) before starting spandsp layer While the FAXOPT function
could be used to set the modem capabilities, the input to that
function was not being applied correctly to the spandsp layer.
This patch applies the current model capabilities before starting
the spandsp layer. (closes issue: ASTERISK-16409) Reported by:
Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson
Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license
5081) spandsp-modems-10.diff uploaded by mnicholson (license
5081) ........ Merged revisions 352144 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352149 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-23 18:34 +0000 [r352093-352134] Jonathan Rose <jrose@digium.com>
* configs/musiconhold.conf.sample, res/res_musiconhold.c, CHANGES:
Add an announcement option to music-on-hold - plays sound when
put on hold/between songs This is a feature patch which allows an
'announcement' option to be specified in musiconhold.conf which
should be set to the name of a sound. If a valid sound is
specified for this option, then it will be played on that music
on hold class whenever a channel bound to that class is put on
hold as well as when Asterisk is able to detect that a song has
ended before starting the next song (excludes external players).
(closes ASTERISK-18977) Reported by: Timo Teräs Patches:
asterisk-moh-announcement.diff uploaded by Timo Teräs (license
5409)
* CHANGES, apps/app_mixmonitor.c: Adds the ability to stop specific
mixmonitors by using unique IDs set at monitor launch. MixMonitor
receives a new option i(channel_variable) which stores the unique
id at said variable. StopMixMonitor now accepts ID as an optional
argument, which if included will make StopMixMonitor specifically
target the mixmonitor on that particular channel. CLI commands
and AMI actions have been ammended to work with the IDs as well.
In addition, monitors across a channel can now be listed be
listed via CLI command "mixmonitor list <channel>" which will
display all of the mixmonitors active on that channel along with
the files they each have open. Created by Sergio González Martín.
(closes issue ASTERISK-19096) Reported by: Sergio González Martín
Review: https://reviewboard.asterisk.org/r/1643/ Review:
https://reviewboard.asterisk.org/r/1682/
2012-01-23 17:36 +0000 [r352092] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the
defined enum values. The invalid value used when notifycid was
enabled was benign. As far as the code was concerned -1 and 1 are
equivalent. (closes issue ASTERISK-19232) Reported by: Eike
Kuiper ........ Merged revisions 352090 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352091 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-21 00:23 +0000 [r352041] Richard Mudgett <rmudgett@digium.com>
* /, funcs/func_timeout.c, main/app.c: Fix ast_app_dtget() time
unit inconsistency. Note: Noone calls ast_app_dtget() with the
timeout parameter of zero so the bad code normally will never get
executed. * Fix unnecessary floating point division in
func_timeout.c timeout_write() when all other values are
integers. (closes issue ASTERISK-16817) Reported by: Dmitry
Andrianov ........ Merged revisions 352029 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352035 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-21 00:11 +0000 [r352018-352019] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Remove XXX comment that is not necessary.
........ Merged revisions 352016 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352017 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c: Fix RTP reference leak. If a blind
transfer were initiated using a REFER without a prior reINVITE to
place the call on hold, AND if Asterisk were sending RTCP
reports, then there was a reference for the RTP instance of the
transferer. This fixes the issue by merging two similar but
slightly conflicting sections of code into a single area. It also
adds a stop_media_flows() call in the case that the transferer's
UA never sends a BYE to us like it is supposed to. (issue
ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/
........ Merged revisions 352014 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 352015 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-20 23:05 +0000 [r351977] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Make CLI sip show channel list the complete
route set. (closes issue ASTERISK-16877) Reported by: klaus3000
Patches: show-complete-routeset-patch.txt (license #5054) patch
uploaded by klaus3000 (modified)
2012-01-20 21:26 +0000 [r351939] Kinsey Moore <kmoore@digium.com>
* channels/chan_sip.c, UPGRADE.txt: SIP session timeout AMI event
Add an AMI event in the Call category that is issued when a call
is terminated due to either RTP stream inactivity or SIP session
timer expiration. Event description: Event: SessionTimeout
Source: source Channel: channel-name Uniqueid: channel-unique-id
`source` can be either RTPTimeout or SIPSessionTimer (closes
issue ASTERISK-16467) Patch-by: Kirill Katsnelson
2012-01-20 20:47 +0000 [r351900-351913] Mark Michelson <mmichelson@digium.com>
* main/features.c, UPGRADE.txt, CHANGES,
configs/features.conf.sample: Various parking improvements. *
Adds per-parking lot options comebackcontext and comebackdialtime
* Makes comebacktoorigin settable per parking lot * Sets a PARKER
channel variable when comebacktoorigin is disabled (closes issue
ASTERISK-16643) Reported by: Mitch Sharp (bluecrow76) Patches:
asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff
by Mitch Sharp (bluecrow76) license 5231 with updates by me.
Review: https://reviewboard.asterisk.org/r/1674 Review:
https://reviewboard.asterisk.org/r/963 Reviewed by Richard
Mudgett
* apps/app_mixmonitor.c: Prevent potential buffer overflow on AMI
MixMonitor command. Don't be alarmed. This only affected trunk,
and it would have required manager access to your system.
2012-01-20 19:36 +0000 [r351817-351862] Kinsey Moore <kmoore@digium.com>
* /, codecs/ilbc/iLBC_test.c: More corrections for the ilbc code
These changes are in a file that is not compiled by default, and
so were missed on earlier checks. ........ Merged revisions
351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 351861 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Restore
LSF_check function calls from set/unused variable removal These
functions are not noops and modify the array that is passed in.
Thanks for the catch Richard. ........ Merged revisions 351818
from http://svn.asterisk.org/svn/asterisk/branches/10
* /, codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Remove
more set, but unused variables in the ilbc codec GCC 4.6.3 caught
these in dev mode as well. ........ Merged revisions 351816 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-20 16:00 +0000 [r351764] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: Adds setting of mwi_from field to
check_auth_result check_peer_ok (closes ASTERISK-19057) Reported
By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license
5242) ........ Merged revisions 351759 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351762 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-20 16:00 +0000 [r351763] Matthew Jordan <mjordan@digium.com>
* /, codecs/ilbc/helpfun.c: Remove unused variable 'tmp' from
helpfun in ilbc codec gcc version 4.6.2 caught an unused variable
in the ilbc codec library. This would prevent compilation with
--enable-dev-mode; variable removed. ........ Merged revisions
351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 351761 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-20 13:12 +0000 [r351709] Stefan Schmidt <sst@sil.at>
* /, contrib/asterisk-ng-doxygen: enable doxygen build for files in
the channels/sip folder like reqresp_parser.c ........ Merged
revisions 351707 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351708 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-19 23:31 +0000 [r351667] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor
fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in
get_calleridname() parsing and ensure that the output buffer is
nul terminated. * Make get_calleridname() truncate the name it
parses if the given buffer is too small rather than abandoning
the parse and not returning anything for the name. Adjusted
get_calleridname_test() unit test to handle the truncation
change. * Fix get_in_brackets_test() unit test to check the
results of get_in_brackets() correctly. * Fix
parse_name_andor_addr() to not return the address of a local
buffer. This function is currently not used. * Fix potential NULL
pointer dereference in sip_sendtext(). * No need to
memset(calleridname) in check_user_full() or tmp_name in
get_name_and_number() because get_calleridname() ensures that it
is nul terminated. * Reply with an accurate response if
get_msg_text() fails in receive_message(). This is academic in
v1.8 because get_msg_text() can never fail. ........ Merged
revisions 351618 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351646 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-19 22:44 +0000 [r351613] Kinsey Moore <kmoore@digium.com>
* res/res_rtp_asterisk.c, /: Correct output of RTCP jitter
statistics in SR and RR reports Change the RTCP RR and SR
generation code to convert Asterisk's internal jitter statistics
to be represented in RTP timestamp units based on the rate of the
codec in use instead of in seconds. (closes issue ASTERISK-14530)
........ Merged revisions 351611 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351612 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-19 21:55 +0000 [r351561] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c, include/asterisk/netsock2.h: Eliminates
doubling the :port part of SIP Notify Message-Account headers.
This patch prevents the domain string from getting mangled during
the initreqprep step by moving the initialization to before its
immediate use. It also documents this pitfall for the
ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported
by: Yuri Review: https://reviewboard.asterisk.org/r/1678/
........ Merged revisions 351559 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351560 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-19 21:13 +0000 [r351506] Joshua Colp <jcolp@digium.com>
* /, channels/chan_sip.c: Prevent crash when an SDP offer is
received with an encrypted video stream when support for video is
disabled and res_srtp is loaded. (closes issue ASTERISK-19202)
Reported by: Catalin Sanda ........ Merged revisions 351504 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351505 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-18 21:06 +0000 [r351452] Matthew Jordan <mjordan@digium.com>
* codecs/ilbc/syntFilter.c (added), /, codecs/ilbc/iCBConstruct.h
(added), codecs/ilbc/iLBC_test.c (added),
codecs/ilbc/syntFilter.h (added), codecs/ilbc/StateConstructW.c
(added), codecs/ilbc/packing.c (added),
codecs/ilbc/StateConstructW.h (added), codecs/ilbc/packing.h
(added), codecs/ilbc/getCBvec.c (added), codecs/ilbc/LPCdecode.c
(added), codecs/ilbc/enhancer.c (added), codecs/ilbc/lsf.c
(added), codecs/ilbc/iLBC_encode.c (added),
codecs/ilbc/getCBvec.h (added), codecs/ilbc/LPCdecode.h (added),
codecs/ilbc/iLBC_define.h (added), codecs/ilbc/FrameClassify.c
(added), codecs/ilbc/enhancer.h (added), codecs/ilbc/lsf.h
(added), codecs/ilbc/extract-cfile.awk (added),
codecs/ilbc/iLBC_encode.h (added), codecs/ilbc/Makefile,
codecs/ilbc/FrameClassify.h (added), codecs/ilbc/helpfun.c
(added), codecs/ilbc/LICENSE_ADDENDUM (added),
codecs/ilbc/doCPLC.c (added), codecs/ilbc/anaFilter.c (added),
codecs/ilbc/helpfun.h (added), codecs/ilbc/createCB.c (added),
codecs/ilbc/doCPLC.h (added), codecs/ilbc/anaFilter.h (added),
codecs/ilbc/constants.c (added), codecs/ilbc/iLBC_decode.c
(added), codecs/ilbc/createCB.h (added), codecs/ilbc/constants.h
(added), codecs/ilbc/iLBC_decode.h (added),
codecs/ilbc/iCBSearch.c (added), codecs/ilbc/filter.c (added),
codecs/ilbc/gainquant.c (added), codecs/ilbc/hpInput.c (added),
codecs/ilbc/hpOutput.c (added), codecs/ilbc/iCBSearch.h (added),
codecs/ilbc/rfc3951.txt (added), codecs/ilbc/filter.h (added),
codecs/ilbc/gainquant.h (added), codecs/ilbc/LPCencode.c (added),
codecs/ilbc/hpInput.h (added), codecs/ilbc/PATENTS (added),
codecs/ilbc/StateSearchW.c (added), codecs/ilbc/hpOutput.h
(added), codecs/codec_ilbc.c, contrib/scripts/get_ilbc_source.sh,
codecs/ilbc/LICENSE (added), codecs/ilbc/LPCencode.h (added),
codecs/ilbc/StateSearchW.h (added), codecs/ilbc/iCBConstruct.c
(added): Include iLBC source code for distribution with Asterisk
This patch includes the iLBC source code for distribution with
Asterisk. Clarification regarding the iLBC source code was
provided by Google, and the appropriate licenses have been
included in the codecs/ilbc folder. Review:
https://reviewboard.asterisk.org/r/1675 Review:
https://reviewboard.asterisk.org/r/1649 (closes issue:
ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan
........ Merged revisions 351450 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351451 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-18 16:02 +0000 [r351409] Stefan Schmidt <sst@sil.at>
* /, channels/chan_sip.c: The get_pai function in chan_sip.c didn't
recognized a proper callerid name and number from a
P-Asserted-Identity cause the header parsing logic was wrong.
Changing the parsing functions to the sip header parsing APIs in
reqresp_parser.h solves this problem. Review:
https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and
Mark Michelson ........ Merged revisions 351396 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351408 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-17 19:45 +0000 [r351360] Walter Doekes <walter+asterisk@wjd.nu>
* Makefile: Fix support for parallel building with make (-j).
Previously make -j <N> would cause a race between doing cleanup
of certain files (defaults.h, menuselect, ...) and creating them
anew. Add a new target that depends on cleanup only and has a
submake doing the rest as command string. This way the cleanup
goes first. (closes issue ASTERISK-18751) Tested by: Jeremy
Kister Reviewed by: Paul Belanger Review:
https://reviewboard.asterisk.org/r/1660
2012-01-17 17:23 +0000 [r351311] Mark Michelson <mmichelson@digium.com>
* res/res_rtp_asterisk.c, /: Eliminate odd initialization of
probation variable. ........ Merged revisions 351306 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351308 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-17 17:15 +0000 [r351290] Jonathan Rose <jrose@digium.com>
* res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES: Adds
pjmedia probation concepts to res_rtp_asterisk's learning mode.
In order to better handle RTP sources with strictrtp enabled
(which is now default in 10) using the learning mode to figure
out new sources when they change is handled by checking for a
number of consecutive (by sequence number) packets received to an
rtp struct based on a new configurable value called 'probation'.
Also, during learning mode instead of liberally accepting all
packets received, we now reject packets until a clear source has
been determined. Review: https://reviewboard.asterisk.org/r/1663/
........ Merged revisions 351287 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351289 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-17 16:56 +0000 [r351288] Mark Michelson <mmichelson@digium.com>
* /, channels/chan_sip.c: Use built-in parsing functions for
Contact and Record-Route headers. If a Contact or a Record-Route
header had a quoted string with an item in angle brackets, then
we would mis-parse it. For instance, "Bob <1234>"
<1234@example.org> would be misparsed as having the URI "1234"
The fix for this is to use parsing functions from
reqresp_parser.h since they are heavily tested and are awesome.
(issue ASTERISK-18990) ........ Merged revisions 351284 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351286 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-17 16:08 +0000 [r351235] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Fix udptl issue with initial INVITE
introduced by r351027 When an inital INVITE occurs that contains
image media, a channel is not yet associated with the SIP dialog.
The file descriptor associated with the udptl session needs to be
set in initialize_udptl or in sip_new to account for this
scenario. ........ Merged revisions 351233 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351234 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-17 01:48 +0000 [r351184] Russell Bryant <russell@russellbryant.com>
* /, channels/chan_sip.c: Merged revisions 351183 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r351183 | russell | 2012-01-16 20:43:19 -0500
(Mon, 16 Jan 2012) | 29 lines Merged revisions 351182 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012)
| 22 lines Add some missing locking in chan_sip. This patch adds
some missing locking to the function
send_provisional_keepalive_full(). This function is called from
the scheduler, which is processed in the SIP monitor thread. The
associated channel (or pbx) thread will also be using the same
sip_pvt and ast_channel so locking must be used. The
sip_pvt_lock_full() function is used to ensure proper locking
order in a safe manner. In passing, document a suspected
reference counting error in this function. The "fix" is left
commented out because when the "fix" is present, crashes occur.
My theory is that fixing it is exposing a reference counting
error elsewhere, but I don't know where. (Or my analysis of this
being a problem could have been completely wrong in the first
place). Leave the comment in the code for so that someone may
investigate it again in the future. Also add a bit of doxygen to
transmit_provisional_response(). (closes issue ASTERISK-18979)
Review: https://reviewboard.asterisk.org/r/1648 ........
................
2012-01-16 21:50 +0000 [r351082-351143] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200
response to INVITE When handling a non-2xx final response on an
INVITE transaction, we have to keep the transaction around after
we send an ACK in case we receive a retransmission of the
response so we can re-transmit the ACK, but also tear down the
ast_channel as soon as we transmit the ACK. Before this patch, we
could fail at both of these things. Calling
sip_alreadygone/needdestroy prevented us from keeping the
transaction up and retransmitting the ACK, and queueing
CONGESTION was not sufficient to cause the channel to be torn
down when originating calls via the CLI, for example. This patch
queues a hangup with CONGESTION instead of just queueing
CONGESTION for these responses and removes the sip_alreadygone
and sip_needdestroy calls from handle_response_invite on non-2xx
responses. It relies on the hangup calling sip_scheddestroy. For
more information, see section 17.1.1.1 of RFC 3261. (closes issue
ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/
........ Merged revisions 351130 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351131 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c: Don't prematurely stop SIP session timer
When Asterisk is the UAS (incoming call, endpoint is re-inviting)
the SIP session timer expires after half the time the sip
endpoint indicates in the Session-expires header in
proc_session_timer(). The session timer was being stopped totally
and being handled as an error case instead of running again until
the second expiry. This patch treats the half-time expiry as a
non-error case and continues the timer until the true expiry.
(closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested
by: Thomas Arimont Patches: session_timer_fix.diff by Terry
Wilson (License #5357) based on session_timer.patch by Thomas
Arimont (License #5525) ........ Merged revisions 351080 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351081 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-16 19:49 +0000 [r351079] Tilghman Lesher <tilghman@meg.abyt.es>
* main/ast_expr2.y, CHANGES, main/ast_expr2.c: Add ABS() absolute
value function to the expression parser.
2012-01-16 19:13 +0000 [r351029] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Create and initialize udptl only when
dialog negotiates for image media Prior to this patch, the udptl
struct was allocated and initialized when a dialog was associated
with a peer that supported T.38, when a new SIP channel was
allocated, or what an INVITE request was received. This resulted
in any dialog associated with a peer that supported T.38 having
udptl support assigned to it, including the UDP ports needed for
communication. This occurred even in non-INVITE dialogs that
would never send image media. This patch creates and initializes
the udptl structure only when the SDP for a dialog specifies that
image media is supported, or when Asterisk indicates through the
appropriate control frame that a dialog is to support T.38.
(closes issue ASTERISK-16698) Reported by: under Tested by:
Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan
(License #6283) (closes issue ASTERISK-16794) Reported by: Elazar
Broad Tested by: Stefan Schmidt review:
https://reviewboard.asterisk.org/r/1668/ ........ Merged
revisions 351027 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 351028 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-16 17:12 +0000 [r350979] Sean Bright <sean@malleable.com>
* /, main/db.c: Sort the output of 'database showkey' as well. You
can pass wildcards (%) to the database CLI commands, so this will
sort the returned list of matches. ........ Merged revisions
350978 from http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-16 17:07 +0000 [r350977] Joshua Colp <jcolp@digium.com>
* main/rtp_engine.c, /: Add missing code to set direct RTP setup
information during dialing. ........ Merged revisions 350975 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350976 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-16 14:31 +0000 [r350939] Sean Bright <sean@malleable.com>
* /, main/db.c: Sort the output of 'database show' by key. This
more closely mimics the behavior of 'database show' before the
conversion to sqlite3. ........ Merged revisions 350938 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-15 20:16 +0000 [r350887-350890] Walter Doekes <walter+asterisk@wjd.nu>
* /, main/asterisk.c: Allow only one thread at a time to do
asterisk cleanup/shutdown. Add locking around the
really-really-quit part of the core stop/restart part. Previously
more than one thread could be called to do cleanup, causing
atexit handlers to be run multiple times, in turn causing
segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson
Review: https://reviewboard.asterisk.org/r/1662/ Review:
https://reviewboard.asterisk.org/r/1658/ ........ Merged
revisions 350888 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350889 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, utils/extconf.c: Fix -Werror=unused-but-set-variable compile
error in utils/extconf.c. Note that I'm not confirming legitimacy
of having that file in tree at all. Is anyone using
aelparse/conf2ael? (issue ASTERISK-15350) ........ Merged
revisions 350885 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350886 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-14 16:43 +0000 [r350791-350839] Kevin P. Fleming <kpfleming@digium.com>
* /, configure, autoconf/ast_gcc_attribute.m4, configure.ac,
autoconf/libcurl.m4: Ensure that all AC_LANG_PROGRAM calls in the
configure script are properly quoted. Recent versions of autoconf
(2.68 on my system) won't properly process the configure script
unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in
the script were, but many were not. This patch corrects the
unquoted calls. ........ Merged revisions 350837 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350838 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_h323.c, addons/chan_mobile.c,
res/res_pktccops.c, contrib/scripts/install_prereq: Multiple
revisions 350788-350789 ........ r350788 | kpfleming | 2012-01-14
09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines Ensure that two
prerequisites are properly installed on Debian-style
distributions. * Don't specify a specific version of libgmime;
newer versions are available now and acceptable. * Install
libsrtp so that res_srtp can be built. ........ r350789 |
kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3
lines Correct some 'set-but-not-used' variable warnings. ........
Merged revisions 350788-350789 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350790 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-13 22:17 +0000 [r350738] Kinsey Moore <kmoore@digium.com>
* /, include/asterisk/autoconfig.h.in: Run bootstrap.sh for the for
the ASTERISK-18929 fix configure and autoconfig.h.in were not
regenerated when the fix was committed. ........ Merged revisions
350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 350737 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-13 21:52 +0000 [r350735] Richard Mudgett <rmudgett@digium.com>
* /, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample:
Correct eventtype names in cel_odbc and cel_pgsql sample files
........ Merged revisions 350733 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350734 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-13 21:42 +0000 [r350732] Kinsey Moore <kmoore@digium.com>
* /, configure.ac, bootstrap.sh, main/asterisk.c: Make sure
asterisk builds on OpenBSD OpenBSD defines SO_PEERCRED, but it
returns a 'struct sockpeercred', not 'struct ucred', which causes
compilation of main/asterisk.c to fail in read_credentials().
This allows configure to check for sockpeercred and asterisk to
deal with it properly. (closes issue ASTERISK-18929) Reported-by:
Barry Miller Patch-by: Barry Miller ........ Merged revisions
350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 350731 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-13 20:32 +0000 [r350681] Mark Michelson <mmichelson@digium.com>
* /, channels/sip/config_parser.c: Set port to a default sane value
if a bogus one is provided when parsing hostnames. ........
Merged revisions 350679 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350680 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-13 18:52 +0000 [r350605-350644] Richard Mudgett <rmudgett@digium.com>
* main/features.c: Remove some dead code in ast_bridge_call(). None
of the parameters to ast_bridge_call() can be NULL for the bridge
to work so no need to check for it.
* configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c,
configs/cel.conf.sample, /, cel/cel_manager.c,
configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample,
main/cel.c, configs/cel_custom.conf.sample: Add missing CEL
logging fields to various CEL backends. Multiple revisions
350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51
-0600 (Fri, 13 Jan 2012) | 12 lines Add missing CEL logging
fields to various CEL backends. * Add missing eventextra to
cel_psql.c and cel_odbc.c. * Add missing PeerAccount and
EventExtra to cel_manager.c. * Add missing userdeftype support
for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample.
(closes issue ASTERISK-17190) Reported by: Bryant Zimmerman
........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13
Jan 2012) | 8 lines Use compatible names for event extra data for
various CEL backends. * Change eventextra to extra in cel_psql.c
and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c.
(issue ASTERISK-17190) ........ Merged revisions 350555,350571
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 350585 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-13 17:00 +0000 [r350551-350554] Matthew Jordan <mjordan@digium.com>
* /, apps/app_queue.c: Realtime queues failed to load queue
information without queue member table Previously, realtime
queues could be loaded without defining the queue member table.
This allowed for queue members to be dynamic, while the realtime
queue definitions could exist in some backing storage. Revision
342223 broke this when it changed the return value for
realtime_multientry to return NULL when no results are returned.
Previously, an empty ast_config object was expected. (closes
issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene
Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt
Jordan (license 6283) ........ Merged revisions 350552 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350553 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, bridges/bridge_builtin_features.c, channels/chan_bridge.c,
include/asterisk/bridging.h, apps/app_confbridge.c,
main/bridging.c: Fix crash from bridge channel hangup race
condition in ConfBridge This patch addresses two issues in
ConfBridge and the channel bridge layer: 1. It fixes a race
condition wherein the bridge channel could be hung up 2. It
removes the deadlock avoidance from the bridging layer and makes
the bridge_pvt an ao2 ref counted object Patch by David Vossel
(mjordan was merely the commit monkey) (issue ASTERISK-18988)
(closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested
by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by
David Vossel (license 5628) (closes issue ASTERISK-19100)
Reported by: Matt Jordan Tested by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1654/ ........ Merged
revisions 350550 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-12 16:10 +0000 [r350503] Jonathan Rose <jrose@digium.com>
* /, main/features.c: Adds peer to CEL report on CEL_BRIDGE_START
and CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic
Colledge Patches: features_18.patch uploaded by Nic Colledge
(license 6245) ........ Merged revisions 350501 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350502 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-11 22:53 +0000 [r350416-350454] Richard Mudgett <rmudgett@digium.com>
* /, main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a
CEL dummy channel. (closes issue ASTERISK-19180) Reported by:
Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license
#5909) patch uploaded by Corey Farrell ........ Merged revisions
350452 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 350453 from
http://svn.asterisk.org/svn/asterisk/branches/10
* apps/app_dial.c, /, CHANGES, apps/app_followme.c: Make FollowMe
optionally update connected line information when the accepting
endpoint is bridged. Like Dial and Queue, FollowMe needs to deal
with AST_CONTROL_CONNECTED_LINE information so when the parties
are initially bridged, the connected line information will be
correct. * Added the 'I' option just like the app_dial and
app_queue 'I' option. * Made 'N' option ignored if the call is
already answered. (closes issue ASTERISK-18969) Reported by:
rmudgett Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1656/ ........ Merged
revisions 350364 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350415 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-11 19:19 +0000 [r350365] Terry Wilson <twilson@digium.com>
* main/channel.c: Always treat arguments to get_by_name_cb as
strings Initially, support was left in for the old style of
searching, even though it wasn't actually used. In the case of
name_len != 0, the OBJ_KEY flag isn't passed because we aren't
matching on a full key and therefor can't use the hash function
to optimize. The code left in to support the old way of searching
unfortunately treated a prefix search like this as though an
ast_channel struct was passed as an arg and caused a crash. This
patch also adds needed parentheses around some matching
conditions. (closes issue ASTERISK-19182)
2012-01-10 22:10 +0000 [r350273-350313] Richard Mudgett <rmudgett@digium.com>
* /, funcs/func_lock.c: Fix absolute/relative time mismatch in LOCK
function. The time passed by the LOCK function to an internal
function was relative time when the function expected absolute
time. * Don't use C++ keywords in get_lock(). (closes issue
ASTERISK-16868) Reported by: Andrey Solovyev Patches:
20101102__issue18207.diff.txt (license #5003) patch uploaded by
Andrey Solovyev (modified) ........ Merged revisions 350311 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350312 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/channel.c: Fix compiler warnings reported by gcc v4.2.4.
2012-01-09 22:15 +0000 [r350223] Terry Wilson <twilson@digium.com>
* main/udptl.c, apps/app_dahdibarge.c, addons/chan_ooh323.c,
channels/chan_local.c, main/rtp_engine.c, main/say.c,
apps/app_record.c, apps/app_test.c, channels/console_video.c,
apps/app_alarmreceiver.c, apps/app_chanisavail.c,
bridges/bridge_multiplexed.c, channels/chan_iax2.c,
main/indications.c, main/cli.c, channels/chan_dahdi.c,
channels/sig_analog.c, channels/chan_skinny.c, main/features.c,
apps/app_dumpchan.c, pbx/pbx_realtime.c, apps/app_amd.c,
channels/chan_alsa.c, apps/app_externalivr.c, main/bridging.c,
apps/app_milliwatt.c, channels/sig_ss7.c, apps/app_dial.c,
main/pbx.c, apps/app_page.c, apps/app_softhangup.c,
apps/app_fax.c, apps/app_dahdiras.c, channels/chan_agent.c,
apps/app_disa.c, include/asterisk/channel.h, main/aoc.c,
apps/app_talkdetect.c, main/cel.c, res/res_mutestream.c,
res/res_monitor.c, apps/app_playback.c, channels/chan_misdn.c,
funcs/func_channel.c, apps/app_macro.c, apps/app_mixmonitor.c,
apps/app_chanspy.c, apps/app_voicemail.c, res/res_calendar.c,
channels/chan_unistim.c, channels/chan_vpb.cc, main/ccss.c,
apps/app_meetme.c, apps/app_readexten.c, res/res_musiconhold.c,
main/autochan.c, channels/chan_gtalk.c, apps/app_followme.c,
res/res_jabber.c, main/cdr.c, main/channel.c, main/dial.c,
channels/chan_phone.c, main/manager.c, funcs/func_groupcount.c,
funcs/func_audiohookinherit.c, funcs/func_frame_trace.c,
res/res_agi.c, apps/app_minivm.c, main/app.c,
apps/app_confbridge.c, apps/app_rpt.c, addons/chan_mobile.c,
apps/app_parkandannounce.c, channels/chan_mgcp.c,
apps/app_jack.c, apps/app_adsiprog.c, channels/chan_sip.c,
res/res_fax.c, apps/app_waitforsilence.c, funcs/func_lock.c,
main/channel_internal_api.c (added), res/res_adsi.c,
pbx/pbx_lua.c, channels/chan_console.c, apps/app_getcpeid.c,
channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c,
funcs/func_global.c, channels/chan_usbradio.c,
channels/chan_jingle.c, apps/app_flash.c,
apps/app_directed_pickup.c, main/abstract_jb.c, main/file.c,
channels/chan_h323.c, res/snmp/agent.c, pbx/pbx_dundi.c,
apps/app_sms.c, channels/chan_nbs.c, apps/app_stack.c,
main/dsp.c: Replace direct access to channel name with accessor
functions There are many benefits to making the ast_channel an
opaque handle, from increasing maintainability to presenting ways
to kill masquerades. This patch kicks things off by taking things
a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and
converting the existing code to using them. When all fields are
done, we can move ast_channel to a C file from channel.h and lop
off the '__do_not_use_'. This patch sets up
main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be
for any API functions in channel.c to use the accessor functions.
No more monkeying around with channel internals. We should use
our own APIs. The interesting changes in this patch are the
addition of channel_internal_api.c, the moving of the AST_DATA
stuff from channel.c to channel_internal_api.c (note: the
AST_DATA stuff will have to be reworked to use accessor functions
when ast_channel is really opaque), and some re-working of the
way channel iterators/callbacks are handled so as to avoid
creating fake ast_channels on the stack to pass in matching data
by directly accessing fields (since "name" is a stringfield and
the fake channel doesn't init the stringfields, you can't use the
ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and
ast_channel_name_set(chan, name) for a setter. The majority of
the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review:
https://reviewboard.asterisk.org/r/1655/
2012-01-09 21:56 +0000 [r350222] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_iax2.c: Fix joinable thread terminating without
joiner memory leak in chan_iax.c. The iax2_process_thread() can
exit without anyone waiting to join the thread. If noone is
waiting to join the thread then a large memory leak occurs. *
Made iax2_process_thread() deatach itself if nobody is waiting to
join the thread. (closes issue ASTERISK-17339) Reported by:
Tzafrir Cohen Patches:
asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch
(license #5617) patch uploaded by Alex Villacis Lasso (modified)
(closes issue ASTERISK-17825) Reported by: wangjin ........
Merged revisions 350220 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350221 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-09 19:37 +0000 [r350181] Walter Doekes <walter+asterisk@wjd.nu>
* /, main/db.c: Fix shutdown handling of sqlite3 astdb. If a
db_sync was scheduled just before shutdown, the atexit code
calling db_sync would have no effect, causing the astdb commit
thread to stay alive. This caused the SIP/realtime_sipregs test
to fail. (The fallback kill would run the atexit code again and
that would wreak havoc.) This fixes that the atexit kill
condition is picked up properly. (closes issue ASTERISK-18883)
Reviewed by: Terry Wilson Review:
https://reviewboard.asterisk.org/r/1659 ........ Merged revisions
350180 from http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-09 18:58 +0000 [r350077-350130] Richard Mudgett <rmudgett@digium.com>
* /, contrib/scripts/valgrind_compare (added): Multiple revisions
350127-350128 ........ r350127 | rmudgett | 2012-01-09 12:40:33
-0600 (Mon, 09 Jan 2012) | 12 lines Update contrib script
live_ast to invoke Asterisk with valgrind and suppression file. *
Added valgrind_compare script to compare two valgrind log files
for differences. (issue ASTERISK-17339) Reported by: Tzafrir
Cohen Patches: valgrind_compare (license #5035) script uploaded
by Tzafrir Cohen live_ast_valgrind.diff (license #5035) patch
uploaded by Tzafrir Cohen live_ast_valgrind_v2.diff (license
#5185) patch uploaded by Paul Belanger ........ r350128 |
rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11
lines live_ast: valgrind: run asterisk under valgrind Adds a new
sub-command, "valgrind" to live_ast. It runs asterisk under
valgrind. The extra command-line parameters are passed to
Asterisk as usual, and parameters to valgrind are passed through
LIVE_AST_VALGRIND_ARGS in live.conf . Review:
https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636
from http://svn.asterisk.org/svn/asterisk/branches/10 ........
Merged revisions 350127-350128 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350129 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/asterisk.c: Make Asterisk -x command line parameter imply
-r parameter presence. The Asterisk -x command line parameter is
documented inconsistently. * Made the -x documentation and
behavior consistent. * Since this is also a new year, updated the
copyright notices while here. (closes issue ASTERISK-19094)
Reported by: Eugene Patches:
issueA19094_correct_asterisk_option_x.patch (license #5674) patch
uploaded by Walter Doekes (modified) Tested by: Eugene ........
Merged revisions 350075 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 350076 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-09 15:40 +0000 [r350025] Kinsey Moore <kmoore@digium.com>
* /, apps/app_meetme.c: Prevent SLA settings from getting wiped out
on reload If SLA was reloaded without the config file being
changed, current settings got wiped out before the SLA reload
code decided it wasn't going to reload the file since nothing was
changed. Moving the settings reset later in the reload process
fixes this. (closes issue AST-744) ........ Merged revisions
350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 350024 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-06 23:31 +0000 [r349978] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Don't leak CID in From header when
presentation=unavailable When someone does
Set(CALLERPRES()=unavailable) (or
Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From
header shows "Anonymous" <anonymous@anonymous.invalid>. When
sendrpid=yes/pai, the From header will still display the callerid
info, even though we supply an rpid header with the anonymous
info. It seems like we shouldn't leak that info in any case.
Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04
seems to indicate that one shouldn't send identifying info in the
From in this case. This patch anonymizes the From header as well
even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review:
https://reviewboard.asterisk.org/r/1649/ ........ Merged
revisions 349968 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349977 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-06 21:26 +0000 [r349929] Kinsey Moore <kmoore@digium.com>
* /, pbx/pbx_lua.c: Fix lua goto detection to prevent unexpected
behavior with confbridge A bug in the pbx_lua goto detection was
causing the dialplan to hangup unexpectedly after confbridge
exited if it had called lua dialplan code during execution.
Patch-by: Timo Teras Acked-by: Matt Nicholson (closes issue
ASTERISK-18976) ........ Merged revisions 349928 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-06 16:50 +0000 [r349874] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_followme.c: Fix memory leaks in app_followme
find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt
Jordan ........ Merged revisions 349872 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349873 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-05 23:58 +0000 [r349823] Matthew Jordan <mjordan@digium.com>
* /, res/res_fax.c: Fix premature free'ing of the frame committed
in r349608 Even though we set the frame to the ast_null_frame and
return that, the caller of the frame hook may still need the
frame. This now is a bit more careful about when it frees the
frame, i.e., only under the same conditions that applied when we
duplicated it in the first place. ........ Merged revisions
349822 from http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-05 23:47 +0000 [r349782-349821] Richard Mudgett <rmudgett@digium.com>
* /, cel/cel_sqlite3_custom.c: Make not assume that the
cel_sqlite3_custom SQL table primary key is AcctId. If a table is
created by some other application and the primary key is not
named "AcctId", cel/cel_sqlite3_custom.c will always try to
create the table and fail because it already exists. * Change the
SQL table query to not require AcctId as the primary key. (closes
issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch
(license #6337) patch uploaded by socketpair ........ Merged
revisions 349819 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349820 from
http://svn.asterisk.org/svn/asterisk/branches/10
* UPGRADE.txt, pbx/pbx_config.c: Make pbx_config.c use Gosub
instead of Macro call for stdexten. Users created by users.conf
with hasvoicemail=yes have been documented as using a Gosub to
stdexten since v1.6.0. However, the code still generates dialplan
to access stdexten as a Macro as documented in v1.4; which does
not work with the newer extensions.conf.sample file. * Make
generated dialplan access the stdexten dialplan with the
documented Gosub instead of the older Macro style. (closes issue
ASTERISK-18809) Reported by: Jay Allen Patches:
gosub_patch-pbx_config.patch (license #6323) patch uploaded by
Jay Allen (modified) Tested by: rmudgett
2012-01-05 22:11 +0000 [r349733] Kinsey Moore <kmoore@digium.com>
* /, main/file.c: Allow playback of formats that don't support
seeking ast_streamfile previously did unconditional seeking on
files that broke playback of formats that don't support that
functionality. This patch avoids the seek that was causing the
problem. This regression was introduced in r158062. (closes issue
ASTERISK-18994) Patch-by: Timo Teras ........ Merged revisions
349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 349732 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-05 22:02 +0000 [r349674-349730] Jonathan Rose <jrose@digium.com>
* /, main/dsp.c: Fix an issue where dsp.c would interpret multiple
dtmf events from a single key press. When receiving calls from a
mobile phone into a DISA system on a connection with significant
interference, the reporter's Asterisk system would interpret DTMF
incorrectly and replicate digits received. This patch resolves
that by increasing the number of frames a mismatch has to be
detected before assuming the DTMF is over by 1 frame and adjusts
dtmf_detect function to reset hits and misses only when an edge
is detected. (closes issue ASTERISK-17493) Reported by: Alec
Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis
(license 5546) Review: https://reviewboard.asterisk.org/r/1130/
........ Merged revisions 349728 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349729 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/asterisk.c: Ensures Asterisk closes when receiving
terminal signals in 'no fork' mode. When catching a signal, in no
fork mode the console thread is identical to the thread
responsible for catching the signal and closing Asterisk, which
requires it to first dispense with the console thread. Prior to
this patch, if these threads were identical, upon receiving a
killing signal, the thread will send an URG signal to itself,
which we also catch and then promptly do nothing with. Obviously
this isn't useful behavior. (closes issue ASTERISK-19127)
Reported By: Bryon Clark Patches: quit_on_signals.patch uploaded
by Bryon Clark (license 6157) ........ Merged revisions 349672
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 349673 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-04 22:23 +0000 [r349609-349634] Matthew Jordan <mjordan@digium.com>
* /, apps/confbridge/conf_config_parser.c: Fix for ConfBridge
config parser unlocking channel mutex too many times When looking
up a ConfBridge profile, the config parser would, if it found a
channel datastore on the channel requesting the bridge profile,
unlock the channel mutex twice. Since that's a little aggressive,
it now only unlocks it once. (closes issue ASTERISK-19042)
Reported by: Matt Jordan Tested by: Matt Jordan Patches: 19042
uploaded by David Vossel (license 5628) ........ Merged revisions
349619 from http://svn.asterisk.org/svn/asterisk/branches/10
* /, res/res_fax.c: Free successfully translated frame in
fax_gateway_framehook A frame that is translated via
ast_translate is also duplicated via ast_frdup. This will
allocate a new frame on the heap, which needs to be free'd at the
appropriate time. This issue reporter used valgrind to find that
this occurred in res_fax's fax_gateway_framehook; a quick search
through the code showed that only place this was currently not
handling the translatted frame properly. (closes issue
ASTERISK-19133) Reported by: Sylvain Rochet ........ Merged
revisions 349608 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-04 20:55 +0000 [r349560] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Fix segfault in chan_dahdi for
CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private
pointer checks in the following chan_dahdi channel callbacks:
dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and
dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by:
Diego Aguirre Tested by: rmudgett ........ Merged revisions
349558 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 349559 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-04 20:24 +0000 [r349506-349535] Kinsey Moore <kmoore@digium.com>
* contrib/init.d/rc.debian.asterisk, /: Make debian init script
conform to the LSB standard Previously, this init script would
return 1 if Asterisk was already running. This is incorrect
behavior according to the LSB standard and has been fixed by
returning 0 instead. (closes issue ASTERISK-17958) Reported-by:
johnc ........ Merged revisions 349529 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349532 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, contrib/scripts/autosupport.8, contrib/scripts/autosupport:
Update autosupport script and man page Added information
collection from the output of the utilities: top, free, uptime,
ifconfig Added information collection from the output of the
Asterisk command 'dahdi show status' Added option / flag '-n,
--non-interactive' Updated man page to reflect new option / flag
'-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes
issue AST-749) ........ Merged revisions 349504 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349505 from
http://svn.asterisk.org/svn/asterisk/branches/10
2012-01-04 19:53 +0000 [r349452-349503] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: Adds Subscription-State header to notify
with call completion. per RFC3265 (Closes issue ASTERISK-17953)
Reported by: George Konopacki Patches: 19400.patch uploaded by
mmichelson (license 5049) ........ Merged revisions 349482 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349502 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/pbx.c, /: Fix documentation for SayNumber to reflect the
fact that language is changed in CHANNEL() (closes issue
ASTERISK-18962) reported by: Nir Simionovich ........ Merged
revisions 349450 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349451 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-31 15:48 +0000 [r349409-349410] Russell Bryant <russell@russellbryant.com>
* channels/chan_sip.c: Fix some minor formatting issues based on
coding guidelines.
* channels/sip/include/dialog.h, channels/chan_sip.c,
include/asterisk/astobj2.h, main/astobj2.c: Constify tag argument
in REF_DEBUG related code.
2011-12-29 15:16 +0000 [r349341] Matthew Jordan <mjordan@digium.com>
* main/rtp_engine.c, /: Handle AST_CONTROL_UPDATE_RTP_PEER frames
in local bridge loop Failing to handle
AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
causes the loop to exit prematurely. This causes a variety of
negative side effects, depending on when the loop exits. This
patch handles the frame by essentially swallowing the frame in
the local loop, as the current channel drivers expect the RTP
bridge to handle the frame, and, in the case of the local bridge
loop, no additional action is necessary. (issue ASTERISK-19040)
(issue ASTERISK-19128) (issue ASTERISK-17725) (issue
ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan
Schmidt Tested by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1640/ ........ Merged
revisions 349339 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349340 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-28 21:39 +0000 [r349291] Sean Bright <sean@malleable.com>
* /, main/audiohook.c: Use ast_audiohook_write_list_empty to
determine if our lists are empty instead of duplicating that
logic. ........ Merged revisions 349289 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349290 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-28 19:00 +0000 [r349249-349251] Kevin P. Fleming <kpfleming@digium.com>
* utils, /: Tell Subversion to gnore the 'astdb2bdb' binary file if
it exists. ........ Merged revisions 349250 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, res/res_fax.c, include/asterisk/dsp.h,
include/asterisk/res_fax.h, res/res_fax_spandsp.c, main/dsp.c:
Improve T.38 gateway V.21 preamble detection. This commit removes
the V.21 preamble detection code previously added to the generic
DSP implementation in Asterisk, and instead enhances the res_fax
module to be able to utilize V.21 preamble detection
functionality made available by FAX technology modules. This
commit also adds such support to res_fax_spandsp, which uses the
Spandsp modem tone detection code to do the V.21 preamble
detection. There should be no functional change here, other than
much more reliable V.21 preamble detection (and thus T.38 gateway
initiation). ........ Merged revisions 349248 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-27 20:55 +0000 [r349196] Matthew Jordan <mjordan@digium.com>
* /, res/res_timing_pthread.c, include/asterisk/module.h,
res/res_timing_dahdi.c, res/res_timing_timerfd.c,
res/res_musiconhold.c: Fix timing source dependency issues with
MOH Prior to this patch, res_musiconhold existed at the same
module priority level as the timing sources that it depends on.
This would cause a problem when music on hold was reloaded, as
the timing source could be changed after res_musiconhold was
processed. This patch adds a new module priority level,
AST_MODPRI_TIMING, that the various timing modules are now loaded
at. This now occurs before loading other resource modules, such
that the timing source is guaranteed to be set prior to resolving
the timing source dependencies. (closes issue ASTERISK-17474)
Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf,
Wes Van Tlghem, elguero, Thomas Arimont Patches:
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff
uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff
uploaded by elguero (License #5026)
asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by
elguero (License #5026) Review:
https://reviewboard.asterisk.org/r/1578/ ........ Merged
revisions 349194 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349195 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-27 17:17 +0000 [r349146] Sean Bright <sean@malleable.com>
* /, main/audiohook.c: Once an audiohook is attached to a channel,
we continue to transcode all of the frames, even after all of the
hooks are detached. This patch short-cicuits us out before we
transcode unnecessarily. ........ Merged revisions 349144 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 349145 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-23 21:19 +0000 [r349106] Matthew Jordan <mjordan@digium.com>
* contrib/realtime/mysql/voicemail.sql,
configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
Allow overriding of IMAP server settings on a user by user basis
This patch allows the imapserver, imapport, and imapflags
settings to be overridden for any voicemail user. It also
documents the settings in the sample voicemail.conf file, and
updates the voicemail schema to allow storage of those columns.
(closes issue ASTERISK-16489) Reporter: Hubert Mickael Tested by:
Matt Jordan Review: https://reviewboard.asterisk.org/r/1614/
2011-12-23 20:42 +0000 [r349097-349098] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c, main/features.c, configs/sip.conf.sample,
channels/sip/include/sip.h: INFO/Record request configurable to
use dynamic features Adds two new options to SIP peers allowing
them to specify features (dynamic or builtin) to use when sending
INFO/record requests. Recordonfeature activates whatever feature
is specified when recieving a record: on request while
recordofffeature activates whatever feature is specified when
receiving a record: off request. Both of these features can be
disabled by setting the feature to an empty string. (closes issue
ASTERISK-16507) Reported by: Jon Bright Review:
https://reviewboard.asterisk.org/r/1634/
* channels/chan_sip.c, configs/sip.conf.sample, CHANGES,
channels/sip/include/sip.h: chan_sip autocreatepeer=persist
option for auto-created peers to survive reload This patch moves
destruction of sip peers to immediately after the general section
of sip.conf is read so that autocreatepeer setting can be read
before deletion of peers. If autocreatepeer=persist at reload,
then peers created by the autocreatepeer setting will be skipped
when purging the current SIP peer list. (closes ASTERISK-16508)
Reported by: Kirill Katsnelson Patches:
017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill
Katsnelson (license 5845)
2011-12-23 17:36 +0000 [r349046] Sean Bright <sean@malleable.com>
* /, apps/app_chanspy.c: Merged revisions 349045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r349045 | seanbright | 2011-12-23 12:32:33 -0500
(Fri, 23 Dec 2011) | 25 lines Merged revisions 349044 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec
2011) | 18 lines In ChanSpy, don't create audiohooks that will
never be used. When ChanSpy is initialized it creates and
attaches 3 audiohooks: 1) Read audio off of the channel that we
are spying on 2) Write audio to the channel that we are spying on
3) Write audio to the channel that is bridged to the channel that
we are spying on. The first is always necessary, but the others
are used only when specific options are passed to the ChanSpy
application (B, d, w, and W to be specific). When those flags are
not passed, neither of those audiohooks are ever sent frames, but
we still try to process the hooks for each voice frame that we
recieve on the channel. So in short - only create and attach
audiohooks that we actually need. ........ ................
2011-12-23 15:26 +0000 [r348994] Kinsey Moore <kmoore@digium.com>
* apps/app_dial.c, /: Fix missing doc tags found while fixing
ASTERISK-18689 Add missing <variable></variable> tags in app_dial
documentation. ........ Merged revisions 348992 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348993 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-23 02:35 +0000 [r348953] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c, /, channels/chan_sip.c, include/asterisk/pbx.h: Fix
extension state callback references in chan_sip. Chan_sip gives a
dialog reference to the extension state callback and assumes that
when ast_extension_state_del() returns, the callback cannot
happen anymore. Chan_sip then reduces the dialog reference count
associated with the callback. Recent changes (ASTERISK-17760)
have resulted in the potential for the callback to happen after
ast_extension_state_del() has returned. For chan_sip, this could
be very bad because the dialog pointer could have already been
destroyed. * Added ast_extension_state_add_destroy() so chan_sip
can account for the sip_pvt reference given to the extension
state callback when the extension state callback is deleted. *
Fix pbx.c awkward statecbs handling in
ast_extension_state_add_destroy() and handle_statechange() now
that the struct ast_state_cb has a destructor to call. * Ensure
that ast_extension_state_add_destroy() will never return -1 or 0
for a successful registration. * Fixed pbx.c statecbs_cmp() to
compare the correct information. The passed in value to compare
is a change_cb function pointer not an object pointer. * Make
pbx.c ast_merge_contexts_and_delete() not perform callbacks with
AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for
deadlocking when those locks are held during the callback. *
Removed unused lock declaration for the pbx.c store_hints list.
(closes issue ASTERISK-18844) Reported by: rmudgett Review:
https://reviewboard.asterisk.org/r/1635/ ........ Merged
revisions 348940 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348952 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-22 22:39 +0000 [r348890] Matthew Jordan <mjordan@digium.com>
* cel/cel_pgsql.c, /: Fix for memory leaks / cleanup in cel_pgsql
There were a number of issues in cel_pgsql's pgsql_log method: *
If either sql or sql2 could not be allocated, the method would
return while the pgsql_lock was still locked * If the execution
of the log statement succeeded, the sql and sql2 structs were
never free'd * Reconnection successes were logged as ERRORs. In
general, the severity of several logging statements was reduced
(closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested
by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/
........ Merged revisions 348888 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348889 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-22 21:12 +0000 [r348849] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Fix segfault on answer. Only
update/change RTP source if RTP has already been started and
connected to the subchannel.
2011-12-22 20:44 +0000 [r348848] Matthew Jordan <mjordan@digium.com>
* /, main/say.c, main/file.c, main/app.c, apps/app_confbridge.c,
main/bridging.c: Add Asterisk TestSuite event hooks to support
ConfBridge testing This patch adds initial testsuite event hooks
so that ConfBridge tests can be executed in the Asterisk
TestSuite. (issue ASTERISK-19059) ........ Merged revisions
348846 from http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-22 20:39 +0000 [r348847] Terry Wilson <twilson@digium.com>
* /, include/asterisk/format_pref.h: Allow packetization vaules >
127 According to the RTP packetization documentation, and the
maximum values listed in AST_FORMAT_LIST, we should support
values > that the signed char array that ast_codec_pref makes
available to store the value. All places in the code treat the
framing field as though it were an int array instaead of a char
array anyway, so this just fixes the type of the array. (closes
issue ASTERISK-18876) Review:
https://reviewboard.asterisk.org/r/1639/ ........ Merged
revisions 348833 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348845 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-21 20:13 +0000 [r348737-348794] Richard Mudgett <rmudgett@digium.com>
* /, codecs/speex: Make codecs/speex ignore *.i files also.
........ Merged revisions 348793 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/confbridge: Make apps/confbridge ignore *.i files also.
........ Merged revisions 348790 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS
number if it is blank. Some ISDN switches complain or block the
call if the RDNIS number is empty. * Made chan_iax2 not save a
RDNIS number into the ast_channel if the string is blank. This is
what other channel drivers do. (closes issue ASTERISK-17152)
Reported by: rmudgett ........ Merged revisions 348735 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348736 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-20 20:06 +0000 [r348698] Matthew Nicholson <mnicholson@digium.com>
* contrib/scripts/safe_asterisk: This adds support for setting
several safe_asterisk parameters using environment variables and
also enables a custom run directory for asterisk (instead of
defaulting to /tmp). Patch by: Byron Clark (byronclark) (closes
ASTERISK-17810)
2011-12-19 21:43 +0000 [r348649] Richard Mudgett <rmudgett@digium.com>
* /, configure, configure.ac: Fix crashes on other platforms caused
by interference from Darwin weak symbol support. Support weak
symbols on a platform specific basis. The Mac OS X (Darwin)
support must be isolated from the other platforms because it has
caused other platforms to crash. Several other platforms
including Linux have GCC versions that define the weak attribute.
However, this attribute is only setup for use in the code by
Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang
Review: https://reviewboard.asterisk.org/r/1617/ ........ Merged
revisions 348647 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348648 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-19 19:55 +0000 [r348606] Leif Madsen <leif@leifmadsen.com>
* /, main/message.c: Update documentation for MESSAGE_SEND_STATUS
variable. (Closes issue ASTERISK-19056) Reported by: Yuri
Patches: 348360.diff uploaded by Yuri (license #5242) ........
Merged revisions 348605 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-19 01:36 +0000 [r348567] Terry Wilson <twilson@digium.com>
* /, res/res_srtp.c: Add a separate buffer for SRTCP packets The
function ast_srtp_protect used a common buffer for both SRTP and
SRTCP packets. Since this function can be called from multiple
threads for the same SRTP session (scheduler for SRTCP and
channel for SRTP) it was possible for the packets to become
corrupted as the buffer was used by both threads simultaneously.
This patch adds a separate buffer for SRTCP packets to avoid the
problem. (closes issue ASTERISK-18889, Reported/patch by Daniel
Collins) ........ Merged revisions 347995 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347996 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-18 18:29 +0000 [r348518] Kevin P. Fleming <kpfleming@digium.com>
* /, configs/sip.conf.sample: Correct two flaws in sip.conf.sample
related to AST-2011-013. * The sample file listed *two* values
for the 'nat' option as being the default. Only 'force_rport' is
the default. * The warning about having differing 'nat' settings
confusingly referred to both peers and users. ........ Merged
revisions 348515 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
Merged revisions 348516 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348517 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-16 23:58 +0000 [r348466] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /, main/features.c: Clean-up on isle five for
__ast_request_and_dial() and ast_call_forward(). * Add locking
when a channel inherits variables and datastores in
__ast_request_and_dial() and ast_call_forward(). Note: The
involved channels are not active so there was minimal potential
for problems. * Remove calls to ast_set_callerid() in
__ast_request_and_dial() and ast_call_forward() because the set
information is for the wrong direction. * Don't use C++ keywords
for variable names in ast_call_forward(). * Run the redirecting
interception macro if defined when forwarding a call in
ast_call_forward(). Note: Currently will never execute because
the only callers that supply a calling channel supply a hungup or
zombie channel. * Make feature_request_and_dial() put the
transferee into autoservice when it calls ast_call_forward() in
case a redirection interception macro is run. Note: Currently
will never happen because the caller channel (Party B) is always
hungup at this time. * Make feature_request_and_dial() ignore the
AST_CONTROL_PROCEEDING frame to silence a log message. ........
Merged revisions 348464 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348465 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-16 22:00 +0000 [r348416] Jonathan Rose <jrose@digium.com>
* configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c:
Voicemail with the saycid option will now play a caller's name
based on cid if available. In order to check the availability of
the caller's name, app_voicemail will check for an audio file in
<astspooldir>/recordings/callerids/ This change sets a precedent
for where to put recordings of names. Currently the idea is that
recordings here could also be used for applications like
confbridge and meetme to find recorded names in this folder from
callerid (when another recording isn't available) (closes issue
ASTERISK-18565) Reporter: Russell Brown Patches: r uploaded by
Russel Brown (license 6182)
2011-12-16 21:30 +0000 [r348312-348408] Richard Mudgett <rmudgett@digium.com>
* main/channel.c, /: Fix cut and past error in ast_call_forward().
(issue ASTERISK-18836) ........ Merged revisions 348401 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348405 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/channel.c, main/pbx.c, /, apps/app_authenticate.c,
funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h,
apps/app_followme.c, apps/app_queue.c, res/res_monitor.c: Fix
crash during CDR update. The ast_cdr_setcid() and
ast_cdr_update() were shown in ASTERISK-18836 to be called by
different threads for the same channel. The channel driver thread
and the PBX thread running dialplan. * Add lock protection around
CDR API calls that access an ast_channel pointer. (closes issue
ASTERISK-18836) Reported by: gpluser Review:
https://reviewboard.asterisk.org/r/1628/ ........ Merged
revisions 348362 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348363 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_parkandannounce.c: Fix ParkAndAnnounce to pass the
CallerID to the announcing channel. ParkAndAnnounce tried to pass
the CallerID to the announcing channel but the ID was wiped out
by the channel masquerade done when parking the call. * Save the
CallerID before parking the channel to pass it to the announcing
channel. * Fixed a minor memory leak in ParkAndAnnounce. *
Updated some ParkAndAnnounce log messages. ........ Merged
revisions 348310 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348311 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-14 22:36 +0000 [r348215-348266] Matthew Jordan <mjordan@digium.com>
* /, apps/app_originate.c: Added support for all slin formats to
app_originate Previously, app_originate could not originate a
call into a non-8kHz conference bridge as the formats for
non-8kHz slin codecs were not applied to the created channel.
This patch adds all of the formats by default, such that if a
created channel has a codec that supports a higher sampling rate,
a translation path can be built between it and other channels.
........ Merged revisions 348265 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_queue.c: Fixed Asterisk crash when function
QUEUE_MEMBER receives invalid input The function QUEUE_MEMBER has
two required parameters (queuename, option). It was only checking
for the presence of queuename. The patch checks for the existence
of the option parameter and provides better error logging when
invalid values are provided for the option parameter as well.
........ Merged revisions 348211 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-14 22:05 +0000 [r348214] Matthew Nicholson <mnicholson@digium.com>
* /, res/res_fax.c: Don't clear LOCALSTATIONID before sending or
receiving. The user may set that variable. ASTERISK-18921
........ Merged revisions 348212 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348213 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-14 21:08 +0000 [r348161] Jonathan Rose <jrose@digium.com>
* main/features.c, configs/features.conf.sample: Add and document
PARKEDCALL variable set during timeout PARKEDCALL variable tracks
which parking lot the call was last parked in. This can be used
afterwards for flow control when returntoorigin is set to off. I
went ahead and documented both this and the existing variable set
during timeout (PARKINGSLOT) in the sample features.conf since
there was no prior mention of variables being set during timeout.
(closes issue ASTERISK-16239) Reported By: Clod Patry Patches:
M17503.diff uploaded by Clod Patry (license 5138)
2011-12-14 20:51 +0000 [r348160] Matthew Jordan <mjordan@digium.com>
* apps/app_confbridge.c: Improve error message in CONFBRIDGE_INFO
Provided a more descriptive error message when a value supplied
for the parameter type is not one of the acceptable values.
(closes issue ASTERISK-18717) Reported by: Paul Belanger Patches:
__20111103-better-confbridge_info-error-msg.txt (License #4999)
2011-12-14 20:37 +0000 [r348156-348159] Jonathan Rose <jrose@digium.com>
* /, configs/features.conf.sample: Fix accidental use of tabs
instead of spaces from previous features.conf.sample change
........ Merged revisions 348157 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348158 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, configs/features.conf.sample: Document PARKINGSLOT variable in
features.conf.sample (issue ASTERISK-16239) ........ Merged
revisions 348154 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348155 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-13 23:10 +0000 [r348103] Richard Mudgett <rmudgett@digium.com>
* /, bridges/bridge_builtin_features.c, apps/app_followme.c: Fix
FollowMe CallerID on outgoing calls. The addition of the
Connected Line support changed how CallerID is passed to outgoing
calls. The FollowMe application was not updated to pass CallerID
to the outgoing calls. * Fix FollowMe CallerID on outgoing calls.
* Restructured findmeexec() to fix several memory leaks and
eliminate some duplicated code. * Made check the return value of
create_followme_number(). Putting a NULL into the numbers list is
bad if create_followme_number() fails. * Fixed a couple uses of
ast_strdupa() inside loops. * The changes to
bridge_builtin_features.c fix a similar CallerID issue with the
bridging API attended and blind transfers. (Not used at this
time.) (closes issue ASTERISK-17557) Reported by: hamlet505a
Tested by: rmudgett Review:
https://reviewboard.asterisk.org/r/1612/ ........ Merged
revisions 348101 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348102 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-13 15:22 +0000 [r348061] Stefan Schmidt <sst@sil.at>
* channels/chan_sip.c: Fix possible misshandling of an incoming SIP
response as a peer poke response. Also make sure peer has even
qualify enabled when handle a peer poke response. (closes issue
ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and
UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed
by: David Vossel ........ Merged revisions 348048 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 348056 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-12 19:35 +0000 [r347997] Matthew Jordan <mjordan@digium.com>
* include/asterisk/logger.h, utils/refcounter.c, main/logger.c,
utils/hashtest.c, UPGRADE.txt, utils/ael_main.c,
utils/hashtest2.c, CHANGES, main/asterisk.c, main/config.c,
configs/logger.conf.sample, main/loader.c, main/cli.c: Backed out
core changes from r346391 During testing, it was discovered that
there were a number of side effects introduced by r346391 and
subsequent check-ins related to it (r346429, r346617, and
r346655). This included the /main/stdtime/ test 'hanging', as
well as the remote console option failing to receive the
appropriate output after a period of time. I only backed out the
changes to main/ and utils/, as this was adequate to reverse the
behavior experienced. (issue ASTERISK-18974)
2011-12-12 17:34 +0000 [r347954] Richard Mudgett <rmudgett@digium.com>
* configs/iax.conf.sample, configs/chan_dahdi.conf.sample, /,
configs/chan_ooh323.conf.sample, configs/vpb.conf.sample,
configs/extensions.lua.sample, configs/sip.conf.sample,
configs/extensions.conf.sample: Update sample configs to put
incoming calls into context public. * Add warning about the SIP
allowguest option in context public. (closes issue
ASTERISK-14122) Reported by: Alec Davis Review:
https://reviewboard.asterisk.org/r/719/ ........ Merged revisions
347953 from http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-09 21:47 +0000 [r347866-347903] Jonathan Rose <jrose@digium.com>
* apps/app_mixmonitor.c: Adds MixMonitor and StopMixMonitor AMI
commands to the manager These commands work much like the
dialplan applications that would otherwise invoke them. A nice
benefit of these is that they can be invoked on a call remotely
and at any time during a call. They work much like the Monitor
and StopMonitor ami commands. (closes issue ASTERISK-17726)
Reported by: Sergio González Martín Patches:
mixmonitor_actions.diff uploaded by Sergio González Martín
(license 5644) Review: https://reviewboard.asterisk.org/r/1193/
* include/asterisk/file.h, apps/app_sayunixtime.c, CHANGES: Remove
autojump extensions from SayUnixTime, make an option to perform
automatic jumps. When a caller sends DTMF while the SayUnixTime
application is saying the time, The call would jump to the next
extension much like it does during Background(). This patch adds
option 'j' to SayUnixTime which when used employs the old
behavior. Also, this patch allows arguments to sayunixtime to not
be used as empty strings in the case of something like
'sayunixtime(,,,j)' or 'sayunixtime(,,pattern). (closes issue
ASTERISK-16675) Reported by: jlpedrosa Patches:
patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license
5959) Review: https://reviewboard.asterisk.org/r/956/
2011-12-09 01:33 +0000 [r347813] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c, /: Fix some parsing issues in
add_exten_to_pattern_tree(). * Simplify compare_char() and avoid
potential sign extension issue. * Fix infinite loop in
add_exten_to_pattern_tree() handling of character set escape
handling. * Added buffer overflow checks in
add_exten_to_pattern_tree() character set collection. * Made
ignore empty character sets. * Added escape character handling to
end-of-range character in character sets. This has a slight
change in behavior if the end-of-range character is an escape
character. You must now escape it. * Fix potential sign extension
issue when expanding character set ranges. * Made remove
duplicated characters from character sets. The duplicate
characters lower extension matching priority and prevent
duplicate extension detection. * Fix escape character handling
when the escape character is trying to escape the end-of-string.
We could have continued processing characters after the end of
the exten string. We could have added the previous character to
the pattern matching tree incorrectly. (closes issue
ASTERISK-18909) Reported by: Luke-Jr ........ Merged revisions
347811 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 347812 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-08 21:32 +0000 [r347735] Walter Doekes <walter+asterisk@wjd.nu>
* /, channels/chan_sip.c: Fix regression when using tcpenable=no
and tlsenable=yes. The tlsenable settings are tucked away in
main/tcptls.c, so I missed them when resolving ASTERISK-18837.
This should resolve the test suite breakage of the sip tls tests.
Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt
Jordan ........ Merged revisions 347718 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347727 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-08 20:55 +0000 [r347658] Jonathan Rose <jrose@digium.com>
* /, apps/app_queue.c: Fix regressed behavior of queue set penalty
to work without specifying 'in <queuename>' r325483 caused a
regression in Asterisk 10+ that would make Asterisk segfault when
attempting to set penalty on an interface without specifying a
queue in the queue set penalty CLI command. In addition, no
attempt would be made whatsoever to perform the penalty setting
on all the queues in the core list with either the cli command or
the non-segfaulting ami equivalent. This patch fixes that and
also makes an attempt to document and rename some functions
required by this command to better represent what they actually
do. Oh yeah, and the use of this command without specifying a
specific queue actually works now. Review:
https://reviewboard.asterisk.org/r/1609/ ........ Merged
revisions 347656 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-08 17:55 +0000 [r347601] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Mark channel running the h exten with the
soft-hangup flag. When a bridge is broken, ast_bridge_call()
might execute the h exten on the calling channel. However, that
channel may not have been the channel that broke the bridge by
hanging up. The channel executing the h exten must be in a hung
up state so things like AGI run in the correct mode. * Make sure
ast_bridge_call() marks the channel it is executing the h exten
on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as
to match the pbx.c main dialplan execution loop when it executes
the h exten.) (closes issue ASTERISK-18811) Reported by: David
Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621)
patch uploaded by rmudgett Tested by: David Hajek, rmudgett
........ Merged revisions 347595 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347600 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-08 16:24 +0000 [r347533] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Don't crash on INFO automon request with
no channel AST-2011-014. When automon was enabled in
features.conf, it was possible to crash Asterisk by sending an
INFO request if no channel had been created yet. (closes issue
ASTERISK-18805) ........ Merged revisions 347530 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
Merged revisions 347531 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347532 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-08 06:59 +0000 [r347490] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Fix segfault on answer. Fix a segfault if
an attempt to answer a call is made between when the inbound call
gives up (and the channel is removed) and when the device is
notified and removes the call from the device.
2011-12-07 21:42 +0000 [r347440] Richard Mudgett <rmudgett@digium.com>
* main/manager.c, /: Update AMI Getvar and Setvar documentation
about supplying a channel name. (closes issue ASTERISK-18958)
Reported by: Red Patches: jira_asterisk_18958_v1.8.patch (license
#5621) patch uploaded by rmudgett ........ Merged revisions
347438 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 347439 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-07 20:34 +0000 [r347395] Jonathan Rose <jrose@digium.com>
* /, apps/app_meetme.c: Fix: Meetme recording variables from
realtime DB use null entries over channel variables Meetme would
attempt to substitute the realtime values of RECORDING_FILE and
RECORDING_FORMAT from the meetme db entry instead of using the
channel variable set for those variables in spite of those
database entries being NULL or even lacking a column to represent
them. (closes issue ASTERISK-18873) Reported by: Byron Clark
Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license
6157) ........ Merged revisions 347369 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347383 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-07 20:15 +0000 [r347345] Terry Wilson <twilson@digium.com>
* Makefile, include/asterisk/paths.h, /,
configs/asterisk.conf.sample, build_tools/make_defaults_h,
main/asterisk.c, main/db.c: Add ASTSBINDIR to the list of
configurable paths This patch also makes astdb2sqlite3 and
astcanary use the configured directory instead of relying on
$PATH. (closes issue ASTERISK-18959) Review:
https://reviewboard.asterisk.org/r/1613/ ........ Merged
revisions 347344 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-06 23:58 +0000 [r347294] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c: Make SIP INFO messages for dtmf-relay
signals case insensitive. (closes issue ASTERISK-18924) Reported
by: Kevin Taylor ........ Merged revisions 347292 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347293 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-06 22:01 +0000 [r347241] Jonathan Rose <jrose@digium.com>
* main/pbx.c, /: Documents CHANNEL(musicclass) taking priority over
m([x]) in waitExten If waitExten specifies a music class to use
with its music on hold option, it will use CHANNEL(musicclass)
instead if that channel variable has been set on the initiating
channel. This documents that behavior in the waitExten app so
that this can be known without checking the documentation of the
code in function local_ast_moh_start. (closes issue
ASTERISK-18804) ........ Merged revisions 347239 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347240 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-06 20:23 +0000 [r347157-347192] Walter Doekes <walter+asterisk@wjd.nu>
* UPGRADE.txt, CHANGES, apps/app_voicemail.c: Add VM_INFO()
dialplan function to gather information about a mailbox.
Deprecates MAILBOX_EXISTS. Provides count, email, exists,
fullname, language, locale, pager, password, tz. (closes issue
ASTERISK-18634) Patch by: Kris Shaw Review:
https://reviewboard.asterisk.org/r/1568 Reviewed by: Walter
Doekes
* /, channels/chan_sip.c: Don't allow transport=tcp when
tcpenable=no. When tcpenable=no, sending to transport=tcp hosts
was still allowed. Resolving the source address wasn't possible
and yielded the string "(null)" in SIP messages. Fixed that and a
couple of not-so-correct log messages. (closes issue
ASTERISK-18837) Reported by: Andreas Topp Review:
https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan
........ Merged revisions 347166 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347167 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_voicemail.c: Add regression tests for issue
ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572
Reviewed by: Matt Jordan ........ Merged revisions 347131 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347146 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_voicemail.c: The voicemail [general] zonetag and
locale variables weren't loaded until after the mailboxes were
initialized. This caused the settings to be unset for those
mailboxes until a reload was performed. (closes issue
ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570
Reviewed by: Matt Jordan ........ Merged revisions 347111 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347124 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-06 19:09 +0000 [r347110] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/dlinkedlists.h, tests/test_linkedlists.c: Doubly
linked lists unit test and update to implementation. Update the
doubly linked list implementation. Now safe traversing can insert
before and after the current node when traversing in either
direction. Updated the linked lists unit test test_linkedlist to
also test doubly linked lists. The old test_dlinkedlist requires
a manual check of results and probably should be removed. Review:
https://reviewboard.asterisk.org/r/1569/
2011-12-06 17:34 +0000 [r347069] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Fixed crash from orphaned MWI
subscriptions in chan_sip This patch resolves the issue where MWI
subscriptions are orphaned by subsequent SIP SUBSCRIBE messages.
When a peer is removed, either by pruning realtime SIP peers or
by unloading / loading chan_sip, the MWI subscriptions that were
orphaned would still be on the event engine list of valid
subscriptions but have a pointer to a peer that no longer was
valid. When an MWI event would occur, this would cause a seg
fault. (closes issue ASTERISK-18663) Reported by: Ross Beer
Tested by: Ross Beer, Matt Jordan Patches:
blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)
Review: https://reviewboard.asterisk.org/r/1610/ ........ Merged
revisions 347058 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347068 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-05 17:44 +0000 [r347008] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /,
channels/sig_analog.h: Restore call progress code for analog
ports. Extracting sig_analog from chan_dahdi lost call progress
detection functionality. * Fix analog ports from considering a
call answered immediately after dialing has completed if the
callprogress option is enabled. (closes issue ASTERISK-18841)
Reported by: Richard Miller Patches: chan_dahdi.diff (license
#5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.c.diff (license #5685) patch uploaded by Richard
Miller (Modified by me) sig_analog.h.diff (license #5685) patch
uploaded by Richard Miller ........ Merged revisions 347006 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 347007 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-05 15:04 +0000 [r346956] Jonathan Rose <jrose@digium.com>
* main/pbx.c, /: Resolve duplicate label used in multiple
priorities for the same extension. Prior to this patch, if labels
with the same name were used for different priorities in the same
extension, the new label would be accepted, but it would be
unusable since attempts to reach that label would just go to the
first one. Now pbx.c detects this, generates a warning in logs,
and culls the label before adding it to the dialplan. (closes
issue ASTERISK-18807) Reported by: Kenneth Shumard Patches:
pbx.c.patch uploaded by Kenneth Shumard (License 5077) ........
Merged revisions 346954 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 346955 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-05 14:47 +0000 [r346953] Kinsey Moore <kmoore@digium.com>
* res/res_jabber.exports.in, /: Fix chan_jingle/gtalk load
regression introduced in r346087 Add missing symbol exports for
ast_aji_client_destroy and ast_aji_buddy_destroy for usage
outside res_jabber. Testing of these changes focused on
res_jabber itself, so this problem was missed. Reported-by:
Michael Spiceland ........ Merged revisions 346951 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 346952 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-04 10:08 +0000 [r346901] Walter Doekes <walter+asterisk@wjd.nu>
* /, channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and
domain ACL bypass. The code that allowed admins to create users
with domain-only uri's had stopped to work in 1.8 because of the
reqresp parser rewrites. This is fixed now: if you have a
[mydomain.com] sip user, you can register with useraddr
sip:mydomain.com. Note that in that case -- if you're using
domain ACLs (a configured domain list) -- mydomain.com must be in
the allow list as well. Reviewboard r1606 shows a list of
registration combinations and which SIP response codes are
returned. Review: https://reviewboard.asterisk.org/r/1533/
Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes
issue ASTERISK-18741) ........ Merged revisions 346899 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 346900 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-02 23:30 +0000 [r346857] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Update SIP MESSAGE To parsing to
correctly handle URI The previous patch (r346040) incorrectly
parsed the URI in the presence of a port, e.g.,
user@hostname:port would fail as the port would be double
appended to the SIP message. This patch uses the parse_uri
function to correctly parse the URI into its username and
hostname parts, and places them in the correct fields in the
sip_pvt structure. (issue ASTERISK-18903) Review:
https://reviewboard.asterisk.org/r/1597/ ........ Merged
revisions 346856 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-02 19:40 +0000 [r346777-346816] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c: implement nat option for rtp channels with
ooh323
* addons/chan_ooh323.c, /, channels/chan_h323.c: Merged revisions
346763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r346763 | may | 2011-12-02 20:42:32 +0400 (Fri,
02 Dec 2011) | 14 lines Merged revisions 346762 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7
lines process null frame pointer returned by
ast_rtp_instance_read correctly (closes issue ASTERISK-16697)
Reported by: under Patches: segfault.diff (License #5871) patch
uploaded by under ........ ................
2011-12-01 21:19 +0000 [r346709] Richard Mudgett <rmudgett@digium.com>
* main/stun.c, /, res/res_stun_monitor.c,
configs/res_stun_monitor.conf.sample, include/asterisk/stun.h:
Re-resolve the STUN address if a STUN poll fails for
res_stun_monitor. The STUN socket must remain open between polls
or the external address seen by the STUN server is likely to
change. However, if the STUN request poll fails then the STUN
server address needs to be re-resolved and the STUN socket needs
to be closed and reopened. * Re-resolve the STUN server address
and create a new socket if the STUN request poll fails. * Fix
ast_stun_request() return value consistency. * Fix
ast_stun_request() to check the received packet for expected
message type and transaction ID. * Fix ast_stun_request() to read
packets until timeout or an associated response packet is found.
The stun_purge_socket() hack is no longer required. * Reduce
ast_stun_request() error messages to debug output. * No longer
pass in the destination address to ast_stun_request() if the
socket is already bound or connected to the destination. (closes
issue ASTERISK-18327) Reported by: Wolfram Joost Tested by:
rmudgett Review: https://reviewboard.asterisk.org/r/1595/
........ Merged revisions 346700 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 346701 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-12-01 20:46 +0000 [r346699] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180
ringing. 183 Ringing isn't even a thing. 183 is actually a
session progress message. (closes issue ASTERISK-18925) Reported
by: Sebastian Denz Tested by: jrose Patches:
asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian
Denz (License #6139) ........ Merged revisions 346697 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 346698 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-30 23:38 +0000 [r346617-346655] Tilghman Lesher <tilghman@meg.abyt.es>
* channels/chan_unistim.c, main/tcptls.c, channels/chan_sip.c,
main/config.c, main/loader.c: Remove the few places where we try
to ast_verbose() without a newline.
* main/asterisk.c: Fix edge case for overflow buffer.
2011-11-30 22:03 +0000 [r346525-346566] Jonathan Rose <jrose@digium.com>
* main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h:
r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) |
18 lines Cleaning up chan_sip/tcptls file descriptor closing.
This patch attempts to eliminate various possible instances of
undefined behavior caused by invoking close/fclose in situations
where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that
don't have a valid index for fd (-1). Thanks for more than a
little help from wdoekes. (closes issue ASTERISK-18700) Reported
by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane
Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas
Review: https://reviewboard.asterisk.org/r/1576/ ........ Merged
revisions 346564 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 346565 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h:
Reverting 346525 due to accidental patch against trunk instead of
1.8
* main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h:
Cleaning up chan_sip/tcptls file descriptor closing. This patch
attempts to eliminate various possible instances of undefined
behavior caused by invoking close/fclose in situations where
fclose may have already been issued on a tcptls_session_instance
and/or closing file descriptors that don't have a valid index for
fd (-1). Thanks for more than a little help from wdoekes. (closes
issue ASTERISK-18700) Reported by: Erik Wallin (issue
ASTERISK-18345) Reported by: Stephane Cazelas (issue
ASTERISK-18342) Reported by: Stephane Chazelas Review:
https://reviewboard.asterisk.org/r/1576/
2011-11-30 19:37 +0000 [r346474] Leif Madsen <leif@leifmadsen.com>
* configs/queues.conf.sample: Update queues.conf.sample
documentation. Update the documentation surrounding the use of
MONITOR_EXEC to make it more clear that it can be used for both
Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413)
Reported by: David Woolley Patches:
issue18817_mixmonitor_queues_doc.diff by Michael L. Young
(License #5026) ........ Merged revisions 346472 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 346473 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-29 20:32 +0000 [r346391-346429] Tilghman Lesher <tilghman@meg.abyt.es>
* utils/refcounter.c, utils/hashtest.c, utils/ael_main.c,
utils/hashtest2.c: Fix compilation of utilities (caught by
Bamboo).
* addons/chan_ooh323.c, channels/chan_sip.c, main/say.c,
res/res_fax.c, UPGRADE.txt, res/res_musiconhold.c,
res/res_jabber.c, CHANGES, configs/logger.conf.sample,
main/cli.c, channels/chan_usbradio.c, include/asterisk/logger.h,
main/dial.c, channels/chan_skinny.c, main/logger.c,
codecs/codec_dahdi.c, apps/app_rpt.c, apps/app_verbose.c,
main/asterisk.c, main/bridging.c, res/res_clialiases.c,
addons/res_config_mysql.c, apps/app_voicemail.c: Allow each
logging destination and console to have its own notion of the
verbosity level. Review: https://reviewboard.asterisk.org/r/1599
2011-11-29 00:03 +0000 [r346350] David Vossel <dvossel@digium.com>
* /, include/asterisk/message.h, main/message.c: Merged revisions
346349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011)
| 10 lines Fixes memory leak in message API. The ast_msg_get_var
function did not properly decrement the ref count of the var it
retrieves. The way this is implemented is a bit tricky, as we
must decrement the var and then return the var's value. As long
as the documentation for the function is followed, this will not
result in a dangling pointer as the ast_msg structure owns its
own reference to the var while it exists in the var container.
........
2011-11-28 14:34 +0000 [r346294] Stefan Schmidt <sst@sil.at>
* res/res_rtp_asterisk.c, /: Fix regression that 'rtp/rtcp set
debup ip' only works when also a port was specified. (closes
issue ASTERISK-18693) Reported by: Davide Dal Fra Review:
https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter
Doekes ........ Merged revisions 346292 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 346293 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-23 23:03 +0000 [r346241] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/acl.h, /, channels/chan_skinny.c,
channels/chan_h323.c, main/acl.c, channels/chan_iax2.c: Fix calls
to ast_get_ip() not initializing the address family. ........
Merged revisions 346239 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 346240 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-23 20:48 +0000 [r346146-346199] Walter Doekes <walter+asterisk@wjd.nu>
* /, channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text()
function. In r116240, get_msg_text() got an extra parameter to
fix the unwanted addition of trailing newlines to SIP MESSAGE
bodies. This caused all linefeeds to be trimmed, which isn't
right either. This is a stop-gap; the right fix is to return the
original SIP request body. Review:
https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan
........ Merged revisions 346147 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 346198 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, include/asterisk/strings.h: Fix ast_str_truncate signedness
warning and documentation. Review:
https://reviewboard.asterisk.org/r/1594 ........ Merged revisions
346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 346145 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-23 17:16 +0000 [r346088] Kinsey Moore <kmoore@digium.com>
* channels/chan_jingle.c, /, include/asterisk/jabber.h,
channels/chan_gtalk.c, res/res_jabber.c: Fix res_jabber resource
leaks This should fix almost all resource leaks in res_jabber
that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous
situation where ast_aji_get_client would sometimes bump an
object's refcount and sometimes not. Review:
https://reviewboard.asterisk.org/r/1553 ........ Merged revisions
346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 346087 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-23 16:23 +0000 [r346053] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Fixed SendMessage stripping extension
from To: header in SIP MESSAGE When using the MessageSend
application to send a SIP MESSAGE to a non-peer, chan_sip
attempted to validate the hostname or IP Address. In the process,
it stripped off the extension and failed to add it back to the
sip_pvt structure before transmitting. This patch adds the full
URI passed in from the message core to the sip_pvt structure.
(closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by:
Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/
........ Merged revisions 346040 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-23 16:12 +0000 [r346033] Terry Wilson <twilson@digium.com>
* /, res/res_musiconhold.c: Resume playing existing hold music for
cached realtime MOH As a result of the fix for ASTERISK-18039,
realtime caching MOH no longer properly resumes playing back a
file between different holds in the same call. This is because
scanning for new files causes the existing file array to be
emptied and we were just comparing that the saved pointer to the
filename matched the pointer to the filename in a particular
position in the array. An easy fix is to save the filename
instead of a pointer to it and then do a strcmp instead of
comparing the addresses. (closes issue ASTERISK-18912) Review:
https://reviewboard.asterisk.org/r/1596/ ........ Merged
revisions 346030 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 346031 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-23 16:10 +0000 [r346032] Paul Belanger <paul.belanger@polybeacon.com>
* /, res/res_format_attr_silk.c, res/res_format_attr_celt.c: Added
support level for new modules ........ Merged revisions 346029
from http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-22 23:06 +0000 [r345978] Richard Mudgett <rmudgett@digium.com>
* main/dnsmgr.c, /, include/asterisk/dnsmgr.h: Fix dnsmgr entries
to ask for the same address family each time. The dnsmgr refresh
would always get the first address found regardless of the
original address family requested. So if you asked for only IPv4
addresses originally, you might get an IPv6 address on refresh. *
Saved the original address family requested by
ast_dnsmgr_lookup() to be used when the address is refreshed.
........ Merged revisions 345976 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345977 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-22 20:32 +0000 [r345925] Walter Doekes <walter+asterisk@wjd.nu>
* include/asterisk/logger.h, /: Clarify why the AST_LOG_* macros
exist next to the LOG_* macros. (issue ASTERISK-17973) ........
Merged revisions 345923 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345924 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-22 16:41 +0000 [r345883] Paul Belanger <paul.belanger@polybeacon.com>
* /, apps/confbridge/conf_config_parser.c: Add missing
sound_only_one config variable (closes issue ASTERISK-18895)
Reported by: zvision Patches: conf_config_parser.diff (license
#5755) patch uploaded by zvision ........ Merged revisions 345882
from http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-21 21:09 +0000 [r345831] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Default
to nat=yes; warn when nat in general and peer differ It is
possible to enumerate SIP usernames when the general and
user/peer nat settings differ in whether to respond to the port a
request is sent from or the port listed for responses in the Via
header. In 1.4 and 1.6.2, this would mean if one setting was
nat=yes or nat=route and the other was either nat=no or
nat=never. In 1.8 and 10, this would mean when one was
nat=force_rport and the other was nat=no. In order to address
this problem, it was decided to switch the default behavior to
nat=yes/force_rport as it is the most commonly used option and to
strongly discourage setting nat per-peer/user when at all
possible. For more discussion of the issue, please see:
http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html
(closes issue ASTERISK-18862) Review:
https://reviewboard.asterisk.org/r/1591/ ........ Merged
revisions 345776 from
http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged
revisions 345800 from
http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........
Merged revisions 345828 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345830 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-21 16:40 +0000 [r345735] Paul Belanger <paul.belanger@polybeacon.com>
* CHANGES, main/config.c: Add #tryinclude statement This provides
the same functionality as #include however an asterisk module
will still load if the filename does not exist. Review:
https://reviewboard.asterisk.org/r/1476/
2011-11-19 15:11 +0000 [r345643-345684] Tilghman Lesher <tilghman@meg.abyt.es>
* /, main/db.c: Update the documentation to better clarify how the
existing commands work. Review:
https://reviewboard.asterisk.org/r/1593/ ........ Merged
revisions 345682 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345683 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/db.c: Fix a change in behavior in 'database show' from
1.8. In 1.8 and previous versions, one could use any fullword
portion of the key name, including the full key, to obtain the
record. Until this patch, this did not work for the full key.
Closes issue ASTERISK-18886 Patch: by tilghman Review: by twilson
(http://pastebin.com/7rtu6bpk) on #asterisk-dev ........ Merged
revisions 345640 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-17 19:47 +0000 [r345560-345601] Matthew Jordan <mjordan@digium.com>
* contrib/realtime/mysql/sipfriends.sql (removed): Accidentally
readded sipfriends.sql in r345560. This was removed in r342871
* configs/confbridge.conf.sample,
apps/confbridge/include/confbridge.h, apps/app_confbridge.c,
CHANGES, contrib/realtime/mysql/sipfriends.sql (added),
apps/confbridge/conf_config_parser.c: Add admin toggle mute all
and participant count menu options to app_confbridge This patch
adds two new menu features to app_confbridge, admin_toggle_menu_
participants and participant_count. The admin action will
globally mute / unmute all non-admin participants on a
converence, while the participant count simply exposes the
existing participant count function to the conference bridge
menu. This also adds configuration options to change the sound
played when the conference is globally muted / unmuted, as well
as the necessary config hooks to place these functions in the
DTMF menus. (closes issue ASTERISK-18204) Reported by: Kevin
Reeves Tested by: Matt Jordan Patches:
app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt,
confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281)
Review: https://reviewboard.asterisk.org/r/1518/
2011-11-17 17:31 +0000 [r345559] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_pri.c: Remove dead code since pri_grab() can
never fail. Dead code makes programmers sick. I am sick of
looking at it. ........ Merged revisions 345546 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345558 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-16 14:56 +0000 [r345489] Jonathan Rose <jrose@digium.com>
* /, apps/app_voicemail.c: Guarantee messages go into the right
folders with multiple recipients Before, using the U flag in
Voicemail with multiple recipients would put urgent messages in
the INBOX folder for all users past the first thanks to a bug
with the message copying function. This would also cause messages
to fail to be sent if the INBOX directory hadn't been created for
that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt
Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1589/ ........ Merged
revisions 345487 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345488 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-15 20:11 +0000 [r345221-345433] Richard Mudgett <rmudgett@digium.com>
* /, res/res_agi.c: Make FastAGI HANGUP show up in AGI debug
output. * Change from using send() to ast_agi_send() so the
HANGUP shows up in the AGI debug output. (closes issue
ASTERISK-18723) Reported by: James Van Vleet Patches:
jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by
rmudgett ........ Merged revisions 345431 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345432 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/sig_pri.c: Fix typo in sig_pri using wrong structure
name. It is fortunate that the typo does not alter generated code
since the e->restart.channel and e->ring.channel members are in
the same position. (closes issue ASTERISK-18868) Reported by:
zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by
zvision ........ Merged revisions 345370 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345371 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_queue.c: Make queue log indicate if ADDMEMBER is
paused for AMI and realtime. * Add parameter to queue log
ADDMEMBER to indicate if the member is paused. (closes issue
ASTERISK-18645) Reported by: garlew Patches: paused.diff (License
#5337) patch uploaded by garlew Tested by: rmudgett, garlew
Review: https://reviewboard.asterisk.org/r/1469/ ........ Merged
revisions 345285 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345290 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c, configs/sip.conf.sample, UPGRADE-1.8.txt,
channels/sip/include/sip.h: Restore SIP DTMF overlap dialing
method. The recent fix for ASTERISK-17288 to get RFC3578 SIP
overlap support working correctly removed a long standing ability
to do overlap dialing using DTMF in the early media phase of a
call. See ASTERISK-18702 it has a very good description of the
issue. I started with Pavel Troller's chan_sip.diff patch on
issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf
allowoverlap config option. The new option value causes the
Incomplte application to not send anything with chan_sip so the
caller can supply more digits via DTMF. * Renames
SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
since that is what it really means. * Fixed get_destination()
inconsistency with the pickup extension matching. * Fixed
initialization of PAGE3 of global_flags in reload_config().
(closes issue ASTERISK-18702) Reported by: Pavel Troller Review:
https://reviewboard.asterisk.org/r/1517/ Review:
https://reviewboard.asterisk.org/r/1582/ ........ Merged
revisions 345273 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345275 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/pbx.c, /: Fix Progress spelling error in main/pbx.c. (closes
issue ASTERISK-18857) Reported by: David M Patches:
mainpbx-trivial.patch (License #6326) patch uploaded by David M
........ Merged revisions 345219 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345220 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-14 19:12 +0000 [r345165] Terry Wilson <twilson@digium.com>
* main/channel.c, /: Don't read past end of input when calling
write() int blah = 1; ... write(chan->alertpipe[1], &blah,
new_frames * sizeof(blah)) != (new_frames * sizeof(blah))) is
only valid when new_frames == 1. Otherwise we start reading into
adjacent variables declared on the stack. The read end discards
what is read, so the values don't matter but it's not a good idea
to read past where we want even though new_frames is almost
always 1 and should never be large. This patch is basically taken
out of kpfleming's eventfd branch, as he mentioned that he
remembered fixing it there when I talked to him about this issue.
Review: https://reviewboard.asterisk.org/r/1583/ ........ Merged
revisions 345163 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345164 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-14 19:03 +0000 [r345162] Walter Doekes <walter+asterisk@wjd.nu>
* /, channels/sip/include/reqresp_parser.h: Update reqresp_parser
parse_uri doxygen comments. The issue mentioned in the bug report
had been fixed recently by twilson. The reporter included this
documentation fix. (closes issue ASTERISK-18572) Reported by:
Richard Miller Patch by: Richard Miller (modified) ........
Merged revisions 345160 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345161 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-14 16:21 +0000 [r345120] Jonathan Rose <jrose@digium.com>
* /, apps/app_voicemail.c: Moves voicemail setup password entry to
the end of the setup process. This change was made because
forcegreeting and forcename settings in voicemail could be
circumvented by hanging up after entering a password, because the
only way voicemail currently observes whether a mailbox is new or
not is by checking to see if the password is the same as the
mailbox number or not. (closes issue ASTERISK-18282) Reported by:
Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/
........ Merged revisions 345062 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345117 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-14 15:11 +0000 [r345065] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Ensure that a null vmexten does not cause
a segfault When sip_send_mwi_to_peer was modified recently to
avoid deadlocks, vmexten was not expected to be null. This change
handles that situation to avoid a segfault. ........ Merged
revisions 345063 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 345064 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-14 01:25 +0000 [r345023] TransNexus OSP Development <support@transnexus.com>
* apps/app_osplookup.c: Increased max number of destinations.
2011-11-12 16:32 +0000 [r344979] Gregory Nietsky <gregory@distrotech.co.za>
* channels/chan_misdn.c, /: mISDN Round Robin break when no channel
is available Prevent channels been parsed repetitively. ........
Merged revisions 344965 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344966 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-12 00:36 +0000 [r344901] Terry Wilson <twilson@digium.com>
* /, res/res_musiconhold.c: Don't forget to rescan MOH files for
cached realtime classes Realtime MOH class caching was
implemented because without it, you would build a completely new
MOH class and would start the music over at the beginning each
time hold was pressed in a conversation. Unfortunately, this
broke re-scanning for file changes for realtime MOH classes. This
patch corrects that issue. (closes issue ASTERISK-18039) Review:
https://reviewboard.asterisk.org/r/1579/ ........ Merged
revisions 344899 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344900 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-11 22:00 +0000 [r344846] Walter Doekes <walter+asterisk@wjd.nu>
* include/asterisk/utils.h, /, main/utils.c,
include/asterisk/stringfields.h: Use __alignof__ instead of
sizeof for stringfield length storage. Kevin P Fleming suggested
that r343157 should use __alignof__ instead of sizeof. For most
systems this won't be an issue, but better fix it now while it's
still fresh. Review: https://reviewboard.asterisk.org/r/1573
........ Merged revisions 344843 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344845 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-11 21:57 +0000 [r344844] Matthew Jordan <mjordan@digium.com>
* /, main/file.c: Video format was treated as audio when removed
from the file playback scheduler This patch fixes the format type
check in ast_closestream and filestream_destructor. Previously a
comparison operator was used, but since audio formats are no
longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats
that have a value greater than the video formats), a bitwise AND
operation is used instead. Duplicated code was also moved to
filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo
Bedrij Tested by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1580/ ........ Merged
revisions 344823 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344842 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-11 21:37 +0000 [r344838-344840] Walter Doekes <walter+asterisk@wjd.nu>
* /, channels/sip/reqresp_parser.c: Remove unneeded if(params)
checks in reqresp_parser. Nick Lewis added them in
https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent
reason. There is no way that params could become NULL in that
piece of code, so I removed these excess checks again. ........
Merged revisions 344837 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344839 from
http://svn.asterisk.org/svn/asterisk/branches/10
* main/manager.c, /: Fix bad quoting of multiline mxml opaque_data
that caused invalid xml. The opaque_data was added and enclosed
in single quotes, assuming it would be only a single line. The
rest of the lines were appended after the closing quote. (closes
issue ASTERISK-18852) Reported by: peep_ on IRC Review:
https://reviewboard.asterisk.org/r/1577 ........ Merged revisions
344835 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 344836 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-11 20:15 +0000 [r344771] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Fix regression introduced by SDP fixups
If capability is adjusted when switching to UDPTL during fax
transmission, fax teardown fails. Make sure capability is only
touched if RTP is active. This regression was introduced in
R344385. ........ Merged revisions 344769 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344770 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-11 18:37 +0000 [r344663-344717] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c: Check sip.conf maxforwards parameter for
range 1 <= x <= 255. JIRA AST-710 ........ Merged revisions
344715 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 344716 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, main/cli.c: Make CLI "core show channel" not hold the channel
lock during console output. Holding the channel lock while the
CLI "core show channel" command is executing can slow down the
system. It could block the system if the console output is halted
or paused. * Made capture the CLI "core show channel" output into
a buffer to be output after the channel is unlocked. * Removed
use of C++ keyword as a variable name. out renamed to obuf. *
Checked allocation of obuf for failure so will not crash. (closes
issue ASTERISK-18571) Reported by: Pavel Troller Tested by:
rmudgett ........ Merged revisions 344661 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344662 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-11 15:47 +0000 [r344610] Jonathan Rose <jrose@digium.com>
* main/pbx.c, /: Fix a segmentation fault when using an extension
with CID matching and no CID. Attempting to call an extension
which used Caller ID matching with a channel that has an empty
caller id string would result in a segmentation fault. (closes
issue ASTERISK-18392 Reported By: Ales Zelenik ........ Merged
revisions 344608 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344609 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-10 23:21 +0000 [r344538-344560] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_macro.c: Fix app_macro.c MODULEINFO section
termination. (closes issue ASTERISK-18848) Reported by: Tony
Mountifield ........ Merged revisions 344557 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_queue.c: Fix potential deadlock calling ast_call()
with channel locks held. Fixed app_queue.c:ring_entry() calling
ast_call() with the channel locks held. Chan_local attempts to do
deadlock avoidance in its ast_call() callback and could deadlock
if a channel lock is already held. ........ Merged revisions
344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 344540 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_queue.c: Make AMI event AgentCalled get
CallerID/ConnectedLine info from the incoming channel. It was
strange that the AgentCalled AMI event would get most of its
information from the incoming channel but then get the CallerID
information from the outgoing channel. Before connected line
support was added, this information was always the same at this
point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham
Tested by: rmudgett ........ Merged revisions 344536 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344537 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-10 21:56 +0000 [r344494] David Vossel <dvossel@digium.com>
* /, main/bridging.c: Merged revisions 344493 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r344493 | dvossel | 2011-11-10 15:54:42 -0600 (Thu, 10 Nov 2011)
| 12 lines Fixes issue with ConfBridge participants hanging up
during DTMF feature menu usage getting stuck in conference
forever. When a conference user enters the DTMF menu they are
suspended from the bridge while the channel is handed off to the
DTMF feature code. If a user entered this state and hungup, there
existed a race condition where the channel could not exit the
conference because it was waiting on a signal that would never
arrive. This patch fixes that, because it would stupid for me to
talk about the problem and commit a patch for something else.
(closes issue ASTERISK-18829) Reported by: zvision ........
2011-11-10 21:15 +0000 [r344387-344441] Kinsey Moore <kmoore@digium.com>
* /, apps/app_meetme.c: Fix another incorrect case with meetme's
PIN logic and add documentation This fixes an issue where a user
of a dynamic conference was asked for a PIN twice. This also adds
documentation to assist in future modifications to the piece of
code responsible for PIN checking. (closes issue AST-670)
........ Merged revisions 344439 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344440 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/chan_sip.c, channels/sip/include/sip.h: Fix several
bugs with SDP parsing and well-formedness of responses Fix bug
ASTERISK-16558 which dealt with the order of responses to
incoming streams defined by SDP. Fix unreported bug where
offering multiple same-type streams would cause Asterisk to reply
with an incorrect SDP response missing one or more streams
without a proper declination. Fix bugs related to a single
non-audio stream being offered with responses requesting codecs
that were not offered in the initial invite along with an
additional audio stream that was not in the initial invite.
Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged
revisions 344385 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344386 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-10 16:29 +0000 [r344335] Matthew Nicholson <mnicholson@digium.com>
* res/res_rtp_asterisk.c, /: only attempt to do stun handling on
ipv4 or ipv4 mapped to ipv6 addresses Patch by: jkonieczny
(modified) ASTERISK-18490 ........ Merged revisions 344330 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344334 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-09 20:55 +0000 [r344272] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c: Fix deadlock during dialplan reload.
Another deadlock between the conlock/hints and channels/channel
locking orders. * Don't hold the channel and private lock in
sip_new() when calling ast_exists_extension(). (closes issue
ASTERISK-18740) Reported by: Byron Clark Patches:
sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by
Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch
uploaded by Byron Clark Tested by: Byron Clark ........ Merged
revisions 344268 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344271 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-09 20:10 +0000 [r344214-344217] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c, channels/sip/reqresp_parser.c,
channels/sip/include/sip.h,
channels/sip/include/reqresp_parser.h: Don't treat a host:port
string as a domain The domain matching code prior to 1.8 used to
manually remove the port from the host:port string when
determining if an incoming request matched the list of domains.
When switching to the new parsing functions, the documentation
implied that the "domain" was being returned by these functions,
when instead it was returning the "hostport" as defined by RFC
3261. This led to confusion and resulted in 1.8+ rejecting an
incoming request from x.x.x.x:xxxxx when domain=x.x.x.x was set
in sip.conf. This patch renames the "domain" variables in the
parsing functions to "hostport" to more accurately describe what
it is that they are returning and also properly truncates the
resulting hostport strings when dealing with domain matching.
Review: https://reviewboard.asterisk.org/r/1574/ ........ Merged
revisions 344215 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344216 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, tests/test_netsock2.c: Add a unit test for
ast_sockaddr_split_hostport Review:
https://reviewboard.asterisk.org/r/1575/ ........ Merged
revisions 344157 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344175 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-09 19:08 +0000 [r344161] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooh323.c, /, addons/ooh323c/src/ooh245.c,
addons/ooh323c/src/ooq931.h, addons/ooh323c/src/ootypes.h,
addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c:
Generate response to Status Enquiry message with Status q.931
message. Some PBXes require this for call status checking (closes
issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches:
ASTERISK-18748-5.patch (License #5415) patch uploaded by may213
Tested by: Fabrizio Lazzaretti ........ Merged revisions 344158
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 344159 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-09 17:15 +0000 [r344104] Kinsey Moore <kmoore@digium.com>
* /, apps/app_meetme.c: Fix pin parameter behavior regression in
MeetMe The last time this code was touched (by me), a subtlety
was missed based on the difference between needing to check a
pin's validity and the need to prompt for a pin. (closes issue
ASTERISK-18488) ........ Merged revisions 344102 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344103 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-09 15:28 +0000 [r344050] Matthew Nicholson <mnicholson@digium.com>
* /, formats/format_wav.c: don't call ltohl() twice on the same
value ASTERISK-18739 Patch by: pawel (modified) ........ Merged
revisions 344048 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 344049 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-08 22:14 +0000 [r344005] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c: Residual changes for Asterisk v10 branch
from ASTERISK-18747. Residual changes for Asterisk v10 branch
from ASTERISK-18747 after
https://reviewboard.asterisk.org/r/1564/ commit and associated
dialogs callid hash key change fix. * Make check_rtp_timeout()
return CMP_MATCH if need to delete dialog from dialogs_rtpcheck.
This is an optimization to avoid an unneeded lock/unlock and
object search when using ao2_unlink. * Prevent crash in
check_rtp_timeout() if dialog->rtp is NULL. Review:
https://reviewboard.asterisk.org/r/1557/ ........ Merged
revisions 344004 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-08 19:29 +0000 [r343951] Walter Doekes <walter+asterisk@wjd.nu>
* /, pbx/pbx_config.c: Fix crash when dialplan remove include is
called with too few arguments. "dialplan remove include x from y"
crashed when the amount of arguments was less than 6. (closes
issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by:
Andrey Solovyev ........ Merged revisions 343936 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 343944 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-08 18:35 +0000 [r343905] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 343900 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011)
| 11 lines Fixes regression caused by r343635 There was a missing
unlock for a function return that is only present in Asterisk 10
and Asterisk Trunk. (closes issue ASTERISK-18839) Reported by:
Michael L. Young Patches:
asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch
uploaded by Michael L. Young ........
2011-11-08 18:02 +0000 [r343853] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c, main/acl.c: Fixed reference to incorrect
variable if unknown host configured crash. * Fixed a LOG_ERROR
message referencing the config variable list v that had
previously been processed and became NULL. * Added error return
value set that was missing in an ast_append_ha() error return
path. (closes issue ASTERISK-18743) Reported by: Michele Patches:
issueA18743-fix_dynamic_exclude_static_bad_host_log.patch
(license #5674) patch uploaded by Walter Doekes Tested by:
Michele ........ Merged revisions 343851 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 343852 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-08 13:23 +0000 [r343790] Leif Madsen <leif@leifmadsen.com>
* /, build_tools/prep_tarball: Fix boo-boo in prep_tarball script.
A hardcoded a branch number was in the prep_tarball which could
not work. Changed it to the variable. ........ Merged revisions
343789 from http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-07 22:37 +0000 [r343744] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Make "sip show settings" CLI command get
RPID flags from the right global page The "Trust RPID" and "Send
RPID" entries in the "sip show settings" CLI command pulled the
flags from the incorrect global flags page. These are now read
from sip global flags page 0. (closes issue AST-711) ........
Merged revisions 343743 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-07 21:58 +0000 [r343693] Leif Madsen <leif@leifmadsen.com>
* configs/dundi.conf.sample, pbx/pbx_dundi.c, CHANGES: Allow built
in variables to be used with dynamic weights. You can now use the
built in variables , , and within a dynamic weight. For example,
this could be useful when you want to pass requested lookup
number to the SHELL() function which could be used to execute a
script to dynamically set the weight of the result. (Closes issue
ASTERISK-13657) Reported by: Joel Vandal Tested by: Leif Madsen,
Russell Bryant Patches: asterisk-1.6-dundi-varhead.patch uploaded
by Joel Vandal (License #5374)
2011-11-07 21:44 +0000 [r343692] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: respect case changes in peer names on sip
reload ASTERISK-18669 ........ Merged revisions 343690 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 343691 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-07 21:29 +0000 [r343684] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly
changing dialogs hash key callid. Changing an object value used
as a container key requires removing the object from the
container and reinserting it. * Created change_callid_pvt() to
call instead of build_callid_pvt(). The change_callid_pvt() will
correctly change the dialog callid so the ao2 conainter can
explicitly unlink it. ........ Merged revisions 343637 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 343677 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-07 20:35 +0000 [r343636] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Prevent BLF subscriptions from causing
deadlocks Fix a locking inversion in sip_send_mwi_to_peer that
was causing deadlocks. This function now requires that both the
peer and associated pvt be unlocked before it is called for cases
where peer and peer->mwipvt form a circular reference. (closes
issue ASTERISK-18663) Review:
https://reviewboard.asterisk.org/r/1563/ ........ Merged
revisions 343621 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 343635 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-07 19:58 +0000 [r343581] Walter Doekes <walter+asterisk@wjd.nu>
* main/udptl.c, /, UPGRADE.txt: Correct the default udptl port
range. The udptl port range was defined as 4000-4999 in the
udptl.conf.sample, as 4500-4599 if you didn't have a config and
4500-4999 if your config was broken. Default is now 4000-4999.
(closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher
Review: https://reviewboard.asterisk.org/r/1565 ........ Merged
revisions 343580 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-07 19:54 +0000 [r343579] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c: Fix deadlock if peer is destroyed while
sending MWI notice. A dialog cannot be destroyed by the
ao2_callback dialog_needdestroy because of a deadlock between the
dialogs container lock and the RWLOCK of the events subscription
list. * Create dialogs_to_destroy container to hold dialogs that
will be destroyed. * Ensure that the event subscription callback
will never happen with an invalid peer pointer by making the
event callback removal the first thing in the peer destructor
callback. NOTE: This particular deadlock will not happen with
Asterisk 10, but some of the changes still apply. (closes issue
ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review:
https://reviewboard.asterisk.org/r/1564/ ........ Merged
revisions 343577 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 343578 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-07 18:42 +0000 [r343534] Matthew Nicholson <mnicholson@digium.com>
* main/format.c, /: list all of the codecs associated with a
particular format id for CLI command "core show codec" AST-699
........ Merged revisions 343533 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-06 09:51 +0000 [r343492] Olle Johansson <oej@edvina.net>
* main/tcptls.c, include/asterisk/tcptls.h: Formatting and doxygen
improvements
2011-11-04 19:50 +0000 [r343448] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooTimer.c,
addons/ooh323c/src/dlist.c, /, addons/ooh323c/src/dlist.h,
addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c:
Final fix memleaks in GkClient codes, same for Timer codes.
(these memleaks stop development of gk codes, now i can continue)
Fix printHandler 'Unbalanced Structure' issues with locking
printHandler data for single thread. ........ Merged revisions
343281 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 343445 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-03 20:37 +0000 [r343394] Walter Doekes <walter+asterisk@wjd.nu>
* /, res/res_config_sqlite.c: Fix sqlite config driver segfault and
broken queries The sqlite realtime handler assumed you had a
static config configured as well. The realtime multientry handler
assumed that you weren't using dynamic realtime. (closes issue
ASTERISK-18354) (closes issue ASTERISK-18355) Review:
https://reviewboard.asterisk.org/r/1561 ........ Merged revisions
343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 343393 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-03 19:57 +0000 [r343338] Richard Mudgett <rmudgett@digium.com>
* /, funcs/func_dialgroup.c: Remove invalid flag given to iterator
in func_dialgroup.c ........ Merged revisions 343336 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 343337 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-03 15:40 +0000 [r343222-343278] Terry Wilson <twilson@digium.com>
* /, channels/sip/include/sip.h: Make room for the fax detect flags
The original REGISTERTRYING flag, in addition to being impossible
to check, also encroached on the space for the flag above it.
This patch moves the flags that were below REGISTERTRYING back to
where they were as though we had just removed the REGISTERTRYING
option. ........ Merged revisions 343276 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 343277 from
http://svn.asterisk.org/svn/asterisk/branches/10
* contrib/realtime/mysql/sippeers.sql, /, channels/chan_sip.c,
channels/sip/include/sip.h: Remove registertrying option in
chan_sip This option is not only useless, but has been broken
since inception since the flag was never copied from the peer
where it is set to the pvt where it was checked. RFC 3261
specificially states that you should not send a provisional
response to a non-INVITE request, and if we did fix the code so
that it worked, it would cause the same kind of user enumeration
vulnerability that we've discussed with the nat= setting. This
patch removes registertrying option and any code that would have
sent a 100 response to a register. Review:
https://reviewboard.asterisk.org/r/1562/ ........ Merged
revisions 343220 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 343221 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-02 22:46 +0000 [r343163-343219] Walter Doekes <walter+asterisk@wjd.nu>
* /, channels/chan_sip.c: Fix improper warning introduced by
r342927 and more tweaks Changeset r342927 introduced a warning
which was only supposed to be emitted when a found realtime peer
had an empty (or no) name. It turned out that there were some
inconsistencies left. Now found peers with an empty name are
explicitly ignored like before r342927 but better. Reviewed by:
Stefan Schmidts, Terry Wilson Review:
https://reviewboard.asterisk.org/r/1560 ........ Merged revisions
343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 343192 from
http://svn.asterisk.org/svn/asterisk/branches/10
* include/asterisk/utils.h, /, main/utils.c,
include/asterisk/stringfields.h: Ensure that string field lengths
are properly aligned Integers should always be aligned. For some
platforms (ARM, SPARC) this is more important than for others.
This changeset ensures that the string field string lengths are
aligned on *all* platforms, not just on the SPARC for which there
was a workaround. It also fixes that the length integer can be
resized to 32 bits without problems if needed. (closes issue
ASTERISK-17310) Reported by: radael, S Adrian Reviewed by:
Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review:
https://reviewboard.asterisk.org/r/1549 ........ Merged revisions
343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 343158 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-02 19:33 +0000 [r343049-343104] Leif Madsen <leif@leifmadsen.com>
* apps/app_authenticate.c: Add note about how Authenticate()
application with option 'd' works. (closes issue ASTERISK-17422)
Reported by: Leif Madsen ........ Merged revisions 343102 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 343103 from
http://svn.asterisk.org/svn/asterisk/branches/10
* configs/queues.conf.sample: Update documentation for leastrecent
strategy. In queues.conf.sample the leastrecent strategy was
incorrectly described. Now updated to reflect how the strategy
actually checks peers. (closes issue ASTERISK-17854) Reported by:
Sebastian Denz Patches: queues.conf-doc_issue.patch (License
#6139) ........ Merged revisions 343047 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 343048 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-02 13:46 +0000 [r342992] Kevin P. Fleming <kpfleming@digium.com>
* /, apps/app_meetme.c: Modify comments in MeetMe application
documentation about DAHDI. The MeetMe application documentation
has some comments about usage of DAHDI, and they were a bit
outdated relative to modern DAHDI releases. This patch changes
the comment to just tell the user that a functional DAHDI timing
source is required, and no longer mention 'dahdi_dummy', since
that module does not exist in current DAHDI releases. ........
Merged revisions 342990 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342991 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-11-01 21:02 +0000 [r342871-342930] Walter Doekes <walter+asterisk@wjd.nu>
* /, channels/chan_sip.c, configs/extconfig.conf.sample,
include/asterisk/config.h, main/config.c: Several fixes to the
chan_sip dynamic realtime peer/user lookup There were several
problems with the dynamic realtime peer/user lookup code. The
lookup logic had become rather hard to read due to lots of
incremental changes to the realtime_peer function. And, during
the addition of the sipregs functionality, several possibilities
for memory leaks had been introduced. The insecure=port matching
has always been broken for anyone using the sipregs family. And,
related, the broken implementation forced those using sipregs to
*still* have an ipaddr column on their sippeers table. Thanks
Terry Wilson for comprehensive testing and finding and fixing
unexpected behaviour from the multientry realtime call which
caused the realtime_peer to have a completely unused code path.
This changeset fixes the leaks, the lookup inconsistenties and
that you won't need an ipaddr column on your sippeers table
anymore (when you're using sipregs). Beware that when you're
using sipregs, peers with insecure=port will now start matching!
(closes issue ASTERISK-17792) (closes issue ASTERISK-18356)
Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry
Wilson Review: https://reviewboard.asterisk.org/r/1395 ........
Merged revisions 342927 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342929 from
http://svn.asterisk.org/svn/asterisk/branches/10
* contrib/realtime/mysql/sippeers.sql (added),
configs/res_config_mysql.conf.sample, /,
configs/extconfig.conf.sample, configs/res_ldap.conf.sample,
res/res_realtime.c, UPGRADE-1.8.txt, configs/dbsep.conf.sample,
main/config.c, contrib/realtime/mysql/sipfriends.sql (removed):
Cleanup references to sipusers and sipfriends dynamic realtime
families Somewhere between 1.4 and 1.8 the sipusers family has
become completely unused. Before that, the sipfriends family had
been obsoleted in favor of separate sipusers and sippeers
families. Apparently, they have been merged back again into a
single family which is now called "sippeers". Reviewed by:
irroot, oej, pabelanger Review:
https://reviewboard.asterisk.org/r/1523 ........ Merged revisions
342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........ Merged revisions 342870 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-31 17:51 +0000 [r342825] Richard Mudgett <rmudgett@digium.com>
* main/format.c, /, main/format_cap.c: Misc format capability
fixes. * Fixed typo in format_cap.c:joint_copy_helper() using the
wrong variable. * Fix potential race between checking if an
interface exists and adding it to the container in
format.c:ast_format_attr_reg_interface(). * Fixed double rwlock
destroy in format.c:ast_format_attr_init() error exit path. *
Simplified format.c:find_interface() and
format.c:has_interface(). ........ Merged revisions 342824 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-31 16:10 +0000 [r342771] Matthew Jordan <mjordan@digium.com>
* main/pbx.c, /, channels/chan_iax2.c: Fixed invalid memory access
when adding extension to pattern match tree When an extension is
removed from a context, its entry in the pattern match tree is
not deleted. Instead, the extension is marked as deleted. When an
extension is removed and re-added, if that extension is also a
prefix of another extension, several log messages would report an
error and did not check whether or not the extension was deleted
before accessing the memory. Additionally, if the extension was
already in the tree but previously deleted, and the pattern was
at the end of a match, the findonly flag was not honored and the
extension would be erroneously undeleted. Additionaly, it was
discovered that an IAX2 peer could be unregistered via the CLI,
while at the same time it could be scheduled for unregistration
by Asterisk. The unregistration method now checks to see if the
peer was already unregistered before continuing with an
unregistration. (closes issue ASTERISK-18135) Reported by: Jaco
Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1526 ........ Merged
revisions 342769 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342770 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-30 02:31 +0000 [r342716] Terry Wilson <twilson@digium.com>
* /, res/res_calendar.c: Don't crash on empty notify channel
........ Merged revisions 342715 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-29 04:41 +0000 [r342663-342664] Richard Mudgett <rmudgett@digium.com>
* include/asterisk/linkedlists.h: Whitespace and some better macro
variable names. * Renamed AST_LIST_TRAVERSE_SAFE_BEGIN __new_prev
to __list_current. * Renamed AST_LIST_MOVE_CURRENT __list_cur to
__extracted.
* /, include/asterisk/linkedlists.h, tests/test_linkedlists.c: Fix
AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable.
AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an
iteration or before AST_LIST_REMOVE_CURRENT() without corrupting
the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the
list if AST_LIST_INSERT_BEFORE_CURRENT() or
AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed
cut and paste error using the wrong variable in
AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests
for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and
AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 342662 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-27 20:11 +0000 [r342606] Matthew Nicholson <mnicholson@digium.com>
* /, main/dsp.c: tweak the v21 detector to detect an additional
pattern of hits and misses ........ Merged revisions 342605 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-27 19:48 +0000 [r342557-342604] Jonathan Rose <jrose@digium.com>
* res/res_rtp_multicast.c, /: Fix sequence number overflow over 16
bits causing codec change in RTP packets. Sequence number was
handled as an unsigned integer (usually 32 bits I think, more
depending on the architecture) and was put into the rtp packet
which is basically just a bunch of bits using an or operation.
Sequence number only has 16 bits allocated to it in an RTP packet
anyway, so it would add to the next field which just happened to
be the codec. This makes sure the sequence number is set to be a
16 bit integer regardless of architecture (hopefully) and also
makes it so the incrementing of the sequence number does bitwise
or at the peak of a 16 bit number so that the value will be set
back to 0 when going beyond 65535 anyway. (closes issue
ASTERISK-18291) Reported by: Will Schick Review:
https://reviewboard.asterisk.org/r/1542/ ........ Merged
revisions 342602 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342603 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, res/res_jabber.c: Cleanup reference leaks in res_jabber
res_jabber.c had a number of places where astobjs would be
referenced and have their reference counts bumped without having
a dereference made before the object lost scope. This patch adds
a number of ASTOBJ_UNREFs to resolve that. Review:
https://reviewboard.asterisk.org/r/1478/ ........ Merged
revisions 342545 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342546 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-25 22:06 +0000 [r342486-342489] Richard Mudgett <rmudgett@digium.com>
* /, main/astobj2.c: Check fopen return value for ao2 reference
debug output. Reported by: wdoekes Patched by: wdoekes Review:
https://reviewboard.asterisk.org/r/1539/ ........ Merged
revisions 342487 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342488 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, channels/sig_pri.c: Change D-channel warning to be less
confusing on non-NFAS setups. The "No D-channels available! Using
Primary channel as D-channel anyway!" WARNING message has been
confusing on non-NFAS setups. The message refers to things that
are NFAS specific. * Changed the warning to several different
warnings to be more accurate for the situation and less confusing
as a result: "No D-channels up! Switching selected D-channel from
X to Y.", "No D-channels up!", and "D-channel is down!". ........
Merged revisions 342484 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342485 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-25 21:11 +0000 [r342382-342437] Terry Wilson <twilson@digium.com>
* /, apps/app_queue.c: Use int for storing ao2_container_count
instad of size_t AST-676 ........ Merged revisions 342435 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342436 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_queue.c: Simplify queue membercount code Despite an
ominous sounding comment stating that membercount was for "logged
in" members only and thus we couldn't use ao2_container_count(),
I could not find a single place in the code where that seemed to
be accurate. The only time we decremented membercount was when we
were marking something dead or actually removing it. The only
places we incremented it were either after ao2_link(), or trying
to correct for having set it to 0 during a reload. In every case
where we were correcting the value, it seemed that we were trying
to make the count actually match what ao2_container_count() would
return. The only place I could find where we made a determination
about something being "logged in" or not, we didn't trust the
membercount, but instead looked at devicestate, paused, etc. This
patch removes membercount, replaces its use with
ao2_container_count, and manually adds the results of
ao2_container_count to a "membercount" field for ast_data queue
query results. This patch also would fix AST-676, but as it is
slightly riskier than the previously committed fix, the two
commits have been made separately. Reivew:
https://reviewboard.asterisk.org/r/1541/ ........ Merged
revisions 342383 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342384 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_queue.c: Properly update membercount for reloaded
members Since q->membercount is set to 0 before reloading, it is
important to increment it again for reloaded members as well as
added. (closes issue AST-676) Review:
https://reviewboard.asterisk.org/r/1541/ ........ Merged
revisions 342380 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342381 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-25 19:09 +0000 [r342278-342330] Kinsey Moore <kmoore@digium.com>
* pbx/pbx_spool.c, /: Fix compilation on Snow Leopard/FreeBSD for
pbx_spool.c One of the changes in the recent spool handling of
hardlinks patch was just outside a HAVE_INOTIFY block and caused
compilation to fail in some build environments. This has been
corrected. ........ Merged revisions 342328 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342329 from
http://svn.asterisk.org/svn/asterisk/branches/10
* pbx/pbx_spool.c, /: Merged revisions 342277 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r342277 | kmoore | 2011-10-25 11:08:04 -0500
(Tue, 25 Oct 2011) | 25 lines Merged revisions 342276 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) |
18 lines Fix spool handling to allow call files to be hardlinked
into place This fixes the inotify code to handle call files being
hardlinked into the spool directory. The smsq utility does this,
instead of rename(), to ensure that it cannot accidentally
overwrite an existing spool file. A rename() might do that, but
link() will definitely not. The inotify code had broken this,
because it would wait for an IN_CLOSE_WRITE event on the file...
which was never forthcoming, since it was never opened. Now we
look for IN_OPEN events following the IN_CREATE event, and only
wait for an IN_CLOSE_WRITE if the file was actually opened.
Patch-by: dwmw2 (closes issue ASTERISK-18331) Review:
https://reviewboard.asterisk.org/r/1391/ ........
................
2011-10-25 01:29 +0000 [r342225] Terry Wilson <twilson@digium.com>
* /, include/asterisk/config.h, main/config.c: Return NULL when no
results returned for realtime_multientry It was not documented
what the return value should be when no entries were returned
with the multientry realtime callback. This change forces
consistent behavior even if the backends return an empty
ast_config. Review: https://reviewboard.asterisk.org/r/1521/
........ Merged revisions 342223 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 342224 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-24 22:37 +0000 [r342184] Richard Mudgett <rmudgett@digium.com>
* /, include/asterisk/astobj2.h: Fix ao2obj.h comment typos and add
missing link/unlink nolock debug defines. ........ Merged
revisions 342183 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-24 22:09 +0000 [r342148] Jonathan Rose <jrose@digium.com>
* main/features.c: Fixes a segfault caused by referencing null
frames introduced in r338623
2011-10-24 21:01 +0000 [r342112] Richard Mudgett <rmudgett@digium.com>
* apps/app_queue.c: Fix use of OBJ_KEY in Queue application. To use
the new OBJ_KEY flag, the container hash and compare callback
functions must be updated to support OBJ_KEY. Otherwise, bad
things happen. (issue ASTERISK-14769)
2011-10-24 20:01 +0000 [r342063] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: Outbound SIP OPTIONS messages will now
include fromuser of related peer. This behavior matches up more
closely with the way invite/register/etc are handled. This patch
also modifies some adjacent code for code style compliance.
Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy
Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded
by Jeremy Kister (license #6232) ........ Merged revisions 342061
from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........
Merged revisions 342062 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-24 07:40 +0000 [r341923-342018] Gregory Nietsky <gregory@distrotech.co.za>
* /, apps/app_queue.c: queues container needs locking when using
the OBJ_NOLOCK flag ........ Merged revisions 342017 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_queue.c: Remove some ref leaks and a return without
unlock. There some resource leaks introduced in asterisk 10 make
sure that locks are not held on return and we release ref's held.
........ Merged revisions 341972 from
http://svn.asterisk.org/svn/asterisk/branches/10
* apps/app_queue.c: Whitespace Fixups / Add Braces This janitorial
patch is related to work on RB1538
2011-10-22 12:03 +0000 [r341869] Alexandr Anikin <may@telecom-service.ru>
* addons/chan_ooh323.c, /: Merged revisions 341313 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r341313 | may | 2011-10-19 03:33:49 +0400 (Wed,
19 Oct 2011) | 10 lines Merged revisions 341312 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct 2011) | 3
lines fix issue on channel numbering (calls could have same
channel number on heavy loaded system) ........ ................
2011-10-21 16:42 +0000 [r341808-341811] Matthew Nicholson <mnicholson@digium.com>
* /, pbx/pbx_lua.c: only process args that exist ASTERISK-18395
........ Merged revisions 341809 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341810 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, pbx/pbx_lua.c: don't limit the length of app and function
arguments ASTERISK-18395 ........ Merged revisions 341806 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341807 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-21 09:16 +0000 [r341769] Gregory Nietsky <gregory@distrotech.co.za>
* res/res_fax.c: White space fixes in res_fax
2011-10-20 22:03 +0000 [r341719] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c, res/res_agi.c, include/asterisk/features.h:
Fix AGI exec Park to honor the Park application parameters. The
fix for ASTERISK-12715 and ASTERISK-12685 added a check for the
Park application because the channel needed to be masqueraded to
prevent a crash. Since the Park application now always
masquerades the channel into the parking lot, the special check
is no longer needed. The fix also resulted in AGI exec Park
attempting to double park the call and not honor the Park
application parameters. * Removed no longer necessary call to
ast_masq_park_call() by AGI exec for the Park application.
(Reverts -r146923) * Fix Park application to only return 0 or -1.
The AGI exec Park was causing broken pipe error messages because
the Park application returned 1 on successful park. (closes issue
ASTERISK-18737) ........ Merged revisions 341717 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341718 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-20 21:28 +0000 [r341666-341713] Paul Belanger <paul.belanger@polybeacon.com>
* /, funcs/func_callerid.c: Fixed typo from previous commit
........ Merged revisions 341704 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341707 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, funcs/func_callerid.c: Updated documentation for the optional
CID parameter with CALLERID ........ Merged revisions 341664 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341665 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-20 18:27 +0000 [r341583-341624] Gregory Nietsky <gregory@distrotech.co.za>
* /, configs/queues.conf.sample: Merged revisions 341599 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
........ r341599 | irroot | 2011-10-20 20:20:08 +0200 (Thu, 20
Oct 2011) | 8 lines add documentation for check_state_unknown in
configs/queues.conf.sample app_queue allows calls to members in a
"Unknown" state to be treated as available setting
check_state_unknown = yes will cause app_queue to query the
channel driver to better determine the state this only applies to
queues with ringinuse or ignorebusy set appropriately. ........
* /, CHANGES, apps/app_queue.c: Merged revisions 341580 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
........ r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20
Oct 2011) | 15 lines Add option to check state when state is
unknown r341486 reverts r325483 this is a rework of the patch.
optimize to minimize load. add option check_state_unknown to
control whether a member with unknown device state is checked
there is a small % chance that calls will be sent to the member
when they on a call. app_queue will see a device with unknown
state as available and does not try verify the state without this
option enabled. Review: https://reviewboard.asterisk.org/r/1535/
........
2011-10-20 15:17 +0000 [r341533] Terry Wilson <twilson@digium.com>
* /, include/asterisk/strings.h: Clean up ast_check_digits The code
was originally copied from the is_int() function in the AEL code.
wdoekes pointed out that the function should take a const char*
and that their was an unneeded variable. This is now fixed.
........ Merged revisions 341529 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341530 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-19 21:24 +0000 [r341487] Matthew Nicholson <mnicholson@digium.com>
* /, apps/app_queue.c: Merged revisions 341486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r341486 | mnicholson | 2011-10-19 16:23:17 -0500 (Wed, 19 Oct
2011) | 18 lines Fix a performance regression introduced in
r325483. The regression was caused by a call to
ast_parse_device_state() in app_queue's ring_entry() function.
The ast_parse_device_state() function eventually calls
ast_channel_get_full() with a channel name prefix which causes it
to walk the channel list causing massive lock contention and slow
downs. This patch fixes the regression by removing the call to
ast_parase_device_state() which should be unnecessary. Queue
member device state should be maintained by device state events.
Some users have seen instances where busy agents were called when
they shouldn't have, which is the reason the call to
ast_parse_device_state() was added. That change appears to have
resolved that issue but also causes this performance regression.
There may still be issues with queue member status, and if so,
alternative methods should be investigated to resolve them.
AST-695 ........
2011-10-19 19:02 +0000 [r341437] Paul Belanger <paul.belanger@polybeacon.com>
* /, channels/chan_gtalk.c: Outgoing calls with Google Voice Google
has recently make some changes (again) to their protocol. Rather
then patching asterisk to flip between the two different methods,
we now allow both. Lets hope this keeps Google Voice happy for a
while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov
Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses
6311) ........ Merged revisions 341435 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341436 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-19 07:45 +0000 [r341381] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c, include/asterisk/strings.h: Don't use
is_int() since it doesn't link well on all platforms Just create
an normal API function in strings.h that does the same thing just
to be safe. ASTERISK-17146 ........ Merged revisions 341379 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341380 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-19 07:27 +0000 [r341378] Stefan Schmidt <sst@sil.at>
* /, channels/chan_sip.c: Don't sent in-dialog requests like UPDATE
when Asterisk has not yet received a Contact URI from a UAS
........ Merged revisions 341366 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341377 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-18 23:45 +0000 [r341316] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Don't resolve numeric hosts or contact
unresolved hosts If a SIP dial string contains a numeric hostname
that is not a peer name, don't try to resolve it as it is
unlikely that someone really means Dial(SIP/0.0.4.26) when
Dial(SIP/1050) is called. Also, make sure that create_addr
returns -1 if an address isn't resolved so that we don't attempt
to send SIP requests to an address that doesn't resolve. (closes
issue ASTERISK-17146, ASTERISK-17716) Review:
https://reviewboard.asterisk.org/r/1532/ ........ Merged
revisions 341314 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341315 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-18 21:15 +0000 [r341256] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_analog.c, /,
channels/chan_sip.c, main/features.c, channels/chan_iax2.c,
channels/sip/include/sip.h, channels/chan_mgcp.c,
include/asterisk/features.h: More parking issues. * Fix potential
deadlocks in SIP and IAX blind transfer to parking. * Fix SIP,
IAX, DAHDI analog, and MGCP channel drivers to respect the
parkext_exclusive option with transfers (Park(,,,,,exclusive_lot)
parameter). Created ast_park_call_exten() and
ast_masq_park_call_exten() to maintian API compatibility. * Made
masq_park_call() handle a failed ast_channel_masquerade() setup.
* Reduced excessive struct parkeduser.peername[] size. ........
Merged revisions 341254 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341255 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-17 17:58 +0000 [r341198] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* /, pbx/pbx_realtime.c: Remove an unused include of md5.h Unused
include of asterisk/md5.h in pbx_realtime.c . A commit needed to
test the commit message. Merged-From:
http://svn.asterisk.org/svn/asterisk/branches/1.8@341074
Merged-From:
http://svn.asterisk.org/svn/asterisk/branches/10@341148
2011-10-17 17:38 +0000 [r341191] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Initialize variables before calling
parse_uri If parse_uri was called with an empty URI, some
pointers would be modified and an invalid read could result. This
patch avoids calling parse_uri with an empty contact uri when
parsing REGISTER requests. AST-2011-012 (closes issue
ASTERISK-18668) ........ Merged revisions 341189 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341190 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-17 16:39 +0000 [r341126-341147] Paul Belanger <paul.belanger@polybeacon.com>
* /, tests/test_format_api.c: Set 'core' support level for
test_format_api.c ........ Merged revisions 341146 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_voicemail.c: Multiple revisions 341108,341112
........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon,
17 Oct 2011) | 2 lines Voicemail compiler flags are 'core'
support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400
(Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged
revisions 341108,341112 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341122 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-17 16:18 +0000 [r341096] Jason Parker <jparker@digium.com>
* /, CHANGES: Add information about limitations of new codec
support in channel drivers. (issue ASTERISK-18680) ........
Merged revisions 341094 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-17 15:45 +0000 [r341090] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Don't try to remove peers without IPs
from peers_by_ip (closes issue ASTERISK-18696) ........ Merged
revisions 341088 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341089 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-14 21:37 +0000 [r341024] Kevin P. Fleming <kpfleming@digium.com>
* /, build_tools/embed_modules.xml, Makefile.moddir_rules: Change
the internal name of the menuselect options that are used to
control whether modules are embedded or not; using just the bare
category name led to accidentally enabling these options when
users used the wrong "--enable" operation on the menuselect
command line. Now the internal option names are prefixed with
"EMBED_", so they won't be the same as the name of the category
containing the modules they control the embedding of. ........
Merged revisions 341022 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 341023 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-14 21:15 +0000 [r340973] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Fix simple switch to not progress a call
when call already progressed. If a simple switch was started on a
device and then a specific call made (such as redial or speed
dial), on timeout of the simple switch the call would be
attempted again. This patch only allows the simple switch to make
a call if the substate is still in the collecting digits mode.
Also added small debug message to dialAndAactivate sub. Tested by
snuff and myself.
2011-10-14 20:51 +0000 [r340972] Kinsey Moore <kmoore@digium.com>
* res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions
340971 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r340971 | kmoore | 2011-10-14 15:50:37 -0500
(Fri, 14 Oct 2011) | 15 lines Merged revisions 340970 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) |
8 lines Quiet RTCP Receiver Reports during fax transmission RTCP
is now disabled for "inactive" RTP audio streams during SIP T.38
sessions. The ability to disable RTCP streams in res_rtp_asterisk
was missing, so this code was added to support the bug fix.
(closes issue ASTERISK-18400) ........ ................
2011-10-14 18:38 +0000 [r340932] Jonathan Rose <jrose@digium.com>
* utils/utils.xml, /, funcs/func_jitterbuffer.c: Some additional
module documentation changes for 10 for the menuselect change.
(issue ASTERISK-18268) ........ Merged revisions 340931 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-14 16:45 +0000 [r340880] Terry Wilson <twilson@digium.com>
* main/channel.c, /: Avoid unnecessary WARNING message Add
AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid
displaying a WARNING message. (closes issue ASTERISK-18610) Patch
by: Kristijan_Vrban ........ Merged revisions 340878 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 340879 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-13 23:08 +0000 [r340811-340813] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Fix DTMF blind transfer continuing to execute
dialplan after transfer. Party A calls Party B. Party A DTMF
blind transfers Party B to Party C. Party A channel continues to
execute dialplan. * Fixed the return value of
builtin_blindtransfer() to return the correct value after a
transfer so the dialplan will not keep executing. * Removed
unnecessary connected line update that did not really do
anything. * Made access to GOTO_ON_BLINDXFR thread safe in
check_goto_on_transfer(). * Fixed leak of xferchan for failure
cases in check_goto_on_transfer(). * Updated debug messages in
builtin_blindtransfer() and check_goto_on_transfer(). (closes
issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett
........ Merged revisions 340809 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 340810 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /: Update 10 merged property.
* /: Restore branch 10 merge properties.
2011-10-13 08:53 +0000 [r340771] Gregory Nietsky <gregory@distrotech.co.za>
* /: Merged revisions 339463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) |
9 lines Only change the capabilities on the gateway when the
session is been destroyed there is still a race condition that
ends in a segfault. if the caps are changed the logic in
res_fax_spandsp will run T30 code not gateway code to end the
session. this has been experienced on a "slower" under spec
system. ........
2011-10-13 07:05 +0000 [r340720] Stefan Schmidt <sst@sil.at>
* channels/chan_sip.c: Merged revisions 340718 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r340718 | schmidts | 2011-10-13 06:59:50 +0000
(Thu, 13 Oct 2011) | 9 lines Merged revisions 340717 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13
Oct 2011) | 3 lines storing the route-set also on a 181 response
not only on 180,182 or 183. ........ ................
2011-10-13 07:02 +0000 [r340665-340719] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Initialize ast_sockaddr before calling
ast_sockaddr_resolve Avoid possible jump based on unitialized
value ........ Merged revisions 340715 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 340716 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, res/res_config_sqlite.c: Don't skip the query field on a
realtime multi query There is no documented reason to not add the
query field to the varlist returned by a realtime multi query,
despite the config category being set to its value. Of course,
there is no documentation that the category should be set to the
value either. There is lots of no documentation when it comes to
realtime. But, other engines do not skip this field so I am
forcing this backend to follow the convention, because not doing
so is very silly. ........ Merged revisions 340662 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 340663 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-12 21:28 +0000 [r340626] Stefan Schmidt <sst@sil.at>
* channels/chan_sip.c: Merged revisions 340577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r340577 | schmidts | 2011-10-12 20:33:37 +0000
(Mit, 12 Okt 2011) | 9 lines Merged revisions 340576 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12
Okt 2011) | 3 lines Store route-set from provisional SIP
responses so early-dialog requests can be routed properly
........ ................
2011-10-12 21:02 +0000 [r340579] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 340578 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r340578 | twilson | 2011-10-12 13:57:19 -0700
(Wed, 12 Oct 2011) | 16 lines Merged revisions 340534 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011)
| 9 lines Update SIP realtime fullcontact regardless of caching
We should update the fullcontact field in the realtime table
whether or not rtcachefriends is set. There is no reason to treat
a non-cached realtime entity differently than a cached in this
regard. (closes issue ASTERISK-18446) Reported by: wdoekes
........ ................
2011-10-12 20:09 +0000 [r340472-340524] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Initialize the PRI channel alarms
properly on startup. The PRI channel alarms were initialized with
an inverted sense. (closes issue ASTERISK-18710) Reported by:
Tzafrir Cohen ........ Merged revisions 340522 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 340523 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_meetme.c: Update MeetMe p and X option documentation
when interacting with the s option. ASTERISK-12175 changed the p
and X options to not interfere with the s option when they are
used together. It makes more sense for the s option to have
priority for the DTMF '*' key since it cannot change its
activation code. Otherwise, you could not use option s with the p
or X options. JIRA AST-671 ........ Merged revisions 340470 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 340471 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-12 16:29 +0000 [r340420] Paul Belanger <paul.belanger@polybeacon.com>
* /, channels/chan_sip.c: Fix verbose messages when IPv6 logic was
added (closes issue ASTERISK-18612) Reported by: Tim Osman
........ Merged revisions 340418 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 340419 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-11 21:06 +0000 [r340318-340367] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c:
Add protection for SS7 channel allocation and better glare
handling. * Added a CLI "ss7 show channels" command that might
prove useful for future debugging. * Made the incoming SS7
channel event check and gripe message uniform. * Made sure that
the DNID string for an incoming call is always initialized.
(issue ASTERISK-17966) Reported by: Kenneth Van Velthoven
Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621)
patch uploaded by rmudgett ........ Merged revisions 340365 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 340366 from
http://svn.asterisk.org/svn/asterisk/branches/10
* channels/sip/include/dialog.h, /, channels/chan_sip.c: Fix some
potential deadlocks pointed out by helgrind. * Fixed deadlock
potential calling dialog_unlink_all() in __sip_autodestruct().
Found by helgrind. * Fixed deadlock potential in
handle_request_invite() after calling sip_new(). Found by
helgrind. * The sip_new() function now returns with the created
channel already locked. * Removed the dead code that starts a PBX
in in sip_new(). No sip_new() callers caused that code to be
executed and it was a bad thing to do anyway. * Removed unused
parameters and return value from dialog_unlink_all(). * Made
dialog_unlink_all() and __sip_autodestruct() safely obtain the
owner and private channel locks without a deadlock avoidance
loop. ........ Merged revisions 340284 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 340310 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-11 19:06 +0000 [r340283] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* main/channel.c, /, main/sha1.c, include/asterisk/sha1.h: Update
SHA1 code to RFC 6234 RFC 6234 is an update to RFC 3174 from
which the code was originally taken. It has a slightly better
code, and a better phrased license (simple 3-clause BSD). *
main/sha1.c is sha1.c from RFC 6234 with formatting changes only.
* include/asterisk/sha1.h merges sha.h and sha-private.h from RFC
6234. * Removed unused include of asterisk/sha1.h from
main/channels.c Review: https://reviewboard.asterisk.org/r/1503/
Merge-From:
http://svn.asterisk.org/svn/asterisk/branches/1.8@340263
Merge-From:
http://svn.asterisk.org/svn/asterisk/branches/10@340280
2011-10-11 18:57 +0000 [r340282] Richard Mudgett <rmudgett@digium.com>
* main/manager.c, /, include/asterisk/manager.h: Convert registered
AMI actions to ao2 objects. * Fixed race between calling an AMI
action callback and unregistering that action. Refixes
ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential
memory leak if an AMI action failed to get registered because is
already was registered. Part of the ao2 conversion. * Fixed AMI
ListCommands action not walking the actions list with a lock
held. * Fix usage of ast_strdupa() and alloca() in loops. Excess
stack usage. * Fix AMI Originate action Variable header requiring
a space after the header colon. Reported by Yaroslav Panych on
the asterisk-dev list. * Increased the number of listed variables
allowed per AMI Originate action Variable header to 64. * Fixed
AMI GetConfigJSON action output format. * Fixed usage of res
contents outside of scope in append_channel_vars(). * Fixed
inconsistency of config file channelvars option. The values no
longer accumulate with every channelvars option in the config
file. Only the last value is kept to be consistent with the CLI
"manager show settings" command. (closes issue ASTERISK-18479)
Reported by: Jaco Kroon ........ Merged revisions 340279 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 340281 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-10-10 23:10 +0000 [r340221-340224] Terry Wilson <twilson@digium.com>
* UPGRADE.txt, main/db.c: Return error when no rows are deleted for
AMI DBDelTree (closes issue AST-654)
* /, main/db.c: Merged revisions 340222 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r340222 | twilson | 2011-10-10 15:55:39 -0700 (Mon, 10 Oct 2011)
| 8 lines On astdb conversion, also warn about permissions
requirements The user running Asterisk must have permission to
the directory the Asterisk database resides in since SQLite 3
needs to be able to create a journal file. (closes issue
ASTERISK-18174) ........
* utils/Makefile, utils/utils.xml, /, UPGRADE.txt,
utils/astdb2bdb.c (added): Merged revisions 340219-340220 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
........ r340219 | twilson | 2011-10-10 15:38:06 -0700 (Mon, 10
Oct 2011) | 8 lines Add astdb conversion utility for Berkeley to
SQLite 3 If someone wants to backtrack from Asterisk 1.8 to 10
they can use the astdb2bdb utility to convert the database back
to the Berkeley format that Asterisk 1.8 uses. Review:
https://reviewboard.asterisk.org/r/1502/ ........ r340220 |
twilson | 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) | 2 lines
Add a missing file for the astdb2bdb conversion utility ........
2011-10-10 20:39 +0000 [r340166] Matthew Jordan <mjordan@digium.com>
* /, channels/chan_sip.c: Merged revisions 340165 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r340165 | mjordan | 2011-10-10 15:30:18 -0500
(Mon, 10 Oct 2011) | 20 lines Merged revisions 340164 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011)
| 13 lines Updated chan_sip to place calls on hold if SDP address
in INVITE is ANY This patch fixes the case where an INVITE is
received with c=0.0.0.0 or ::. In this case, the call should be
placed on hold. Previously, we checked for the address being
null; this patch keeps that behavior but also checks for the ANY
IP addresses. Review: https://reviewboard.asterisk.org/r/1504/
(closes issue ASTERISK-18086) Reported by: James Bottomley Tested
by: Matt Jordan ........ ................
2011-10-10 14:16 +0000 [r340110] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, main/manager.c, /, res/res_fax.c, apps/app_fax.c,
include/asterisk/module.h, res/res_agi.c,
include/asterisk/xmldoc.h, doc/appdocsxml.dtd, main/loader.c,
main/xmldoc.c: Merged revisions 340109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r340109 | mnicholson | 2011-10-10 09:15:41 -0500
(Mon, 10 Oct 2011) | 18 lines Merged revisions 340108 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct
2011) | 11 lines Load the proper XML documentation when multiple
modules document the same application. This patch adds an
optional "module" attribute to the XML documentation spec that
allows the documentation processor to match apps with identical
names from different modules to their documentation. This patch
also fixes a number of bugs with the documentation processor and
should make it a little more efficient. Support for multiple
languages has also been properly implemented. ASTERISK-18130
Review: https://reviewboard.asterisk.org/r/1485/ ........
................
2011-10-10 00:57 +0000 [r339993-340071] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Add skinny version 17 protocol support.
Added some data to skinny packet structures to make compatible
with v17. Added protocolversion to device, set on registration
based on the version provided by device. v17 includes some
increased ip space for ip6. This patch increases ip space in the
packets but still only uses ip4. Some packet structures
duplicated (ip4 and ip6 types). ip4 type used unless version is
greater or equal to 17. Tested by snuff and myself on 7961 with
recent 8.5 firmware. Also tested compatible with old 7960 and
older 30VIPs.
* channels/chan_skinny.c: Increase SKINNY_MAX_PACKET and add some
logging. Increase SKINNY_MAX_PACKET to 2000 bytes to handle some
messages in v17 that are greater than the old 1000 bytes. Also
add some useful logging regarding packet and session handling. A
device (with protocol v17) was sending a packet with length
greater than 1000 which resulted in the TCP session being
destroyed and registration being retryed.
* /, channels/chan_skinny.c: Merged revisions 340031 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r340031 | wedhorn | 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011)
| 8 lines Return -1 to skinny_session if register rejected. If
device registration is rejected, return -1 so that the session is
destroyed immediately. Previously, a segfault would occur on a
graceful shutdown if a register is rejected and the
skinny_session has not yet timed out. ........
* /, channels/chan_skinny.c: Merged revisions 339992 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r339992 | wedhorn | 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011)
| 9 lines Remove log message on traverse session list. On
destroying a session, a list of sessions is traversed to find the
matching session. For each session not matching, skinny
erroneously logged that the session was not matched. While
technically correct the message was misleading, and tended to
indicate errors that were not there. ........
2011-10-09 01:19 +0000 [r339832-339947] Igor Goncharovskiy <igor.goncharovsky@gmail.com>
* channels/chan_unistim.c, /: Merged revisions 339942 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r339942 | igorg | 2011-10-09 08:18:02 +0700
(Вск, 09 Окт 2011) | 12 lines Merged revisions 339938 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) |
6 lines Fix compilation issue, caused by missed session structure
(closes issue ASTERISK-18694) Reported by: alex70 ........
................
* channels/chan_unistim.c, /: Merged revisions 339885 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r339885 | igorg | 2011-10-08 22:46:27 +0700
(Сбт, 08 Окт 2011) | 13 lines Merged revisions 339884 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) |
7 lines Fix segfault in Unistim channel (closes issue
ASTERISK-18638) Reported by: jonnt ........ ................
* channels/chan_unistim.c, /: Merged revisions 339831 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r339831 | igorg | 2011-10-08 22:01:35 +0700
(Сбт, 08 Окт 2011) | 14 lines Merged revisions 339830 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) |
8 lines Fix char array cast as short array in send_client()
function (for ARM platform) (closes issue ASTERISK-17314)
Reported by: jjoshua ........ ................
2011-10-07 19:37 +0000 [r339721-339778] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_url.c: Merged revisions 339777 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r339777 | rmudgett | 2011-10-07 14:36:24 -0500
(Fri, 07 Oct 2011) | 12 lines Merged revisions 339776 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011)
| 5 lines Initialize option flags for SendURL application.
(closes issue ASTERISK-18574) Reported by: marcelloceschia
........ ................
* /: Recorded merge of revisions 339681 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r339681 | wedhorn | 2011-10-06 15:47:08 -0500 (Thu, 06 Oct 2011)
| 10 lines Fixed segfault on core stop gracefully. There was an
issue that the cap and confcap pointers for each line and device
were being memcpy'd so they all pointed to the same
ast_format_cap. On destroying, a segfault occured on the second
call to the same struct. skinny reload now works again as well.
Tested by snuff (in trunk) and myself. ........
* /, configure, include/asterisk/autoconfig.h.in, configure.ac,
autoconf/ast_ext_lib.m4: Merged revisions 339720 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r339720 | rmudgett | 2011-10-06 17:58:40 -0500
(Thu, 06 Oct 2011) | 27 lines Merged revisions 339719 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011)
| 20 lines Fix regression in configure script for libpri
capability checks. JIRA AST-598 added the PRI_L2_PERSISTENCE
option to fix BRI PTMP TE layer 2 persistence issues with some
telcos. ASTERISK-18535 attempted to fix the unexpected
requirement that libpri *must* have that feature to work with
Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI
optional features required. Unfortunately, I thought
AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri
and deleted those lines for libpri. The result was the
HAVE_PRI_xxx defines that control the ability to use optional
libpri features were also deleted. * Created
AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
features in a library that the source code could take advantage
of if the code supports the feature. (closes issue
ASTERISK-18687) Reported by: Norbert Tested by: rmudgett ........
................
2011-10-06 20:18 +0000 [r339680] Damien Wedhorn <voip@facts.com.au>
* channels/chan_skinny.c: Fixed segfault on core stop gracefully.
There was an issue that the cap and confcap pointers for each
line and device were being memcpy'd so they all pointed to the
same ast_format_cap. On destroying, a segfault occured on the
second call to the same struct. skinny reload now works again as
well. Tested by snuff and myself.
2011-10-06 17:54 +0000 [r339627] Richard Mudgett <rmudgett@digium.com>
* main/udptl.c, /, channels/chan_sip.c: Merged revisions 339626 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r339626 | rmudgett | 2011-10-06 12:53:00 -0500
(Thu, 06 Oct 2011) | 25 lines Merged revisions 339625 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011)
| 18 lines Fix debugging messages generated by 'udptl debug'. *
Makes chan_sip set the tag to the channel name. * Fixes received
debug message sequence number. * Removed tx/rx debug message type
since it was hard coded to 0. * Made udptl.c logged message
header consistent if possible: "UDPTL (%s): ". * Removed unused
rx_expected_seq_no from struct ast_udptl. (closes issue
ASTERISK-18401) Reported by: Kevin P. Fleming Patches:
jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: Matthew Nicholson ........ ................
2011-10-06 13:43 +0000 [r339587] Leif Madsen <leif@leifmadsen.com>
* build_tools/prep_tarball: Merged revisions 339586 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r339586 | lmadsen | 2011-10-06 08:43:21 -0500
(Thu, 06 Oct 2011) | 16 lines Merged revisions 339566 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339566 | lmadsen | 2011-10-05 16:30:11 -0500 (Wed, 05 Oct 2011)
| 8 lines Update prep_tarball script to download pre-exported
documentation. I've updated the prep_tarball script to now
download the pre-exported documentation from the Asterisk wiki.
This will give us more control over what is being included in the
tarball releases, and will make both the PDF and HTML exported
documentation look much better (especially when viewing from a
console). (Closes issue ASTERISK-18677) ........ ................
2011-10-05 17:02 +0000 [r339510-339513] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, /: Merged revisions 339512 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r339512 | rmudgett | 2011-10-05 12:01:46 -0500
(Wed, 05 Oct 2011) | 9 lines Merged revisions 339511 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05
Oct 2011) | 1 line Fix Dial F option notes formatting. ........
................
* main/manager.c, /: Merged revisions 339508 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r339508 | rmudgett | 2011-10-05 11:35:02 -0500
(Wed, 05 Oct 2011) | 18 lines Merged revisions 339504,339506 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339504 | rmudgett | 2011-10-05 11:26:45 -0500 (Wed, 05 Oct 2011)
| 7 lines Add missing documentation of required AMI action
Challenge AuthType header. (closes issue ASTERISK-18554) Reported
by: Vlad Povorozniuc Patches:
__20110919-manager-challenge-docs.patch.txt (license #4999) patch
uploaded by Leif Madsen ........ r339506 | rmudgett | 2011-10-05
11:32:03 -0500 (Wed, 05 Oct 2011) | 1 line Fix XML error in AMI
action Challenge. ........ ................
2011-10-05 16:35 +0000 [r339509] Matthew Nicholson <mnicholson@digium.com>
* /, res/res_fax.c: Merged revisions 339507 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r339507 | mnicholson | 2011-10-05 11:32:59 -0500
(Wed, 05 Oct 2011) | 10 lines Merged revisions 339505 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct
2011) | 3 lines The app name in the documentation must match what
we register the application as. ........ ................
2011-10-05 06:50 +0000 [r339464-339465] Gregory Nietsky <gregory@distrotech.co.za>
* res/res_fax.c, include/asterisk/res_fax.h, CHANGES: Add generic
faxdetect framehook to res_fax Added func
FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no to enable dialplan
faxdetect allowing more flexibility. as soon as a fax tone is
detected the framehook is removed. there is a penalty involved in
running this framehook on non G711 channels as they will be
transcoded. CNG tone is suppresed using the SQUELCH flag to allow
WaitForNoise to be run on the channel to detect Voice. (Closes
issue ASTERISK-18569) Reported by: Myself Reviewed by: Matthew
Nicholson, Kevin Fleming Review:
https://reviewboard.asterisk.org/r/1116/
* /, res/res_fax.c: Merged revisions 339463 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) |
9 lines Only change the capabilities on the gateway when the
session is been destroyed there is still a race condition that
ends in a segfault. if the caps are changed the logic in
res_fax_spandsp will run T30 code not gateway code to end the
session. this has been experienced on a "slower" under spec
system. ........
2011-10-04 22:59 +0000 [r339408] Richard Mudgett <rmudgett@digium.com>
* Makefile, /: Merged revisions 339407 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r339407 | rmudgett | 2011-10-04 17:56:25 -0500
(Tue, 04 Oct 2011) | 15 lines Merged revisions 339406 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339406 | rmudgett | 2011-10-04 17:54:15 -0500 (Tue, 04 Oct 2011)
| 8 lines Make always create the MOH directory
(/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported
by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license
#5903) patch uploaded by abelbeck Tested by: abelbeck, Michael
Keuter ........ ................
2011-10-04 19:51 +0000 [r339315-339354] Jonathan Rose <jrose@digium.com>
* /, main/say.c: Merged revisions 339353 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r339353 | jrose | 2011-10-04 14:44:02 -0500
(Tue, 04 Oct 2011) | 18 lines Merged revisions 339352 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339352 | jrose | 2011-10-04 14:33:12 -0500 (Tue, 04 Oct 2011) |
12 lines Removes improper use of sound 'and' in German language
mode from application saynumber Asterisk would say 'Five hundert
und sechs und zwanzig' instead of 'Five hundert sechs und
zwanzig'... which is both weird sounding and wrong. This patch
makes sure Asterisk will only say the 'and' word between the
single digit and double digit places. (closes issue
ASTERISK-18212) Reported By: Lionel Elie Mamane Patches:
upstream_germand_no_and.diff (License #5402) uploaded by Lionel
Elie Mamane ........ ................
* /, res/res_jabber.c: Merged revisions 339298 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r339298 | jrose | 2011-10-04 09:09:50 -0500
(Tue, 04 Oct 2011) | 19 lines Merged revisions 339297 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) |
13 lines Reverting revision 333265 due to component connection
problems it introduces. I'm going to attempt some generic
res_jabber cleanup and come up with a new fix for this problem,
but first it seems prudent to remove this rather broad attempt to
fix it and instead approach this problem either from the same
angle but looking only at canceling (or possibly rescheduling)
the send when we absolutely know it will cause a segfault or, if
that can't be easily accomplished, strictly from the devstate
side of things. Also, I'm pretty sure a lot of the code in
res_jabber isn't thread safe. (issue ASTERISK-18626) (issue
ASTERISK-18078) ........ ................
2011-10-04 12:27 +0000 [r339262] Alexandr Anikin <may@telecom-service.ru>
* /, addons/ooh323c/src/memheap.c: Merged revisions 339245 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r339245 | may | 2011-10-04 15:49:49 +0400 (Tue,
04 Oct 2011) | 9 lines Merged revisions 339244 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339244 | may | 2011-10-04 15:44:55 +0400 (Tue, 04 Oct 2011) | 2
lines fix forget declaration in previous change ........
................
2011-10-04 09:43 +0000 [r339206] Olle Johansson <oej@edvina.net>
* main/manager.c, CHANGES: Generate error message when AMI action
originate extension doesn't exist Review:
https://reviewboard.asterisk.org/r/1445/ Is this a bug or a new
feature? No responses on Asterisk-dev so I'm committing to trunk
only.
2011-10-03 20:13 +0000 [r339146-339149] Leif Madsen <leif@leifmadsen.com>
* channels/chan_sip.c: Merged revisions 339148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r339148 | lmadsen | 2011-10-03 15:13:16 -0500
(Mon, 03 Oct 2011) | 14 lines Merged revisions 339147 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339147 | lmadsen | 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011)
| 6 lines Remove duplicated Maxforwards line in AMI output.
(Closes issue ASTERISK-18637) Reported by: Jacek Konieczny
Patches: asterisk-sipshowpeer.patch (License #6298) uploaded by
Jacek Konieczny ........ ................
* apps/app_dial.c: Merged revisions 339145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r339145 | lmadsen | 2011-10-03 14:55:15 -0500
(Mon, 03 Oct 2011) | 13 lines Merged revisions 339144 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011)
| 6 lines Make documentation for Dial() options 'F' and 'F()'
more clear. (Closes issue ASTERISK-18646) Reported by: Physis
Heckman Tested by: Richard Mudgett ........ ................
2011-10-03 19:16 +0000 [r339091] Alexandr Anikin <may@telecom-service.ru>
* /, addons/ooh323c/src/memheap.c: Merged revisions 339089 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r339089 | may | 2011-10-03 22:52:55 +0400 (Mon,
03 Oct 2011) | 10 lines Merged revisions 339087 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339087 | may | 2011-10-03 22:42:49 +0400 (Mon, 03 Oct 2011) | 4
lines destroy memheap mutex properly before memheap deleted (fix
memory leak occured after r304950 changes with DEBUG_THREAD
compile option) ........ ................
2011-10-03 18:58 +0000 [r339090] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c, main/file.c: Merged revisions 339088 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r339088 | twilson | 2011-10-03 11:44:27 -0700
(Mon, 03 Oct 2011) | 17 lines Merged revisions 339086 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011)
| 10 lines Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more
places After the change in r336294, the new
AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite
happens. If we receive a re-invite from a device the
waitstream_core was not aware of the new control frame and would
drop the call. (closes issue ASTERISK-18610) Reported by:
Kristijan_Vrban ........ ................
2011-10-03 15:55 +0000 [r339021-339046] Matthew Nicholson <mnicholson@digium.com>
* /, res/res_fax.c: Merged revisions 339045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r339045 | mnicholson | 2011-10-03 10:54:55 -0500 (Mon, 03 Oct
2011) | 4 lines Ported ast_fax_caps_to_str() to 10, not sure why
it wasn't already here. This function prints a list of caps
instead of a hex bitfield. ........
* /, res/res_fax.c: Merged revisions 339043 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r339043 | mnicholson | 2011-10-03 10:41:36 -0500 (Mon, 03 Oct
2011) | 2 lines Don't clear the AST_FAX_TECH_MULTI_DOC flag right
after we set it. ........
* /, res/res_fax.c: Merged revisions 339011 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r339011 | mnicholson | 2011-10-03 10:19:44 -0500 (Mon, 03 Oct
2011) | 2 lines properly remove the AST_FAX_TECH_GATEWAY flag
(instead of setting all of the other flags) ........
2011-10-03 14:40 +0000 [r338905-338998] Gregory Nietsky <gregory@distrotech.co.za>
* /, CHANGES: Merged revisions 338997 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r338997 | irroot | 2011-10-03 16:38:25 +0200 (Mon, 03 Oct 2011) |
1 line Documentation noting the extension of CHANNEL() for
chan_ooh323 ........
* addons/chan_ooh323.c, /, funcs/func_channel.c: Merged revisions
338995 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r338995 | irroot | 2011-10-03 16:21:40 +0200 (Mon, 03 Oct 2011) |
6 lines Remove the channel function OOH323() and place its
options into CHANNEL() channel drivers should not have there own
dialplan functions. ........
* /, res/res_fax.c: Merged revisions 338950 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r338950 | irroot | 2011-10-03 11:37:59 +0200 (Mon, 03 Oct 2011) |
14 lines Fixup a race condition in res_fax.c where
FAXOPT(gateway)=no will turn off the gateway but the framehook is
not destroyed. this problem happens when a gateway is attempted
in the dialplan and the device is not available i may want to do
fax to mail in the server it will not be allowed. instead of
checking only AST_FAX_TECH_GATEWAY also check gateway_id Reverts
338904 Fix some white space. ........
* /, res/res_fax.c: Merged revisions 338904 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r338904 | irroot | 2011-10-02 16:17:32 +0200 (Sun, 02 Oct 2011) |
8 lines Remove T38 Gateway capability when detaching framehook.
SET(FAXOPT(gateway)=no) does not remove the capability when
detaching the framehook. small patch to fix this problem.
........
2011-10-01 01:56 +0000 [r338855] TransNexus OSP Development <support@transnexus.com>
* configure: Update "configure" based on r338139.
2011-09-30 22:08 +0000 [r338802] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 338801 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r338801 | rmudgett | 2011-09-30 17:06:48 -0500
(Fri, 30 Sep 2011) | 19 lines Merged revisions 338800 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011)
| 12 lines Fix segfault in analog_ss_thread() not checking
ast_read() for NULL. NOTE: The problem was reported against
v1.6.2. It is unlikely to ever happen on v1.8 and above since
chan_dahdi.c:analog_ss_thread() is unlikely to be used. The
version in sig_analog.c has largely replaced it. (closes issue
ASTERISK-18648) Reported by: Stephan Bosch Patches:
jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: Stephan Bosch ........ ................
2011-09-30 19:25 +0000 [r338755] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Formatting changes only --Denna och
nedanstående rader kommer inte med i loggmeddelandet-- M
channels/chan_sip.c
2011-09-30 18:59 +0000 [r338720] Jonathan Rose <jrose@digium.com>
* /, configs/queues.conf.sample: Merged revisions 338719 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r338719 | jrose | 2011-09-30 13:55:27 -0500
(Fri, 30 Sep 2011) | 9 lines Merged revisions 338718 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r338718 | jrose | 2011-09-30 13:54:30 -0500 (Fri, 30 Sep
2011) | 1 line Adds documentation for QueueMemberStatus event
generation ........ ................
2011-09-30 16:40 +0000 [r338665] Richard Mudgett <rmudgett@digium.com>
* /, channels/chan_sip.c: Fix formatting of AMI header for SIP show
peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes
issue ASTERISK-18649) Reported by: Jacek Konieczny Patches:
asterisk-sipshowpeer_response_end.patch (license #6298) patch
uploaded by Jacek Konieczny ........ Merged revisions 338663 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged
revisions 338664 from
http://svn.asterisk.org/svn/asterisk/branches/10
2011-09-30 13:21 +0000 [r338623] Olle Johansson <oej@edvina.net>
* main/features.c: Preserve DTMF length in main/features.c Review:
https://reviewboard.asterisk.org/r/1463/ A small part of much
larger work with DTMF duration in Asterisk, funded by IPvision AS
in Denmark. Thanks to irroot for the review!
2011-09-29 21:16 +0000 [r338557] Paul Belanger <paul.belanger@polybeacon.com>
* tests/test_security_events.c, /, tests/test_locale.c,
tests/test_logger.c, tests/test_dlinklists.c,
tests/test_linkedlists.c, tests/test_amihooks.c: Merged revisions
338556 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r338556 | pabelanger | 2011-09-29 17:14:34 -0400
(Thu, 29 Sep 2011) | 9 lines Merged revisions 338555 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r338555 | pabelanger | 2011-09-29 17:12:21 -0400 (Thu,
29 Sep 2011) | 2 lines Test modules should depend on the
TEST_FRAMEWORK flag ........ ................
2011-09-29 20:55 +0000 [r338553] Jason Parker <jparker@digium.com>
* /, tests/test_db.c, tests/test_netsock2.c: Merged revisions
338552 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r338552 | qwell | 2011-09-29 15:54:55 -0500
(Thu, 29 Sep 2011) | 9 lines Merged revisions 338551 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r338551 | qwell | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep
2011) | 1 line Test modules have a support level of core.
........ ................
2011-09-29 12:22 +0000 [r338435] Gregory Nietsky <gregory@distrotech.co.za>
* /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
revisions 338417 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r338417 | irroot | 2011-09-29 14:16:42 +0200
(Thu, 29 Sep 2011) | 19 lines Merged revisions 338416 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) |
12 lines The rtptimeout setting is ignored on a per peer basis.
Not only is the rtptimeout ignored in some cases but rtpkeepalive
and rtpholdtimeout is affected. this commit also removes
rtptimeout/rtpholdtimeout on text rtp. (closes issue
ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452
........ ................
2011-09-29 12:03 +0000 [r338377-338415] Olle Johansson <oej@edvina.net>
* cdr/cdr_pgsql.c, CHANGES: Add CLI command "cdr show pgsql status"
based on "cdr mysql status" Review:
https://reviewboard.asterisk.org/r/923/ Thanks all for the code
reviews and feedback.
* res/res_agi.c: Just formatting.
2011-09-28 22:38 +0000 [r338284-338324] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_pri.c: Merged revisions 338323 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r338323 | rmudgett | 2011-09-28 17:36:57 -0500
(Wed, 28 Sep 2011) | 12 lines Merged revisions 338322 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011)
| 5 lines Make duplicate call ptr warning message more helpful. *
Adds the value of the call ptr to the duplicate call ptr message
to help trace why there is a duplicate call ptr. ........
................
* include/asterisk/logger.h, /: Merged revisions 338253 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r338253 | rmudgett | 2011-09-28 16:22:05 -0500
(Wed, 28 Sep 2011) | 14 lines Merged revisions 338235 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r338235 | rmudgett | 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011)
| 7 lines Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE
declaration. (closes issue ASTERISK-17973) Reported by: Luke H
Patches: logger_h.patch (license #6278) patch uploaded by Luke H
........ ................
2011-09-28 20:55 +0000 [r338229] Jason Parker <jparker@digium.com>
* build_tools/cflags.xml, channels/chan_usbradio.c,
build_tools/cflags-devmode.xml, agi/agi.xml, utils/utils.xml, /,
build_tools/embed_modules.xml, tests/test_db.c,
tests/test_netsock2.c: Merged revisions 338228 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r338228 | qwell | 2011-09-28 15:54:35 -0500
(Wed, 28 Sep 2011) | 9 lines Merged revisions 338227 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep
2011) | 1 line Add support levels to non-module sections of
menuselect (cflags, utils, etc). ........ ................
2011-09-28 20:28 +0000 [r338226] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 338225 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r338225 | rmudgett | 2011-09-28 15:26:39 -0500
(Wed, 28 Sep 2011) | 12 lines Merged revisions 338224 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011)
| 5 lines Fix chan_dahd compiling with gcc 4.6 when PRI and SS7
not present. (closes issue ASTERISK-18357) Reported by: Matthew
Nicholson ........ ................
2011-09-28 17:00 +0000 [r338187-338188] Terry Wilson <twilson@digium.com>
* CHANGES: Update CHANGES to reflect autopausebusy not being in
Asterisk 10
* configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add
autopausebusy and autopauseunavail queue options Make it possible
to autopause on a busy or unavailable response from a device.
(closes issue ASTERISK-16112) Reported by: jlpedrosa Patches:
autopausebusy.txt by twilson Review:
https://reviewboard.asterisk.org/r/1399/
2011-09-28 07:30 +0000 [r338136-338139] TransNexus OSP Development <support@transnexus.com>
* configure.ac: Updated for checking OSP Toolkit version 4.0.0.
* apps/app_osplookup.c: Updated for OSP Toolkit 4.0.0.
2011-09-27 20:15 +0000 [r338086] Paul Belanger <paul.belanger@polybeacon.com>
* /, apps/app_macro.c: Merged revisions 338085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r338085 | pabelanger | 2011-09-27 16:13:14 -0400
(Tue, 27 Sep 2011) | 9 lines Merged revisions 338084 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r338084 | pabelanger | 2011-09-27 16:10:13 -0400 (Tue,
27 Sep 2011) | 2 lines Upgrade app_macro to core ........
................
2011-09-27 12:45 +0000 [r338042] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Whitespace (red blobs) fixes
2011-09-26 19:40 +0000 [r337975] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /,
include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c,
include/asterisk/channel.h, main/cel.c, main/manager.c,
funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c,
main/logger.c, cel/cel_sqlite3_custom.c, cdr/cdr_custom.c,
cdr/cdr_manager.c, apps/app_voicemail.c: Merged revisions 337974
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r337974 | rmudgett | 2011-09-26 14:35:23 -0500
(Mon, 26 Sep 2011) | 37 lines Merged revisions 337973 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011)
| 30 lines Fix deadlock when using dummy channels. Dummy channels
created by ast_dummy_channel_alloc() should be destoyed by
ast_channel_unref(). Using ast_channel_release() needlessly grabs
the channel container lock and can cause a deadlock as a result.
* Analyzed use of ast_dummy_channel_alloc() and made use
ast_channel_unref() when done with the dummy channel. (Primary
reason for the reported deadlock.) * Made
app_dial.c:dial_exec_full() not call ast_call() holding any
channel locks. Chan_local could not perform deadlock avoidance
correctly. (Potential deadlock exposed by this issue. Secondary
reason for the reported deadlock since the held lock was part of
the deadlock chain.) * Fixed some uses of
ast_dummy_channel_alloc() not checking the returned channel
pointer for failure. * Fixed some potential chan=NULL pointer
usage in func_odbc.c. Protected by testing the bogus_chan value.
* Fixed needlessly clearing a 1024 char auto array when setting
the first char to zero is enough in manager.c:action_getvar().
(closes issue ASTERISK-18613) Reported by: Thomas Arimont
Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch
uploaded by rmudgett Tested by: Thomas Arimont ........
................
2011-09-23 19:20 +0000 [r337855-337910] Gregory Nietsky <gregory@distrotech.co.za>
* /, contrib/init.d/rc.archlinux.asterisk: Merged revisions 337902
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r337902 | irroot | 2011-09-23 21:18:14 +0200
(Fri, 23 Sep 2011) | 10 lines Merged revisions 337898 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) |
4 lines Spelling fix ........ ................
* /, apps/app_queue.c: Merged revisions 337840 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r337840 | irroot | 2011-09-23 10:39:22 +0200
(Fri, 23 Sep 2011) | 17 lines Merged revisions 337839 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) |
11 lines Make sure a CDR is on the stack for call in the Queue.
Only let update_cdr act on the last CDR in the stack. In some
circumstances [Attended transfer to queue] a CDR record is not
inserted for this call where it should. (closes issue
ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266
........ ................
2011-09-23 00:47 +0000 [r337776] Russell Bryant <russell@russellbryant.com>
* /, configs/res_pktccops.conf.sample: Merged revisions 337775 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r337775 | russell | 2011-09-22 19:45:35 -0500
(Thu, 22 Sep 2011) | 18 lines Merged revisions 337774 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011)
| 11 lines Comment out entries in sample res_pktccops.conf. With
these options enabled, they can cause Asterisk to freak out by
SYN flooding a network and eating the CPU. Obviously it would be
good to fix the code so that this can't happen, but we can at
least change the default configuration so it doesn't happen. This
was reported downstream to the Fedora issue tracker:
https://bugzilla.redhat.com/show_bug.cgi?id=658431 ........
................
2011-09-22 21:42 +0000 [r337722] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_pri.c: Merged revisions 337721 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r337721 | rmudgett | 2011-09-22 16:37:41 -0500
(Thu, 22 Sep 2011) | 25 lines Merged revisions 337720 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011)
| 18 lines Made ISDN not add numbering plan prefix strings to
empty numbers. When the Caller-ID is restricted, the expected
behavior is for the Caller-ID to be blank. In chan_dahdi, the
national prefix is placed onto the Caller-ID number even if it is
restricted (empty) causing the Caller-ID to be the national
prefix rather than blank. This behavior was lost when sig_pri was
extracted from chan_dahdi. * Made not add prefix strings to empty
connected line, calling, and ANI number strings. (closes issue
ASTERISK-18577) Reported by: Kris Shaw Patches:
jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: Kris Shaw ........ ................
2011-09-22 16:35 +0000 [r337600] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c, include/asterisk/event_defs.h,
main/security_events.c, channels/sip/security_events.c (added),
main/event.c, CHANGES, channels/sip/include/security_events.h
(added), channels/sip/include/sip.h,
include/asterisk/security_events_defs.h,
configs/logger.conf.sample: Merged revisions 337595,337597 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
........ r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep
2011) | 12 lines Generate Security events in chan_sip using new
Security Events Framework Security Events Framework was added in
1.8 and support was added for AMI to generate events at that
time. This patch adds support for chan_sip to generate security
events. (closes issue ASTERISK-18264) Reported by: Michael L.
Young Patches: security_events_chan_sip_v4.patch (license #5026)
by Michael L. Young Review:
https://reviewboard.asterisk.org/r/1362/ ........ r337597 | jrose
| 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines Forgot
to svn add new files to r337595 Part of Generating security
events for chan_sip (issue ASTERISK-18264) Reported by: Michael
L. Young Patches: security_events_chan_sip_v4.patch (License
#5026) by Michael L. Young Reviewboard:
https://reviewboard.asterisk.org/r/1362/ ........
2011-09-22 11:46 +0000 [r337432-337543] Gregory Nietsky <gregory@distrotech.co.za>
* /, res/res_srtp.c: Merged revisions 337542 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r337542 | irroot | 2011-09-22 13:44:22 +0200
(Thu, 22 Sep 2011) | 14 lines Merged revisions 337541 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) |
8 lines Add warned to ast_srtp to prevent errors on each frame
from libsrtp The first 9 frames are not reported as some devices
dont use srtp from first frame these are suppresed. the warning
is then output only once every 100 frames. ........
................
* /, channels/chan_h323.c: Merged revisions 337487 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r337487 | irroot | 2011-09-22 11:26:26 +0200
(Thu, 22 Sep 2011) | 16 lines Merged revisions 337486 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) |
10 lines If IP address is used in chan_h323 host parameter of
peer configuration. module tries to resolve IP address to IP
address and fails. Simple fix to set family of socket this is a
hangover from ipv6 changes. (closes issue ASTERISK-18237) (issue
ASTERISK-17278) (issue ASTERISK-17500) ........ ................
* main/channel.c, /: Merged revisions 337431 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r337431 | irroot | 2011-09-22 08:29:09 +0200
(Thu, 22 Sep 2011) | 25 lines Merged revisions 337430 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) |
19 lines Its possible to loose audio on ast_write when the
channel is not transcoded correctly. in the case of DAHDI the
channel is hungup. This patch tries to "fix" the problem and make
the channel compatiable and warn the user of this problem. Please
note there is a underlying problem with codec negotion this does
not fix the problem it does try to rectify it and prevent loss of
service. Review: https://reviewboard.asterisk.org/r/1442/ (closes
issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue
ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325)
(issue ASTERISK-18422) ........ ................
2011-09-21 21:26 +0000 [r337343-337385] Tilghman Lesher <tilghman@meg.abyt.es>
* /, apps/app_voicemail.c: More silly spacing changes ..... Merged
revisions 337353 from
http://svn.asterisk.org/svn/asterisk/branches/1.8 ..... Merged
revisions 337380 from
http://svn.asterisk.org/svn/asterisk/branches/10
* /, apps/app_voicemail.c: ................ ........ Dumb little
spacing fix. ........ Merged revisions 337344 from
http://svn.asterisk.org/svn/asterisk/branches/1.8
................ Merged revisions 337345 from
http://svn.asterisk.org/svn/asterisk/branches/10
* funcs/func_curl.c, /: ................ ........ Escape commas in
keys and values, when keys and values are enumerated by commas.
Review: https://reviewboard.asterisk.org/r/1433 ........ Merged
revisions 337325 from
https://origsvn.digium.com/svn/asterisk/branches/1.8
................ Merged revisions 337342 from
https://origsvn.digium.com/svn/asterisk/branches/10
2011-09-21 11:21 +0000 [r337262-337283] Gregory Nietsky <gregory@distrotech.co.za>
* /, configs/sip.conf.sample: Merged revisions 337263 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r337263 | irroot | 2011-09-21 13:15:48 +0200 (Wed, 21 Sep 2011) |
1 line Whitespace fixup from SRTP patch ........
* /, apps/app_originate.c, CHANGES: Merged revisions 337261 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
........ r337261 | irroot | 2011-09-21 12:42:06 +0200 (Wed, 21
Sep 2011) | 10 lines Adds a timeout argument to app_originate the
default is 30s this will be used if the timout supplied is
invalid or no timeout is supplied. Contributed by: jacco (thank
you for the work) Review:
https://reviewboard.asterisk.org/r/1310/ ........
2011-09-21 09:39 +0000 [r337179-337220] Olle Johansson <oej@edvina.net>
* main/pbx.c, /, CHANGES, configs/extensions.conf.sample: Merged
revisions 337219 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13
lines Make ast_pbx_run() not default to s@default if extension is
not found Review: https://reviewboard.asterisk.org/r/1446/ This
is a bug - or architecture mistake - that has been in Asterisk
for a very long time. It was exposed by the AMI originate action
and possibly some other applications. Most channel drivers checks
if an extension exists BEFORE starting a pbx on an inbound call,
so most calls will not depend on this issue. Thanks everyone
involved in the review and on IRC and the mailing list for a
quick review and all the feedback. (closes issue ASTERISK-18578)
........
* res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES:
Merged revisions 337178 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r337178 | oej | 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) | 14
lines Change strictrtp option to default to yes in the RTP module
Suggested by Kapejod on Facebook Review:
https://reviewboard.asterisk.org/r/1448/ (closes issue
ASTERISK-18587) Thanks for quick feedback to kpfleming and
Tilghman --Denna och nedanstående rader kommer inte med i
loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M
res/res_rtp_asterisk.c ........
2011-09-20 23:02 +0000 [r337124] Matthew Jordan <mjordan@digium.com>
* apps/app_dial.c, include/asterisk/app.h, apps/app_meetme.c,
apps/app_minivm.c, main/app.c, apps/app_confbridge.c,
apps/app_followme.c, apps/app_voicemail.c: Merged revisions
337120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r337120 | mjordan | 2011-09-20 17:49:36 -0500
(Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011)
| 21 lines Fix for incorrect voicemail duration in external
notifications This patch fixes an issue where the voicemail
duration was being reported with a duration significantly less
than the actual sound file duration. Voicemails that contained
mostly silence were reporting the duration of only the sound in
the file, as opposed to the duration of the file with the
silence. This patch fixes this by having two durations reported
in the __ast_play_and_record family of functions - the
sound_duration and the actual duration of the file. The
sound_duration, which is optional, now reports the duration of
the sound in the file, while the actual full duration of the file
is reported in the duration parameter. This allows the voicemail
applications to use the sound_duration for minimum duration
checking, while reporting the full duration to external parties
if the voicemail is kept. (issue ASTERISK-2234) (closes issue
ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad
House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review:
https://reviewboard.asterisk.org/r/1443 ........ ................
2011-09-20 22:54 +0000 [r337121-337123] Richard Mudgett <rmudgett@digium.com>
* /, funcs/func_strings.c: Merged revisions 337119 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r337119 | rmudgett | 2011-09-20 17:47:45 -0500 (Tue, 20 Sep 2011)
| 16 lines Fix crash with STRREPLACE function. The
ast_func_read() function calls the .read2 callback with the len
parameter set to zero indicating no size restrictions on the
supplied ast_str buffer. The value was used to dimension a local
starts[] array with the array subsequently used. * Reworked the
strreplace() function to perform the string replacement in a
straight forward manner. Eliminated the need for the starts[]
array. (closes issue ASTERISK-18545) Reported by: Federico Alves
Patches: jira_asterisk_18545_v10.patch (license #5621) patch
uploaded by rmudgett Tested by: rmudgett, Federico Alves ........
* /: Updated 10 merge property.
* /: Restore branch-10 merge properties.
2011-09-20 22:29 +0000 [r337117] Leif Madsen <leif@leifmadsen.com>
* /, contrib/init.d/rc.redhat.asterisk: Merged revisions 337115 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011)
| 7 lines Update RedHat Init script to work with Heartbeat. The
current RedHat init script was not LSB compatible. This change
will make it LSB compatible so that it can work correctly with
Heartbeat. (Closes issue ASTERISK-18253) Reported by: c0rnoTa
........
2011-09-20 21:05 +0000 [r337063] Kinsey Moore <kmoore@digium.com>
* main/pbx.c, /, tests/test_pbx.c: Merged revisions 337062 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r337062 | kmoore | 2011-09-20 16:05:01 -0500
(Tue, 20 Sep 2011) | 18 lines Merged revisions 337061 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) |
11 lines Make CANMATCH with the new pattern match engine behave
more like the old one When checking an extension for E_CANMATCH
using the new extension matching algorithm, an exact match was
not returned as a possible match resulting in the queue failing
to allow a caller to exit on DTMF. This removes the requirement
that an extension be longer than acquired digits for an
E_CANMATCH operation to succeed. (closes issue ASTERISK-18044)
Review: https://reviewboard.asterisk.org/r/1367/ ........
................
2011-09-20 19:13 +0000 [r336988-337009] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_ss7.c: Merged revisions 337008 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r337008 | rmudgett | 2011-09-20 14:12:24 -0500
(Tue, 20 Sep 2011) | 22 lines Merged revisions 337007 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011)
| 15 lines Check if a channel was created before using the
pointer in sig_ss7_new_ast_channel(). Fixes the crash in
ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing
libss7 access lock protection. * Prevent cancelling the
ss7_linkset() thread at inoportune times just like the
pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M
Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621)
patch uploaded by rmudgett (attached to related ASTERISK-17966)
........ ................
* /, channels/sig_ss7.c: Merged revisions 336978 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r336978 | rmudgett | 2011-09-20 13:14:40 -0500
(Tue, 20 Sep 2011) | 28 lines Merged revisions 336977 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011)
| 21 lines Fix deadlock from not releasing SS7 linkset lock.
sig_ss7_hangup() failed to release the SS7 linkset lock if the
call had the alreadyhungup flag set. * Made unlock the SS7
linkset lock in sig_ss7_hangup() if the alreadyhungup flag is
set. * Made ss7_start_call() not hold any locks while creating
the channel for an incoming call to prevent deadlock. * Made
ss7_grab() a void function, since it could never fail, to
simplify calling code. * Made obtain the channel lock to do
softhangup in some places. Patches: jira_ast_668_v1.8.patch
(license #5621) patch uploaded by rmudgett JIRA AST-668 ........
................
2011-09-20 16:56 +0000 [r336937] Gregory Nietsky <gregory@distrotech.co.za>
* channels/sip/sdp_crypto.c, /, channels/chan_sip.c,
channels/sip/include/sdp_crypto.h, channels/sip/include/srtp.h,
configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h:
Merged revisions 336936 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) |
14 lines Allow Setting Auth Tag Bit length Based on invite or
config option Update the SIP SRTP API to allow use of 32 or 80
bit taglen. Curently only 80 bit is supported. The outgoing
invite will use the taglen of the incoming invite preventing
one-way audio. (Closes issue ASTERISK-17895) Review:
https://reviewboard.asterisk.org/r/1173/ ........
2011-09-20 01:11 +0000 [r336879] Russell Bryant <russell@russellbryant.com>
* res/res_rtp_asterisk.c, /: Merged revisions 336878 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r336878 | russell | 2011-09-19 20:03:55 -0500
(Mon, 19 Sep 2011) | 43 lines Merged revisions 336877 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011)
| 36 lines Fix crashes in ast_rtcp_write(). This patch addresses
crashes related to RTCP handling. The backtraces just show a
crash in ast_rtcp_write() where it appears that the RTP instance
is no longer valid. There is a race condition with scheduled RTCP
transmissions and the destruction of the RTP instance. This patch
utilizes the fact that ast_rtp_instance is a reference counted
object and ensures that it will not get destroyed while a
reference is still around due to scheduled RTCP transmissions.
RTCP transmissions are scheduled and executed from the chan_sip
scheduler context. This scheduler context is processed in the SIP
monitor thread. The destruction of an RTP instance occurs when
the associated sip_pvt gets destroyed (which happens when the
sip_pvt reference count reaches 0). However, the SIP monitor
thread is not the only thread that can cause a sip_pvt to get
destroyed. The sip_hangup function, executed from a channel
thread, also decrements the reference count on a sip_pvt and
could cause it to get destroyed. While this is being changed
anyway, the patch also removes calling ast_sched_del() from
within the RTCP scheduler callback. It's not helpful. Simply
returning 0 prevents the callback from being rescheduled. (closes
issue ASTERISK-18570) Related issues that look like they are the
same problem: (issue ASTERISK-17560) (issue ASTERISK-15406)
(issue ASTERISK-15257) (issue ASTERISK-13334) (issue
ASTERISK-9977) (issue ASTERISK-9716) Review:
https://reviewboard.asterisk.org/r/1444/ ........
................
2011-09-19 22:28 +0000 [r336837] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 336792 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r336792 | twilson | 2011-09-19 17:13:34 -0500
(Mon, 19 Sep 2011) | 9 lines Merged revisions 336791 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19
Sep 2011) | 2 lines Don't interfere with T.38 reinvites This is
an update to the fix for ASTERISK-18340 and ASTERISK-17725
........ ................
2011-09-19 21:42 +0000 [r336735-336790] Tilghman Lesher <tilghman@meg.abyt.es>
* /, funcs/func_strings.c: Merged revisions 336789 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r336789 | tilghman | 2011-09-19 16:41:16 -0500 (Mon, 19 Sep 2011)
| 2 lines Ensure substring will not be found in the previous
match. ........
* Makefile, /, configure, include/asterisk/autoconfig.h.in,
main/Makefile, codecs/gsm/Makefile, configure.ac, Makefile.rules,
include/asterisk/optional_api.h: Merged revisions 336734 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r336734 | tilghman | 2011-09-19 15:29:40 -0500
(Mon, 19 Sep 2011) | 18 lines Merged revisions 336733 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011)
| 11 lines Various changes to allow 1.8 to compile on Mac OS X
Lion (10.7) * Makefile workaround for 10.6 extended to work on
10.7 and later. * Now uses the 'weak' symbol for Lion systems,
which no longer support 'weak_import' Closes ASTERISK-17612.
Closes ASTERISK-18213. Tested by: tilghman, oej. ........
................
2011-09-19 20:23 +0000 [r336732] Jonathan Rose <jrose@digium.com>
* /, apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c,
apps/app_morsecode.c, res/res_musiconhold.c, apps/app_queue.c,
apps/app_mixmonitor.c: Merged revisions 336717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r336717 | jrose | 2011-09-19 15:16:23 -0500
(Mon, 19 Sep 2011) | 14 lines Merged revisions 336716 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) |
7 lines Document applications that play audio and do not answer
unanswered calls. This patch is part of an effort to document
early media and its usage. If you are interested in contributing
to this documentation effort, there are probably other
applications worth documenting as well as an Asterisk wiki
article at
https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
........ ................
2011-09-19 19:03 +0000 [r336660-336662] Richard Mudgett <rmudgett@digium.com>
* apps/app_dial.c, /, UPGRADE-1.8.txt: Merged revisions 336659 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r336659 | rmudgett | 2011-09-19 13:51:19 -0500
(Mon, 19 Sep 2011) | 38 lines Merged revisions 336658 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011)
| 31 lines Made Dial d and H options no longer immediately
auto-answer the calling leg. The Dial d and H options break DTMF
attended transfer atxferdropcall option. 1) Party A calls party
B. 2) Party B does a DTMF attended transfer to Party C. If the
dialplan uses the Dial d or H options to call Party C then the
Dial application answers the call immediately before initiating
the call leg to Party C. The premature answer causes the transfer
code to not invoke the atxferdropcall=no behavior for a blonde
transfer since Party C has "answered". The transfer code thinks
that Party B has "consulted" with Party C when Party B hangs up
and completes the transfer to Party A. Party A now hears ringback
until Party C actually answers. ASTERISK-13294 Dial d option.
ASTERISK-11067 Dial H option to disconnect before answer. The
referenced issues made Dial answer with the d and H options
because many SIP and ISDN phones cannot send DTMF before the call
is connected. * Made require the dialplan to control when or if
the call needs to be answered to use the Dial application d and H
options. (The call is no longer surprise answered when using the
Dial d or H options.) Review:
https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA
AST-666 ........ ................
* /: Update merge 10 branch merge propterty.
* /: Restore 10 branch merge properties.
2011-09-19 16:22 +0000 [r336600] Jason Parker <jparker@digium.com>
* cel/cel_odbc.c, configs/cel_odbc.conf.sample, sounds/Makefile:
Remove weird mergeinfo props that make merges annoying sometimes.
2011-09-19 15:48 +0000 [r336574] Leif Madsen <leif@leifmadsen.com>
* /, contrib/scripts/get_ilbc_source.sh: Merged revisions 336572
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011)
| 7 lines Update get_ilbc_source.sh script to work again.
Recently iLBC support in Asterisk has changed after the
acquisition of GIPS by Google. More information about how this
may affect you is available in a blog post at:
http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/
........
2011-09-19 15:36 +0000 [r336571] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_pri.c: Merged revisions 336570 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r336570 | rmudgett | 2011-09-19 10:32:00 -0500
(Mon, 19 Sep 2011) | 11 lines Merged revisions 336569 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011)
| 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA
AST-675 ........ ................
2011-09-19 13:57 +0000 [r336505] Olle Johansson <oej@edvina.net>
* /, channels/chan_sip.c: Merged revisions 336502 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r336502 | oej | 2011-09-19 15:38:53 +0200 (Mån,
19 Sep 2011) | 12 lines Merged revisions 336501 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5
lines Add diversion header to a 302 redirect response if we have
diversion data (closes issue ASTERISK-18143) patch by oej
........ ................
2011-09-19 13:41 +0000 [r336503] Gregory Nietsky <gregory@distrotech.co.za>
* /, channels/chan_h323.c: Merged revisions 336500 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r336500 | irroot | 2011-09-19 15:31:50 +0200
(Mon, 19 Sep 2011) | 19 lines Merged revisions 336499 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) |
13 lines A long time ago in a galaxy far far away a IPv6 update
was made, chan_h323 was not updated causeing all to flee to
chan_ooh323. the brave Jedi [asterisk developers] pondered this
miscarrige of justice and restored order to the force for the
sake of closing out 2 old issues. (closes issue ASTERISK-17278)
(closes issue ASTERISK-17500) Reported by: dread, sybasesql
Tested by: irroot Reviewed by: IRC (russellb, kpfleming) ........
................
2011-09-19 12:20 +0000 [r336382-336453] Olle Johansson <oej@edvina.net>
* main/manager.c, /: Merged revisions 336441 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r336441 | oej | 2011-09-19 14:15:06 +0200 (Mån,
19 Sep 2011) | 9 lines Merged revisions 336440 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2
lines Make sure manager_debug option is reset at reload ........
................
* /, channels/chan_sip.c: Merged revisions 336381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r336381 | oej | 2011-09-19 12:05:00 +0200 (Mån,
19 Sep 2011) | 16 lines Merged revisions 336378 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9
lines Add missing unlock at MWI message sending time (closes
issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041)
by Gregory Hinton Nietsky Thanks to irrot for the reminder, to
Gregory for the patch! ........ ................
2011-09-16 22:12 +0000 [r336315-336317] Terry Wilson <twilson@digium.com>
* /, funcs/func_frame_trace.c: Merged revisions 336316 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r336316 | twilson | 2011-09-16 17:11:39 -0500
(Fri, 16 Sep 2011) | 9 lines Merged revisions 336314 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16
Sep 2011) | 2 lines Whitespace fix ........ ................
* /, funcs/func_frame_trace.c: Merged revisions 336313 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r336313 | twilson | 2011-09-16 17:07:00 -0500
(Fri, 16 Sep 2011) | 12 lines Merged revisions 336312 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011)
| 5 lines Add missing frame types to func_frame_trace Also casts
control frames to the proper enum so that the compile will catch
new additions. ........ ................
2011-09-16 21:20 +0000 [r336311] Jonathan Rose <jrose@digium.com>
* main/channel.c, main/rtp_engine.c, /, channels/chan_sip.c,
include/asterisk/frame.h: Merged revisions 336307 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r336307 | jrose | 2011-09-16 16:09:20 -0500
(Fri, 16 Sep 2011) | 20 lines Merged revisions 336294 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) |
13 lines Fix bad RTP media bridges in directmedia calls on peers
separated by multiple Asterisk nodes. In a situation involving
devices on separate Asterisk trunks, the remote RTP bridge would
break when starting a call with directmedia. This patch queues a
new type of control frame so that our RTP bridge loop can
properly detect when these situations occur and check to see if
peers need to be updated in order to send their media to the
proper location. (Closes issue ASTERISK-18340) Reported by:
Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk
Tested by: twilson, jrose ........ ................
2011-09-16 19:11 +0000 [r336236] Sean Bright <sean@malleable.com>
* /, UPGRADE-1.8.txt: Merged revisions 336235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r336235 | seanbright | 2011-09-16 15:10:39 -0400
(Fri, 16 Sep 2011) | 9 lines Merged revisions 336234 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri,
16 Sep 2011) | 2 lines Make a note that inotify won't work with
an NFS mounted spooler directory. ........ ................
2011-09-16 10:16 +0000 [r336095-336168] Gregory Nietsky <gregory@distrotech.co.za>
* channels/chan_misdn.c, /: Merged revisions 336167 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r336167 | irroot | 2011-09-16 12:12:03 +0200
(Fri, 16 Sep 2011) | 22 lines Merged revisions 336166 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) |
16 lines The round robin routing routine in chan_misdn.c is
broken. it rotates between ports but never checks the channels in
the ports. i have extensivly tested it and verified it works on 1
upto 4 ports. before the patch only 1 out of each port was used
now all are used as expected. (closes issue ASTERISK-18413)
Reported by: irroot Tested by: irroot Reviewed by: irroot Review:
https://reviewboard.asterisk.org/r/1410/ ........
................
* /, apps/app_queue.c: Merged revisions 336094 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r336094 | irroot | 2011-09-15 17:54:46 +0200
(Thu, 15 Sep 2011) | 26 lines Merged revisions 336093 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) |
20 lines Locking order in app_queue.c causes deadlocks. a channel
lock must never be held with the queues container lock held. the
deadlock occured on masquerade. the queues container lock is a
relic of the past the old queue module lock. with ao2 there is no
need to hold this lock when dealing with members this patch
removes unneeded locks. (closes issue ASTERISK-18101) (closes
issue ASTERISK-18487) Reported by: Paul Rolfe, Jason Legault
Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by: Matthew
Nicholson Review: https://reviewboard.asterisk.org/r/1402/
........ ................
2011-09-15 15:19 +0000 [r336092] David Vossel <dvossel@digium.com>
* /, main/format_cap.c: Merged revisions 336091 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r336091 | dvossel | 2011-09-15 10:19:10 -0500 (Thu, 15 Sep 2011)
| 2 lines Removes some no-op code found in format_cap.c. ........
2011-09-15 12:50 +0000 [r336043] Olle Johansson <oej@edvina.net>
* CREDITS, /, apps/app_meetme.c, CHANGES: Merged revisions 336042
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12
lines Meetme: Introducing a new option "k" to kill a conference
if there's only a single member left. When using Meetme as a
modular call bridge from third party applications, it's handy to
make it behave like a normal call bridge. When the second to last
person exists, the last person will be kicked out of the
conference when this option is enabled. (closes issue
ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/
Patch by oej, sponsored by ClearIT, Solna, Sweden ........
2011-09-15 08:40 +0000 [r335993] Gregory Nietsky <gregory@distrotech.co.za>
* /, channels/chan_agent.c: Merged revisions 335991 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r335991 | irroot | 2011-09-15 10:29:12 +0200
(Thu, 15 Sep 2011) | 17 lines Merged revisions 335978 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) |
11 lines lock the channel before calling ast_bridged_channel() to
prevent a seg fault. AMI agents list called on shutdown causes a
segfault, introducing proper locking will prevent this. (closes
issue ASTERISK-18092) Reported by: agustina Patches:
chan_agent.patch (License #5041) patch uploaded by irroot
........ ................
2011-09-14 18:38 +0000 [r335853-335913] Richard Mudgett <rmudgett@digium.com>
* /, configure, include/asterisk/autoconfig.h.in, configure.ac:
Merged revisions 335912 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r335912 | rmudgett | 2011-09-14 13:31:15 -0500
(Wed, 14 Sep 2011) | 20 lines Merged revisions 335911 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011)
| 13 lines Remove unnecessary libpri dependency checks in the
configure script. Using the --with-pri option with the configure
script generated an error about not having PRI_L2_PERSISTENCE if
you did not have the absolute latest libpri SVN checkout
installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the
configure.ac script seems to be for libraries that are dependent
upon other libraries and not necessarily for optional/added
features within a library. (closes issue ASTERISK-18535) Reported
by: Michael Keuter ........ ................
* channels/chan_dahdi.c, /: Merged revisions 335852 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r335852 | rmudgett | 2011-09-14 11:00:37 -0500
(Wed, 14 Sep 2011) | 18 lines Merged revisions 335851 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011)
| 11 lines Fixed cut-n-paste regression using the wrong variable.
Fixes the missing DAHDI channels when using the newer
chan_dahdi.conf sections for channel configuration. (closes issue
ASTERISK-18496) Reported by: Sean Darcy Patches:
jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: Sean Darcy, rmudgett ........
................
2011-09-14 13:29 +0000 [r335792] Matthew Nicholson <mnicholson@digium.com>
* main/manager.c, /: Merged revisions 335791 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r335791 | mnicholson | 2011-09-14 08:28:50 -0500
(Wed, 14 Sep 2011) | 11 lines Merged revisions 335790 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep
2011) | 4 lines The tech and data members of
fast_originate_helper are not string fields. ASTERISK-17709
........ ................
2011-09-13 22:11 +0000 [r335722] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_directed_pickup.c: Merged revisions 335721 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r335721 | rmudgett | 2011-09-13 17:10:44 -0500
(Tue, 13 Sep 2011) | 9 lines Merged revisions 335720 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13
Sep 2011) | 1 line Remove obsolete todo comment about
PICKUPRESULT. ........ ................
2011-09-13 21:52 +0000 [r335719] Paul Belanger <paul.belanger@polybeacon.com>
* main/dnsmgr.c: Additional updates for parsing dnsmgr.conf Review:
https://reviewboard.asterisk.org/r/1432/
2011-09-13 21:40 +0000 [r335718] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* main/asterisk.c: do parse defaultlanguage from asterisk.conf Do
parse the option "defaultlanguage" from the [options] section of
asterisk.conf, as in the sample config file. Otherwise the
build-time default language (normally "en") is always the default
one. Review: https://reviewboard.asterisk.org/r/1342/
Signed-off-by: Tzafrir Cohen (License #5035)
<tzafrir.cohen@xorcom.com> Original-Commit:
http://svn.digium.com/svn/asterisk/branches/1.8@335716
Original-Commit:
http://svn.digium.com/svn/asterisk/branches/10@335717
2011-09-13 18:56 +0000 [r335657] Tilghman Lesher <tilghman@meg.abyt.es>
* /, configure, configure.ac: Merged revisions 335656 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r335656 | tilghman | 2011-09-13 13:55:33 -0500
(Tue, 13 Sep 2011) | 11 lines Merged revisions 335655 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011)
| 4 lines Move mandatory checks closer to the beginning of the
file. If these are going to fail, they should fail as quickly as
possible. ........ ................
2011-09-13 18:49 +0000 [r335654] Matthew Nicholson <mnicholson@digium.com>
* main/pbx.c, main/manager.c, /: Merged revisions 335653 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r335653 | mnicholson | 2011-09-13 13:47:57 -0500
(Tue, 13 Sep 2011) | 12 lines Merged revisions 335618 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep
2011) | 5 lines Don't limit the size of appdata for manager
originate actions. ASTERISK-17709 Patch by: tilghman (with
modifications) ........ ................
2011-09-13 18:11 +0000 [r335555-335603] Paul Belanger <paul.belanger@polybeacon.com>
* UPGRADE.txt, main/dsp.c: Clean up dsp.conf parsing Review:
https://reviewboard.asterisk.org/r/1434/
* UPGRADE.txt, cdr/cdr_csv.c: Clean up cdr.conf parsing for [csv]
section Review: https://reviewboard.asterisk.org/r/1427/
* main/dnsmgr.c, UPGRADE.txt: Clean up dnsmgr.conf parsing Review:
https://reviewboard.asterisk.org/r/1432/
2011-09-13 07:35 +0000 [r335511] Russell Bryant <russell@russellbryant.com>
* include/asterisk/event.h, /, res/ais/evt.c, main/event.c: Merged
revisions 335510 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r335510 | russell | 2011-09-13 02:24:34 -0500
(Tue, 13 Sep 2011) | 22 lines Merged revisions 335497 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011)
| 15 lines Fix a crash in res_ais. This patch resolves a crash
observed in a load testing environment that involved the use of
the res_ais module. I observed some crashes where the event
delivery callback would get called, but the length parameter
incidcating how much data there was to read was 0. The code
assumed (with good reason I would think) that if this callback
got called, there was an event available to read. However, if the
rare case that there's nothing there, catch it and return instead
of blowing up. More specifically, the change always ensure that
the size of the received event in the cluster is always big
enough to be a real ast_event. Review:
https://reviewboard.asterisk.org/r/1423/ ........
................
2011-09-12 15:56 +0000 [r335435] Matthew Nicholson <mnicholson@digium.com>
* main/channel.c, /: Merged revisions 335434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r335434 | mnicholson | 2011-09-12 10:55:48 -0500
(Mon, 12 Sep 2011) | 13 lines Merged revisions 335433 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep
2011) | 6 lines Properly set caller_warning and callee_warning
before we try to use them. ASTERISK-18199 Patch by: elguero
Testing by: rtang ........ ................
2011-09-12 14:33 +0000 [r335385] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Documentation updates
2011-09-12 14:24 +0000 [r335354] Kinsey Moore <kmoore@digium.com>
* apps/app_dial.c, /: Merged revisions 335346 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r335346 | kmoore | 2011-09-12 09:22:15 -0500
(Mon, 12 Sep 2011) | 17 lines Merged revisions 335341 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) |
10 lines Ensure frames are not written to dialed channel if
ringback is requested When a single channel was dialed and there
was media to be forwarded to the calling channel, the media was
written without regard for ringback causing silence to be heard
in some circumstances. This regression was introduced when the
meaning of "single" changed to mean only the number of channels
dialed. (closes issue ASTERISK-18083) ........ ................
2011-09-12 14:22 +0000 [r335324-335349] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Small documentation updates
* CREDITS, channels/chan_sip.c, include/asterisk/indications.h,
UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h:
New sip.conf option for setting default tonezone for channel or
individual devices Review:
https://reviewboard.asterisk.org/r/1429/ (closes issue
ASTERISK-18497) Thanks to russellb for peer review.
* /, channels/chan_sip.c: Merged revisions 335323 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån,
12 Sep 2011) | 19 lines Merged revisions 335319 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12
lines Lock the peer->mvipvt to avoid crashes with SIP history
enabled After the launch of 1.6 event-based MWI we have two
threads handling the peer->mwipvt, which cause issues with SIP
history additions in combination with the max limit for number of
history entries. Review: https://reviewboard.asterisk.org/r/1373/
(closes issue ASTERISK-18288) Thanks to irrot for peer review.
Work with this bug funded by IPvision AS ........
................
2011-09-12 13:27 +0000 [r335322] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_iax2.c: Merged revisions 335321 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r335321 | kmoore | 2011-09-12 08:27:04 -0500
(Mon, 12 Sep 2011) | 16 lines Merged revisions 335320 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) |
9 lines Prevent IAX2 from getting IPv6 addresses via DNS IAX2
does not support IPv6 and getting such addresses from DNS can
cause error messages on the remote end involving bad IPv4 address
casts in the presence of IPv6/IPv4 tunnels. This patch ensures
that IAX2 will not encounter IPv6 addresses via DNS queries.
(closes issue ASTERISK-18090) ........ ................
2011-09-12 11:15 +0000 [r335261] Stefan Schmidt <sst@sil.at>
* /, channels/chan_sip.c: Merged revisions 335260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r335260 | schmidts | 2011-09-12 11:11:45 +0000
(Mon, 12 Sep 2011) | 12 lines Merged revisions 335259 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011)
| 6 lines build_peer doesnt unlink a peer object from peers_by_ip
container which leads to a wrong refcounter value. adding an
ao2_unlink from the peers_by_ip container fix it. Review:
https://reviewboard.asterisk.org/r/1428/ ........
................
2011-09-12 03:10 +0000 [r335170-335212] Paul Belanger <paul.belanger@polybeacon.com>
* UPGRADE.txt: Be more specific on which section has changed.
* main/cdr.c, UPGRADE.txt: Iterate though cdr.conf setting Review:
https://reviewboard.asterisk.org/r/1426/
2011-09-11 17:09 +0000 [r335129] Terry Wilson <twilson@digium.com>
* configs/res_config_sqlite3.conf.sample (added),
res/res_config_sqlite3.c (added): Add SQLite 3 realtime support
2011-09-09 16:28 +0000 [r335079] Matthew Jordan <mjordan@digium.com>
* channels/chan_unistim.c, apps/app_dial.c, main/pbx.c,
addons/chan_ooh323.c, channels/chan_sip.c,
channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c,
main/channel.c, channels/chan_usbradio.c, main/dial.c,
channels/chan_dahdi.c, channels/chan_misdn.c,
channels/chan_skinny.c, funcs/func_frame_trace.c,
main/features.c, channels/chan_h323.c, channels/chan_alsa.c,
include/asterisk/frame.h, channels/sig_ss7.c,
channels/chan_mgcp.c: Merged revisions 335078 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r335078 | mjordan | 2011-09-09 11:27:01 -0500
(Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011)
| 23 lines Updated SIP 484 handling; added Incomplete control
frame When a SIP phone uses the dial application and receives a
484 Address Incomplete response, if overlapped dialing is enabled
for SIP, then the 484 Address Incomplete is forwarded back to the
SIP phone and the HANGUPCAUSE channel variable is set to 28.
Previously, the Incomplete application dialplan logic was
automatically triggered; now, explicit dialplan usage of the
application is required. Additionally, this patch adds a new
AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel
driver receives this control frame, it is an indication that the
dialplan expects more digits back from the device. If the device
supports overlap dialing it should attempt to notify the device
that the dialplan is waiting for more digits; otherwise, it can
handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested
by: Matthew Jordan Review:
https://reviewboard.asterisk.org/r/1416/ ........
................
2011-09-09 07:28 +0000 [r335015] Gregory Nietsky <gregory@distrotech.co.za>
* funcs/func_dialplan.c, /, apps/app_readexten.c, CHANGES: Merged
revisions 335014 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r335014 | irroot | 2011-09-09 09:23:53 +0200 (Fri, 09 Sep 2011) |
9 lines Move code for VALID_EXTEN from app_readexten to
func_dialplan Mark VALID_EXTEN deprecated. Review:
https://reviewboard.asterisk.org/r/1396/ ........
2011-09-08 22:30 +0000 [r334955] Richard Mudgett <rmudgett@digium.com>
* /, main/logger.c: Merged revisions 334954 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r334954 | rmudgett | 2011-09-08 17:28:56 -0500
(Thu, 08 Sep 2011) | 17 lines Merged revisions 334953 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011)
| 10 lines Fix crash with res_fax when MALLOC_DEBUG and "core
stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is
enabled when res_fax tries to unregister its logger level. * Make
ast_logger_unregister_level() use ast_free() instead of free().
When MALLOC_DEBUG is enabled, ast_free() does not degenerate into
a call to free(). Therefore, if you allocated memory with a form
of ast_malloc you must free it with ast_free. ........
................
2011-09-08 13:36 +0000 [r334907] Jonathan Rose <jrose@digium.com>
* main/cdr.c, main/pbx.c: Removes colorful verb statements
erroneously commited with r332760
2011-09-07 19:38 +0000 [r334845] Paul Belanger <paul.belanger@polybeacon.com>
* /, channels/chan_iax2.c: Merged revisions 334844 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r334844 | pabelanger | 2011-09-07 15:37:24 -0400
(Wed, 07 Sep 2011) | 11 lines Merged revisions 334843 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep
2011) | 4 lines Cleanup chan_iax2.c log messages Review:
https://code.asterisk.org/code/cru/CR-AST-11 ........
................
2011-09-07 19:35 +0000 [r334842] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c: Merged revisions 334841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r334841 | rmudgett | 2011-09-07 14:33:38 -0500
(Wed, 07 Sep 2011) | 17 lines Merged revisions 334840 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011)
| 10 lines Fix AMI action Park crash. * Made AMI action Park not
say anything to the parker channel (AMI header Channel2) since
the AMI action is a third party parking the call. (This is a
change in behavior that cannot be preserved without a lot of
effort.) * Made not play pbx-parkingfailed if the Park 's' option
is used. JIRA AST-660 ........ ................
2011-09-07 15:37 +0000 [r334683-334792] Stefan Schmidt <sst@sil.at>
* /, main/features.c: Merged revisions 334747 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r334747 | schmidts | 2011-09-07 15:10:37 +0000
(Wed, 07 Sep 2011) | 9 lines Merged revisions 334682 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07
Sep 2011) | 3 lines Adding the Feature to sent a Reason Header in
a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE
before doing a masquerade in the pickup function. ........
................
* main/features.c: clean up wrong merged stuff
* /, main/features.c: Merged revisions 334682 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011)
| 3 lines Adding the Feature to sent a Reason Header in a SIP
Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before
doing a masquerade in the pickup function. ........
* main/features.c: Adding the Feature to sent a Reason Header in a
SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE
before doing a masquerade in the pickup function.
2011-09-07 08:17 +0000 [r334618-334623] Alec L Davis <sivad.a@paradise.net.nz>
* /, CHANGES, apps/app_queue.c: Merged revisions 334621 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r334621 | alecdavis | 2011-09-07 20:14:50 +1200
(Wed, 07 Sep 2011) | 9 lines Merged revisions 334620 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07
Sep 2011) | 2 lines peroid typo ........ ................
* main/logger.c: log Asterisk Version number, Build etc into each
log file Allow tracking of previous versions through log file
records to be tracked. Each time log file is created or opened,
log Asterisk Version, Buildinfo. etc. alecdavis (license 585)
Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1409/
* main/pbx.c, /: Merged revisions 334617 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r334617 | alecdavis | 2011-09-07 19:45:00 +1200
(Wed, 07 Sep 2011) | 17 lines Merged revisions 334616 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep
2011) | 10 lines Prevent segfault if call arrives before Asterisk
is fully booted. Prevent ast_pbx_start and ast_run_start from
starting a new thread unless asterisk is fully booted. alecdavis
(license 585) Tested by: alecdavis Review:
https://reviewboard.asterisk.org/r/1407/ ........
................
2011-09-07 00:54 +0000 [r334574] Tilghman Lesher <tilghman@meg.abyt.es>
* main/frame.c, contrib/realtime/mysql/iaxfriends.sql,
contrib/realtime/postgresql/realtime.sql,
configs/sip.conf.sample, CHANGES,
contrib/realtime/mysql/sipfriends.sql: Implement the '!' negation
element to negate codecs directly in the allow keyword. This
permits the list of codecs to be specified in one configuration
line, instead of two or more, generally with the aim of either
allowing all codecs with the exception of a few or disallowing
most but permitting a few. Review:
https://reviewboard.asterisk.org/r/1411/
2011-09-06 16:15 +0000 [r334519] Gregory Nietsky <gregory@distrotech.co.za>
* /, apps/app_voicemail.c: Merged revisions 334455 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r334455 | irroot | 2011-09-06 15:58:56 +0200
(Tue, 06 Sep 2011) | 18 lines Merged revisions 334453 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) |
13 lines Make SQL query in app_voicemail.c portable LIMIT is not
portable. Regression from r312212 (closes issue ASTERISK-18255)
Reported by: Leif Madsen Tested by: Leif Madsen Review:
https://reviewboard.asterisk.org/r/1415/ ........
................
2011-09-06 16:08 +0000 [r334517] Paul Belanger <paul.belanger@polybeacon.com>
* configs/iax.conf.sample, /, CHANGES, channels/chan_iax2.c: Merged
revisions 334514 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r334514 | pabelanger | 2011-09-06 11:47:59 -0400 (Tue, 06 Sep
2011) | 6 lines authdebug is now disabled by default To enable
this functionaility again set authdebug = yes in iax.conf Review:
https://reviewboard.asterisk.org/r/1414/ ........
2011-09-06 16:04 +0000 [r334472-334515] Gregory Nietsky <gregory@distrotech.co.za>
* /, apps/app_voicemail.c: Revert r334472 due to properties going
missing
* /, apps/app_voicemail.c: Merged revisions 334455 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r334455 | irroot | 2011-09-06 15:58:56 +0200
(Tue, 06 Sep 2011) | 18 lines Merged revisions 334453 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) |
13 lines Make SQL query in app_voicemail.c portable LIMIT is not
portable. Regression from r312212 (closes issue ASTERISK-18255)
Reported by: Leif Madsen Tested by: Leif Madsen Review:
https://reviewboard.asterisk.org/r/1415/ ........
................
2011-09-02 21:09 +0000 [r334304-334358] Richard Mudgett <rmudgett@digium.com>
* /, res/res_musiconhold.c: Merged revisions 334357 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r334357 | rmudgett | 2011-09-02 16:08:16 -0500
(Fri, 02 Sep 2011) | 26 lines Merged revisions 334355 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011)
| 19 lines MusicOnHold has extra unref which may lead to memory
corruption and crash. The problem happens when a call is
disconnected and you had started a MOH class that does not use
the files mode. If you define REF_DEBUG and recreate the problem,
it will announce itself with the following warning: Attempt to
unref mohclass 0xb70722e0 (default) when only 1 ref remained, and
class is still in a container! * Fixed moh_alloc() and
moh_release() functions not handling the state->class reference
consistently. (closes issue ASTERISK-18346) Reported by: Mark
Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621)
patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski
Review: https://reviewboard.asterisk.org/r/1404/ ........
................
* /, include/asterisk/config.h, main/config.c: Merged revisions
334297 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r334297 | rmudgett | 2011-09-02 12:15:08 -0500
(Fri, 02 Sep 2011) | 46 lines Merged revisions 334296 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011)
| 39 lines Fix potential memory allocation failure crashes in
config.c. * Added required checks to the returned memory
allocation pointers to prevent crashes. * Made
ast_include_rename() create a replacement ast_variable list node
if the new filename is longer than the available space. Fixes
potential crash and memory leak. * Factored out
ast_variable_move() from ast_variable_update() so
ast_include_rename() can also use it when creating a replacement
ast_variable list node. * Made the filename stuffed at the end of
the struct a minimum allocated size in ast_variable_new() in case
ast_include_rename() changes the stored filename. * Constify
struct char pointers pointing to strings stuffed at the end of
the struct for: ast_variable, cache_file_mtime, and
ast_config_map. * Factored out cfmtime_new() to remove inlined
code and allow some struct pointers to become const. * Removed
the list lock from struct cache_file_mtime that was never used. *
Added doxygen comments to several structure elements and better
documented what strings are stuffed at the struct end char array.
* Reworked ast_config_text_file_save() and set_fn() to handle
allocation failure of the include file scratch pad object
tracking blank lines. * Made ast_config_text_file_save() fn[]
declared with PATH_MAX to ensure it is long enough for any
filename with path. Also reduced the number of container fileset
buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review:
https://reviewboard.asterisk.org/r/1378/ ........
................
2011-09-01 17:41 +0000 [r334231-334236] Tilghman Lesher <tilghman@meg.abyt.es>
* main/pbx.c, /: Merged revisions 334235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r334235 | tilghman | 2011-09-01 12:39:32 -0500
(Thu, 01 Sep 2011) | 9 lines Merged revisions 334234 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r334234 | tilghman | 2011-09-01 12:38:33 -0500 (Thu, 01
Sep 2011) | 2 lines Remove 1.6 compatibility documentation from
1.8, as it no longer applies. ........ ................
* res/res_config_odbc.c, /: Merged revisions 334230 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r334230 | tilghman | 2011-09-01 12:30:19 -0500
(Thu, 01 Sep 2011) | 25 lines Merged revisions 334229 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011)
| 18 lines Create a local alias for ast_odbc_clear_cache. As a
function pointer, the reference has to be resolved at load time
irrespective of the RTLD_LAZY flag. Creating a local alias solves
this problem, because the structure is initialized with that
local function pointer, while the actual function can remain
lazily linked until runtime. The reason why this is important is
because we lazily load function references during the module
loading process, in order to obtain priority values for each
module, ensuring that modules are loaded in the correct order.
Previous to this change, when this module was initially loaded,
the module loader would emit a symbol resolution error, because
of the above requirement. Closes ASTERISK-18399 (reported by
Mikael Carlsson, fix suggested by Walter Doekes, patch by me)
........ ................
2011-08-31 18:54 +0000 [r334158] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c: Merged revisions 334157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r334157 | mnicholson | 2011-08-31 13:53:40 -0500
(Wed, 31 Aug 2011) | 11 lines Merged revisions 334156 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug
2011) | 4 lines Disable T.38 when we get a invite with image
media port set to 0 ASTERISK-17678 ........ ................
2011-08-31 18:11 +0000 [r334115] Richard Mudgett <rmudgett@digium.com>
* channels/chan_sip.c: Optimize chan_sip.c check_rtp_timeout()
function. * Make check_rtp_timeout() remember the values returned
by ast_rtp_instance_get_timeout(),
ast_rtp_instance_get_hold_timeout(), and
ast_rtp_instance_get_keepalive() instead of repeatedly calling
them. (closes issue ASTERISK-18319) Reported by: Rob Gagnon
Patches: issue-18319-trunk-r333066.diff (License #6159) patch
uploaded by Rob Gagnon Review:
https://reviewboard.asterisk.org/r/1377/
2011-08-31 16:31 +0000 [r334067] Matthew Nicholson <mnicholson@digium.com>
* /, res/res_fax.c: Merged revisions 334064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r334064 | mnicholson | 2011-08-31 11:31:00 -0500 (Wed, 31 Aug
2011) | 4 lines only alter the gateway_timeout when attching the
gateway to a channel ASTERISK-18219 ........
2011-08-31 16:02 +0000 [r334011-334014] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 334013 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r334013 | rmudgett | 2011-08-31 11:00:49 -0500
(Wed, 31 Aug 2011) | 30 lines Merged revisions 334012 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011)
| 23 lines No DAHDI channel available for conference, user
introduction disabled. The following error will consistently
occur when trying to dial into a MeetMe conference when the
server does not have DAHDI hardware installed: app_meetme.c: No
DAHDI channel available for conference, user introduction
disabled (is chan_dahdi loaded?) While chan_dahdi is loaded
correctly during compilation and install of Asterisk/Dahdi,
including associated modules, etc., a chan_dahdi.conf
configuration file in /etc/asterisk is not created by FreePBX if
hardware does not exist, causing MeetMe to be unable to open a
DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo
channel when there is no chan_dahdi.conf file to load. (closes
issue ASTERISK-17398) Reported by: Preston Edwards Patches:
jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by
rmudgett Tested by: rmudgett ........ ................
* main/channel.c, /, channels/chan_agent.c: Merged revisions 334010
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r334010 | rmudgett | 2011-08-31 10:23:11 -0500
(Wed, 31 Aug 2011) | 50 lines Merged revisions 334009 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011)
| 43 lines Call pickup race leaves orphaned channels or crashes.
Multiple users attempting to pickup a call that has been forked
to multiple extensions either crashes or fails a masquerade with
a "bad things may happen" message. This is the scenario that is
causing all the grief: 1) Pickup target is selected 2) target is
marked as being picked up in ast_do_pickup() 3) target is
unlocked by ast_do_pickup() 4) app dial or queue gets a chance to
hang up losing calls and calls ast_hangup() on target 5) SINCE A
MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with
ast_channel_masquerade(), ast_hangup() completes successfully and
the channel is no longer in the channels container. 6)
ast_do_pickup() then calls ast_channel_masquerade() to schedule
the masquerade on the dead channel. 7) ast_do_pickup() then calls
ast_do_masquerade() on the dead channel 8) bad things happen
while doing the masquerade and in the process ast_do_masquerade()
puts the dead channel back into the channels container 9) The
"orphaned" channel is visible in the channels list if a crash
does not happen. This patch does the following: * Made
ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up
channel and not release the channel lock until that has happened.
* Made __ast_channel_masquerade() not setup a masquerade if
either channel has AST_FLAG_ZOMBIE set. * Fix chan_agent misuse
of AST_FLAG_ZOMBIE since it would no longer work. (closes issue
ASTERISK-18222) Reported by: Alec Davis Tested by: rmudgett, Alec
Davis, irroot, Karsten Wemheuer (closes issue ASTERISK-18273)
Reported by: Karsten Wemheuer Tested by: rmudgett, Alec Davis,
irroot, Karsten Wemheuer Review:
https://reviewboard.asterisk.org/r/1400/ ........
................
2011-08-31 15:20 +0000 [r334008] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Merged revisions 334007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r334007 | kmoore | 2011-08-31 10:19:30 -0500
(Wed, 31 Aug 2011) | 14 lines Merged revisions 334006 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) |
7 lines Correct an AMI protocol violation with SIPshowpeer The
response of SIPshowpeer ends with "\r\n\r\n". Since other
commands are ended by using \r\n this confuses any interfacing
script. (closes issue ASTERISK-17486) ........ ................
2011-08-30 22:16 +0000 [r333963] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, /,
addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/oochannels.c,
addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: Merged
revisions 333961-333962 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r333961 | may | 2011-08-31 01:21:53 +0400 (Wed,
31 Aug 2011) | 11 lines Merged revisions 333947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5
lines cleanups in ACF/ARJ GK replies processing fixed long (24
sec) pause if acf/arj proccessed before ast_cond_wait called to
wait this ........ ................ r333962 | may | 2011-08-31
01:53:42 +0400 (Wed, 31 Aug 2011) | 3 lines security fix. really
drop call if signalling addr is not same as socket addr
................
2011-08-30 14:03 +0000 [r333896] Matthew Nicholson <mnicholson@digium.com>
* /, res/res_fax.c: Merged revisions 333895 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r333895 | mnicholson | 2011-08-30 09:01:31 -0500 (Tue, 30 Aug
2011) | 6 lines Replaced FAXOPT(gwtimeout) with a second
parameter to FAXOPT(gateway). Patch by: irroot Review:
https://reviewboard.asterisk.org/r/1385/ ASTERISK-18219 ........
2011-08-29 21:43 +0000 [r333838] Terry Wilson <twilson@digium.com>
* /, channels/chan_sip.c: Merged revisions 333837 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r333837 | twilson | 2011-08-29 16:41:13 -0500
(Mon, 29 Aug 2011) | 22 lines Merged revisions 333836 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011)
| 15 lines Refresh peer address if DNS unavailable at peer
creation If Asterisk starts and no DNS is available, outbound
registrations will fail indefinitely. This patch copies the
address from the sip_registry struct, which will be updated, to
the peer->addr when necessary. If dnsmgr is enabled, the
registration fails without the patch because even though the
address on the registry is updated via dnsmgr, the address is
just copied on the first try. Since we use ast_sockaddr_copy,
dnsmgr can't update the address that is copied to the sip_pvt or
peers. Closes issue ASTERISK-18000 Review:
https://reviewboard.asterisk.org/r/1335/ ........
................
2011-08-29 21:17 +0000 [r333789] Richard Mudgett <rmudgett@digium.com>
* /, include/asterisk/channel.h, addons/chan_mobile.c: Merged
revisions 333786 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r333786 | rmudgett | 2011-08-29 16:12:29 -0500
(Mon, 29 Aug 2011) | 13 lines Merged revisions 333784-333785 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011)
| 2 lines Fix deadlock potential of
chan_mobile.c:mbl_ast_hangup(). ........ r333785 | rmudgett |
2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line Add some do
not hold locks notes to channel.h ........ ................
2011-08-29 18:28 +0000 [r333736] Matthew Nicholson <mnicholson@digium.com>
* /, res/res_fax_spandsp.c: Merged revisions 333716 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r333716 | mnicholson | 2011-08-29 13:22:58 -0500 (Mon, 29 Aug
2011) | 5 lines It is possible for the gateway to be attached
when the channel is still negotiating T.38. This change handles
that case. ASTERISK-18329 ........
2011-08-29 17:31 +0000 [r333689] Terry Wilson <twilson@digium.com>
* main/channel.c, /, CHANGES: Merged revisions 333681 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r333681 | twilson | 2011-08-29 12:28:59 -0500 (Mon, 29 Aug 2011)
| 7 lines Use realtime text when it is negotiated This patch make
use of wirte_text() realtime text instead of send_text() if T.140
is in native formats. ASTERISK-17937 Review:
https://reviewboard.asterisk.org/r/1356/ ........
2011-08-29 17:14 +0000 [r333632] Matthew Jordan <mjordan@digium.com>
* apps/app_voicemail.c: Merged revisions 333631 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r333631 | mjordan | 2011-08-29 12:12:55 -0500
(Mon, 29 Aug 2011) | 9 lines Merged revisions 333630 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r333630 | mjordan | 2011-08-29 12:11:15 -0500 (Mon, 29
Aug 2011) | 1 line Fixed improperly formatted TestEvent AMI
message in app_voicemail ........ ................
2011-08-29 15:58 +0000 [r333571] Jonathan Rose <jrose@digium.com>
* /, res/res_jabber.c: Merged revisions 333570 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r333570 | jrose | 2011-08-29 10:56:56 -0500
(Mon, 29 Aug 2011) | 11 lines Merged revisions 333569 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r333569 | jrose | 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) |
4 lines Accidental use of variable client->status instead of
client->state in from ASTERISK-18078 (issue ASTERISK-18078)
........ ................
2011-08-28 09:57 +0000 [r333509] Tzafrir Cohen <tzafrir.cohen@xorcom.com>
* channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6)
GCC 4.6 detects variables that get assined to, but never used
later. Also removes some remmed-out lines that become invalid.
(closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen
(License #5035) <tzafrir.cohen@xorcom.com>,
2011-08-26 16:38 +0000 [r333428] Jonathan Rose <jrose@digium.com>
* /, res/res_jabber.c: Merged revisions 333410 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r333410 | jrose | 2011-08-26 11:28:03 -0500
(Fri, 26 Aug 2011) | 19 lines Merged revisions 333378 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) |
13 lines [patch] Buddies are always auto-registered when
processing the roster Reporter said autoregister flag was ignored
for registering 'buddies' which had a subscription to us.
Verified that this was the case and observed how the patch
addressed this and made sure it didn't break anything. (closes
issue ASTERISK-14233) Reported by: Simon Arlott Patches:
asterisk-0015229.patch (license #5756) patch uploaded by Simon
Arlott Tested by: Jonathan Rose ........ ................
2011-08-26 16:12 +0000 [r333371] Matthew Jordan <mjordan@digium.com>
* /, apps/app_voicemail.c: Merged revisions 333370 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r333370 | mjordan | 2011-08-26 10:58:37 -0500
(Fri, 26 Aug 2011) | 26 lines Merged revisions 333339 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r333339 | mjordan | 2011-08-26 08:36:36 -0500 (Fri, 26 Aug 2011)
| 20 lines Bug fixes for voicemail user emailsubject / emailbody.
This code change fixes a few issues with the voicemail user
override of emailbody and emailsubject, including escaping the
strings, potential memory leaks, and not overriding the voicemail
defaults. Revision 325877 fixed this for ASTERISK-16795, but did
not fix it for ASTERISK-16781. A subsequent check-in prevented
325877 from being applied to 10. This check-in resolves both
issues, and applies the changes to 1.8, 10, and trunk. (closes
issue ASTERISK-16781) Reported by: Sebastien Couture Tested by:
mjordan (closes issue ASTERISK-16795) Reported by: mdeneen Tested
by: mjordan Review: https://reviewboard.asterisk.org/r/1374
........ ................
2011-08-25 19:13 +0000 [r333276] Jonathan Rose <jrose@digium.com>
* /, res/res_jabber.c: Merged revisions 333266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r333266 | jrose | 2011-08-25 14:00:05 -0500
(Thu, 25 Aug 2011) | 20 lines Merged revisions 333265 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) |
14 lines Segfault when publishing device states via XMPP and not
connected When using publishing device state with res_jabber,
Asterisk will attempt to send a device state using the
unconnected client using iks_send_raw and crash. This patch
checks the validity of the connection before attempting to send
the device state. (closes issue ASTERISK-18078) Reported by:
Michael L. Young Patches:
res_jabber-segfault-pubsub-not-connected2.patch (license #5026)
patch uploaded by Michael L. Young Tested by: Jonathan Rose
........ ................
2011-08-25 19:01 +0000 [r333159-333269] Jason Parker <jparker@digium.com>
* Makefile, /: Merged revisions 333268 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r333268 | qwell | 2011-08-25 14:01:18 -0500
(Thu, 25 Aug 2011) | 9 lines Merged revisions 333267 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r333267 | qwell | 2011-08-25 14:00:55 -0500 (Thu, 25 Aug
2011) | 2 lines Fix for DESTDIR spaces patch. ........
................
* Makefile, build_tools/mkpkgconfig, /, configure, configure.ac,
makeopts.in, sounds/Makefile: Merged revisions 333203 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r333203 | qwell | 2011-08-25 10:29:56 -0500
(Thu, 25 Aug 2011) | 15 lines Merged revisions 333201 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) |
8 lines Fix installation into directories containing spaces. This
also vastly simplifies the logic in sounds/Makefile (Closes issue
ASTERISK-18290) Reported by: Paul Belanger Review:
https://reviewboard.asterisk.org/r/1379/ ........
................
* channels/chan_local.c: Fix typo from r333070
2011-08-24 16:52 +0000 [r333117] Matthew Nicholson <mnicholson@digium.com>
* /, res/res_fax.c: Merged revisions 333115 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r333115 | mnicholson | 2011-08-24 11:51:42 -0500 (Wed, 24 Aug
2011) | 4 lines Changed the "timeout" option to "gwtimeout".
ASTERISK-18219 ........
2011-08-24 09:17 +0000 [r333070-333075] Olle Johansson <oej@edvina.net>
* channels/chan_local.c: Formatting changes - Removing some red
white space and adding some curly brackets.
* CHANGES: Add documentation for new manager event in chan_local
AST-17623
* channels/chan_local.c: Add manager event for local channel
semi-bridge (issue AST-17623) Review:
https://reviewboard.asterisk.org/r/1154
2011-08-23 18:17 +0000 [r332881-333014] Richard Mudgett <rmudgett@digium.com>
* /, apps/app_queue.c: Merged revisions 333011 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r333011 | rmudgett | 2011-08-23 13:15:49 -0500
(Tue, 23 Aug 2011) | 19 lines Merged revisions 333010 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011)
| 12 lines Memory Leak in app_queue The patch that was committed
in the 1.6.x versions of Asterisk for ASTERISK-15862 actually
fixed two issues. One was not applicable to 1.8 but the other is.
queue_leak.patch fixes the portion applicable to 1.8. (closes
issue ASTERISK-18265) Reported by: Fred Schroeder Patches:
queue_leak.patch (license #5049) patch uploaded by mmichelson
Tested by: Thomas Arimont ........ ................
* /, main/config.c: Merged revisions 332940 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332940 | rmudgett | 2011-08-22 16:23:40 -0500
(Mon, 22 Aug 2011) | 14 lines Merged revisions 332939 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332939 | rmudgett | 2011-08-22 16:22:24 -0500 (Mon, 22 Aug 2011)
| 7 lines Minor code optimizations. * Simplify
ast_category_browse() logic for easier understanding. * Remove
dead code in ast_variable_delete() and simplify some of its
logic. ........ ................
* /, apps/app_queue.c: Merged revisions 332875,332878 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r332875 | rmudgett | 2011-08-22 14:41:03 -0500
(Mon, 22 Aug 2011) | 1 line Fix merge property. ................
r332878 | rmudgett | 2011-08-22 14:46:25 -0500 (Mon, 22 Aug 2011)
| 25 lines Merged revisions 332874 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011)
| 18 lines Reference leaks in app_queue. * Fixed
load_realtime_queue() leaking a queue reference when it
overwrites q when processing a realtime queue. (issue
ASTERISK-18265) * Make join_queue() unreference the queue
returned by load_realtime_queue() when it is done with the
pointer. The load_realtime_queue() returns a reference to the
just loaded realtime queue. * Fixed queues container reference
leak in queues_data_provider_get(). * queue_unref() should not
return q that was just unreferenced. * Made logic in
__queues_show() and queues_data_provider_get() when calling
load_realtime_queue() easier to understand. ........
................
2011-08-22 19:56 +0000 [r332880] Paul Belanger <paul.belanger@polybeacon.com>
* /, channels/chan_gtalk.c: Merged revisions 332877 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r332877 | pabelanger | 2011-08-22 15:43:33 -0400
(Mon, 22 Aug 2011) | 13 lines Merged revisions 332876 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332876 | pabelanger | 2011-08-22 15:41:24 -0400 (Mon, 22 Aug
2011) | 6 lines Revert previous commit It seems google is still
making changes to the protocol. (issue ASTERISK-18301) ........
................
2011-08-22 19:52 +0000 [r332879] Richard Mudgett <rmudgett@digium.com>
* /: Fix merge 10 branch merge properties.
2011-08-22 19:19 +0000 [r332844] Matthew Jordan <mjordan@digium.com>
* include/asterisk/test.h, main/manager.c, /, main/file.c,
main/test.c, main/app.c, configs/manager.conf.sample,
include/asterisk/manager.h, apps/app_voicemail.c: Merged
revisions 332817 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011)
| 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This
update adds a new AMI event, TestEvent, which is enabled when the
TEST_FRAMEWORK compiler flag is defined. It also adds initial
usage of this event to app_voicemail. The TestEvent AMI event is
used extensively by the voicemail tests in the Asterisk Test
Suite. ........
2011-08-22 18:33 +0000 [r332762-332831] Richard Mudgett <rmudgett@digium.com>
* res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged
revisions 332830 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332830 | rmudgett | 2011-08-22 13:32:09 -0500
(Mon, 22 Aug 2011) | 15 lines Merged revisions 332816 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011)
| 8 lines Memory leaks in realtime_multi_xxx() when database
access returns error. * Fix realtime_multi_pgsql() configuration
memory leak when the database access returns an error. * Fix
realtime_multi_odbc() configuration category use after free when
the database access returns an error. ........ ................
* /, main/config.c: Merged revisions 332761 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332761 | rmudgett | 2011-08-22 12:05:35 -0500
(Mon, 22 Aug 2011) | 22 lines Merged revisions 332759 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011)
| 15 lines Memory leak reading realtime database variable list.
Calling ast_load_realtime() can leak the last list node if the
read list only contains empty variable value items. * Fixed list
filter loop in ast_load_realtime() to delete the list node
immediately instead of the next time through the loop. The next
time through the loop may not happen if the node to delete is the
last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265)
Patches: jira_asterisk_18265_v1.8_config.patch (license #5621)
patch uploaded by rmudgett ........ ................
2011-08-22 17:05 +0000 [r332760] Jonathan Rose <jrose@digium.com>
* main/cdr.c, main/pbx.c, configs/cdr.conf.sample,
include/asterisk/cdr.h, CHANGES: Add option for logging congested
calls as CONGESTION instead of NO_ANSWER in CDR This patch adds a
CDR option to cdr.conf that will allow CDR files to log calls
ending with congestion in a way that is unique from other
unanswered calls. (closes issue ASTERISK-14842) Reported by: Alec
Davis Patches: cdr_congestion.diff.txt (License #5546) patch
uploaded by Alec Davis
2011-08-22 16:31 +0000 [r332757] Matthew Nicholson <mnicholson@digium.com>
* /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions
332756 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r332756 | mnicholson | 2011-08-22 11:29:45 -0500 (Mon, 22 Aug
2011) | 4 lines add a way to disable and/or modify the gateway
timeout ASTERISK-18219 ........
2011-08-21 14:34 +0000 [r332701] Paul Belanger <paul.belanger@polybeacon.com>
* /, channels/chan_gtalk.c: Merged revisions 332700 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r332700 | pabelanger | 2011-08-21 10:33:23 -0400
(Sun, 21 Aug 2011) | 12 lines Merged revisions 332699 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332699 | pabelanger | 2011-08-21 10:31:31 -0400 (Sun, 21 Aug
2011) | 5 lines Fix outgoing calls in chan_gtalk (closes issue
ASTERISK-18301) Reported by: az1324 ........ ................
2011-08-19 20:00 +0000 [r332655] Kinsey Moore <kmoore@digium.com>
* /, apps/app_confbridge.c: Merged revisions 332654 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r332654 | kmoore | 2011-08-19 14:59:34 -0500 (Fri, 19 Aug 2011) |
8 lines Make CONFBRIDGE_INFO behave more nicely CONFBRIDGE_INFO
doesn't behave as well in edge cases as MEETME_INFO. With this
patch, CONFBRIDGE_INFO should behave in a much more reasonable
manner when presented with invalid conferences and keywords.
Review: https://reviewboard.asterisk.org/r/1359/ ........
2011-08-19 17:24 +0000 [r332615] Richard Mudgett <rmudgett@digium.com>
* res/res_config_ldap.c: Fix infinite loop releasing the same
memory in ldap_loadentry(). * Fixed memory leak of vars in
ldap_loadentry(). * Fixed potential NULL ptr dereference of vars
in ldap_loadentry().
2011-08-18 21:39 +0000 [r332561] Terry Wilson <twilson@digium.com>
* main/netsock2.c, /: Merged revisions 332560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332560 | twilson | 2011-08-18 16:34:04 -0500
(Thu, 18 Aug 2011) | 12 lines Merged revisions 332559 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011)
| 5 lines Fix possible error on stringification of IPv4-mapped
addrs The FreeBSD netsock2 test has been failing for a while. We
were pasing sa->len to getnameinfo instead of sa_tmp->len.
ASTERISK-18289 ........ ................
2011-08-18 19:30 +0000 [r332505] Kinsey Moore <kmoore@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 332504 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r332504 | kmoore | 2011-08-18 14:29:15 -0500
(Thu, 18 Aug 2011) | 15 lines Merged revisions 332503 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) |
8 lines CRC4 in "dahdi show status" gives wrong impression to T1
users Change CRC4 to CRC in the output of "dahdi show status" so
that it can apply in more situations without confusing users,
especially since T1 lines use CRC6 instead of CRC4. (closes issue
AST-471) ........ ................
2011-08-18 14:49 +0000 [r332388-332448] Tilghman Lesher <tilghman@meg.abyt.es>
* build_tools/cflags.xml, build_tools/cflags-devmode.xml, /: Merged
revisions 332447 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332447 | tilghman | 2011-08-18 09:48:40 -0500
(Thu, 18 Aug 2011) | 9 lines Merged revisions 332446 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r332446 | tilghman | 2011-08-18 09:46:54 -0500 (Thu, 18
Aug 2011) | 2 lines Move BETTER_BACKTRACES out of development
mode, as it's useful when DEBUG_THREADS is enabled. ........
................
* Makefile, agi/Makefile, utils/Makefile, /, configure,
include/asterisk/autoconfig.h.in, configure.ac,
Makefile.moddir_rules, makeopts.in, sounds/Makefile: Merged
revisions 332369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332369 | tilghman | 2011-08-17 14:24:59 -0500
(Wed, 17 Aug 2011) | 17 lines Merged revisions 332355 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011)
| 10 lines Re-add support for spaces in pathnames, including now
spaces in DESTDIR. This was initially added to 1.8 prior to
release, primarily to support the standard paths on Mac OS X, but
was partially reverted recently in Subversion, due to the lack of
support for spaces in DESTDIR. This commit restores support for
the standard paths on Mac OS X, and also includes support for
spaces in DESTDIR. (closes issue ASTERISK-18290) Reported by:
pabelanger Review: https://reviewboard.asterisk.org/r/1326/
........ ................
2011-08-17 18:31 +0000 [r332337] Terry Wilson <twilson@digium.com>
* /, res/res_timing_timerfd.c: Merged revisions 332321 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r332321 | twilson | 2011-08-17 13:09:49 -0500
(Wed, 17 Aug 2011) | 17 lines Merged revisions 332320 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17 Aug 2011)
| 10 lines Don't read from a disarmed or invalid timerfd Numerous
isues have been reported for deadlocks that are caused by a
blocking read in res_timing_timerfd on a file descriptor that
will never be written to. This patch adds some checks to make
sure that the timerfd is both valid and armed before calling
read(). Should fix: ASTERISK-18142, ASTERISK-18166,
ASTERISK-18197, AST-486, AST-495, AST-507 and possibly others.
Review: https://reviewboard.asterisk.org/r/1361/ ........
................
2011-08-17 16:18 +0000 [r332270] Richard Mudgett <rmudgett@digium.com>
* channels/sig_pri.h, channels/chan_dahdi.c,
configs/chan_dahdi.conf.sample, /, configure,
include/asterisk/autoconfig.h.in, configure.ac,
channels/sig_pri.c: Merged revisions 332265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332265 | rmudgett | 2011-08-17 11:01:29 -0500
(Wed, 17 Aug 2011) | 33 lines Merged revisions 332264 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011)
| 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with
HA8, HB8, and B410P cards. France Telecom brings layer 2 and
layer 1 down on BRI lines when the line is idle. When layer 1
goes down Asterisk cannot make outgoing calls and the HA8 and HB8
cards also get IRQ misses. The inability to make outgoing calls
is because the line is in red alarm and Asterisk will not make
calls over a line it considers unavailable. The IRQ misses for
the HA8 and HB8 card are because the hardware is switching clock
sources from the line which just brought layer 1 down to internal
timing. There is a DAHDI option for the B410P card to not tell
Asterisk that layer 1 went down so Asterisk will allow outgoing
calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI
option for the HA8 and HB8 cards: "modprobe wctdm24xxp
bri_teignored=1". Unfortunately that will not clear up the IRQ
misses when the telco brings layer 1 down. * Add layer 2
persistence option to customize the layer 2 behavior on BRI PTMP
lines. The new option has three settings: 1) Use libpri default
layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when
the peer brings it down. 3) Leave layer 2 down when the peer
brings it down. Layer 2 will be brought up as needed for outgoing
calls. JIRA AST-598 ........ ................
2011-08-16 20:15 +0000 [r332178] Paul Belanger <paul.belanger@polybeacon.com>
* tests/test_substitution.c, tests/test_heap.c, /,
tests/test_expr.c, tests/test_ast_format_str_reduce.c,
tests/test_logger.c, tests/test_gosub.c, tests/test_app.c,
tests/test_dlinklists.c, tests/test_event.c, tests/test_db.c,
tests/test_linkedlists.c, tests/test_sched.c,
tests/test_netsock2.c, tests/test_strings.c, tests/test_pbx.c,
tests/test_func_file.c, tests/test_security_events.c,
tests/test_stringfields.c, tests/test_time.c, tests/test_skel.c,
tests/test_acl.c, tests/test_locale.c, tests/test_utils.c,
tests/test_devicestate.c, tests/test_aoc.c, tests/test_astobj2.c,
tests/test_poll.c, tests/test_amihooks.c: Merged revisions 332177
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332177 | pabelanger | 2011-08-16 16:11:49 -0400
(Tue, 16 Aug 2011) | 11 lines Merged revisions 332176 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332176 | pabelanger | 2011-08-16 16:10:13 -0400 (Tue, 16 Aug
2011) | 4 lines Flag test modules as 'core' Review:
https://reviewboard.asterisk.org/r/1369/ ........
................
2011-08-16 17:53 +0000 [r332120] Jonathan Rose <jrose@digium.com>
* /, channels/chan_sip.c: Merged revisions 332119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332119 | jrose | 2011-08-16 12:45:38 -0500
(Tue, 16 Aug 2011) | 23 lines Merged revisions 332118 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) |
16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects
mailbox value, multiple MWI subs Before, having multiple
subscriptions to mailboxes on a sip peer set via the mailbox
setting in sip.conf would only result in updates being sent on
whichever mailbox triggered the mwi event. Now all of them get
counted regardless. Also fixes a bug involving parsing of the
mailbox option in sip.conf so that trailing and leading spaces
before/after commas are trimmed. (closes issue ASTERISK-18067)
Reported by: aragon (closes issue ASTERISK-15479) Reported by:
Ben Winslow Patches:
chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288)
patch uploaded by Ben Winslow ........ ................
2011-08-16 17:23 +0000 [r332117] Richard Mudgett <rmudgett@digium.com>
* /, main/features.c, CHANGES, configs/features.conf.sample,
main/asterisk.c: Merged revisions 332101 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332101 | rmudgett | 2011-08-16 12:17:28 -0500
(Tue, 16 Aug 2011) | 140 lines Merged revisions 332100 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011)
| 133 lines Fix multiple parking issues. JIRA ASTERISK-17183
Multi-parkinglot directs calls to wrong parkinglot. JIRA
ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430
ParkedCall() with no extension should pickup first available call
and does not. JIRA AST-576 Issues with parking lots * Removed
searching for parking lots by extension. Parking lots can only be
found by the parking lot name since parking lot access extensions
and spaces are not guaranteed to be unique. * Added
parking_lot_name option to the Park and ParkedCall applications.
Updated documentation for Park and ParkedCall applications. * Add
parkext_exclusive configuration option to make parking entry
extensions specify which parking lot they access. (closes issue
ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett,
David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi
Quezada (closes issue ASTERISK-17430) Reported by: Philippe
Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA
AST-624 'next' setting for findslot does nothing * Reimplemented
since findslot feature option broken by -r114655. (closes issue
ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett
JIRA ASTERISK-15792 Dialplan continues execution after transfer
to park. This happens for DTMF attended transfer, DTMF blind
transfer, and DTMF one-touch-parking if the party initiating
these features also initiated the call. * Fixed the return code
from the affected builtin features when parking a call. (closes
issue ASTERISK-15792) Reported by: Mat Murdock Tested by:
rmudgett, twilson JIRA AST-607 The courtesytone is not playing to
the expected call when picking up a parked call. This is mostly a
documentation problem. However, the option is not reset to the
default when features.conf is reloaded. * Updated
features.conf.sample documentation for courtesytone and
parkedplay options. * Reset the parkedplay option to default when
features.conf is reloaded. JIRA AST-615 AMI Park action followed
by features reload results in orphaned channels in parking lot. *
Reloading features.conf will not touch parking lots that have
calls still parked in them. Reload again at a later time. Misc
additional fixes: * Added unit test for parking lot dialplan
usage checking. * Made update connected line when a parked call
is retrieved from a parking lot. * Made retrieved parked call
stop ringing or MOH depending upon how the call was waiting in
the parking lot. * Made CLI "features show" indicate if the
parking lot is enabled for use. * Added PARKINGDYNEXTEN channel
variable to allow dynamic parking lots to specify the parking lot
access extension. * Made AMI ParkedCalls action ParkedCall events
have a Parkinglot header. * Made AMI ParkedCalls action
ParkedCallsComplete event have a Total header. * Fixed potential
deadlock from AMI Park action holding channel locks while calling
masq_park_call(). * Fixed several places where ast_strdupa() were
used inside of loops. (Mostly fixed by refactoring the loop body
into its own function.) * Fixed copy_parkinglot() copying too
much from the source parking lot. Extracted the parking lot
configuration settings into struct parkinglot_cfg. * Refactored
courtesytone playing code to put the channel not playing the tone
in autoservice. * Fix when pbx-parkingfailed is played that the
other channel is put in autoservice if it exists. * Fixed
parkinglot reference leak in parked_call_exec() error paths. *
Fixed parkinglot_unref() use of parkinglot after it was unreffed.
* Made destroy the struct ast_parkinglot parkings lock when done.
* Refactored the features.conf parking lot configuration code to
eliminate redundancy. * Fixed feature reload to better protect
parking lots. * Fixed parking lot container reference leak in
handle_parkedcalls(). * Fixed the total count in
handle_parkedcalls(). Review:
https://reviewboard.asterisk.org/r/1358/ ........
................
2011-08-16 15:21 +0000 [r332028-332044] Matthew Nicholson <mnicholson@digium.com>
* /, channels/sip/include/sip.h: Merged revisions 332042 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
........ r332042 | mnicholson | 2011-08-16 10:20:48 -0500 (Tue,
16 Aug 2011) | 2 lines fix a code comment AST-580 ........
* /, UPGRADE.txt, CHANGES: Merged revisions 332029 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r332029 | mnicholson | 2011-08-16 10:17:16 -0500 (Tue, 16 Aug
2011) | 2 lines Moved notes about 'storesipcause' to UPGRADE.txt
from CHANGES AST-580 ........
* /, channels/chan_sip.c, channels/sip/include/sip.h: Merged
revisions 332027 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332027 | mnicholson | 2011-08-16 10:08:40 -0500
(Tue, 16 Aug 2011) | 9 lines Merged revisions 332026 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue,
16 Aug 2011) | 2 lines use DEFAULT_STORE_SIP_CAUSE to set the
default value for the 'storesipcause' option AST-580 ........
................
2011-08-16 14:47 +0000 [r332024] Olle Johansson <oej@edvina.net>
* channels/chan_local.c: Formatting changes while working with
DTMF...
2011-08-16 14:41 +0000 [r332023] Matthew Nicholson <mnicholson@digium.com>
* /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged
revisions 332022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r332022 | mnicholson | 2011-08-16 09:40:37 -0500
(Tue, 16 Aug 2011) | 16 lines In 10 and trunk this option is
disabled by default. Merged revisions 332021 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug
2011) | 7 lines Added the 'storesipcause' option to sip.conf to
allow the user to disable the setting of HASH(SIP_CAUSE,<chan
name>) on the channel. Having chan_sip set HASH(SIP_CAUSE,<chan
name>) on the channel carries a significant performance penalty
because of the usage of the MASTER_CHANNEL() dialplan function.
AST-580 ........ ................
2011-08-15 17:36 +0000 [r331957] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 331956 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r331956 | rmudgett | 2011-08-15 12:35:03 -0500
(Mon, 15 Aug 2011) | 20 lines Merged revisions 331955 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011)
| 13 lines Fix some minor chan_dahdi config load issues. *
Address chan_dahdi.conf dahdichan option todo item about needing
line number. * Make ignore_failed_channels option also apply to
dahdichan option. * Don't attempt to create a default pseudo
channel if the chan_dahdi.conf channel/channels option is not
allowed. * Add a similar check for dahdichan in normal
chan_dahdi.conf sections as is done in users.conf. ........
................
2011-08-15 15:24 +0000 [r331903] Paul Belanger <paul.belanger@polybeacon.com>
* main/rtp_engine.c, /: Merged revisions 331894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331894 | pabelanger | 2011-08-15 11:22:45 -0400
(Mon, 15 Aug 2011) | 12 lines Merged revisions 331886 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331886 | pabelanger | 2011-08-15 11:21:16 -0400 (Mon, 15 Aug
2011) | 5 lines Fix noisy message when briding channels (closes
issue ASTERISK-18270) Reported by: Federico Alves ........
................
2011-08-15 15:15 +0000 [r331869] David Vossel <dvossel@digium.com>
* /, channels/chan_sip.c: Merged revisions 331868 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331868 | dvossel | 2011-08-15 10:14:13 -0500
(Mon, 15 Aug 2011) | 12 lines Merged revisions 331867 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011)
| 6 lines Fixes locking inversion issues present in the handling
of the sip REFER method. (closes issue ASTERISK-18082) Reported
by: James Van Vleet ........ ................
2011-08-15 13:27 +0000 [r331830] Olle Johansson <oej@edvina.net>
* channels/chan_sip.c: Formatting guideline fixes
2011-08-12 19:06 +0000 [r331776] Matthew Nicholson <mnicholson@digium.com>
* /, apps/app_queue.c: Merged revisions 331775 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331775 | mnicholson | 2011-08-12 14:03:31 -0500
(Fri, 12 Aug 2011) | 17 lines Merged revisions 331774 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331774 | mnicholson | 2011-08-12 14:01:27 -0500 (Fri, 12 Aug
2011) | 11 lines Unlock the channel before calling update_queue.
Holding the channel lock when calling update_queue which attempts
to lock the queue lock can cause a deadlock. This deadlock
involves the following chain: 1. hold chan lock -> wait queue
lock 2. hold queue lock -> wait agent list lock 3. hold agent
list lock -> wait chan list lock 4. hold chan list lock -> wait
chan lock ........ ................
2011-08-12 19:01 +0000 [r331773] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 331772 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r331772 | rmudgett | 2011-08-12 13:59:45 -0500
(Fri, 12 Aug 2011) | 15 lines Merged revisions 331771 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011)
| 8 lines Suppress warning message when using DAHDITransfer or
DAHDIHangup. * The fake event should only be processed by the
channel that currently owns the private and not the associated
call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616
........ ................
2011-08-12 18:03 +0000 [r331717] Jonathan Rose <jrose@digium.com>
* apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331644
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331644 | jrose | 2011-08-12 11:18:57 -0500
(Fri, 12 Aug 2011) | 9 lines Merged revisions 331635 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug
2011) | 1 line Fixes 32bit compilation warnings brought on by
331634 in app_dial and app_meetme ........ ................
2011-08-12 17:56 +0000 [r331716] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, /: Merged revisions 331715 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r331715 | rmudgett | 2011-08-12 12:54:47 -0500
(Fri, 12 Aug 2011) | 29 lines Merged revisions 331714 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011)
| 22 lines AMI actions DAHDIHangup and DAHDITransfer have no
effect. The AMI actions DAHDIHangup and DAHDITransfer have no
effect on a DAHDI channel. These two AMI actions are highly
specialized to analog channels and appear to make the channel
behave like a jack port for headsets. * Made the faked DAHDI
event get processed before a normal media stream read in
dahdi_read() instead of trying to trigger an exception read by
setting the AST_FLAG_EXCEPTION flag. Apparently a change was made
long ago that changed how AST_FLAG_EXCEPTION is processed in the
core. Unfortunately, the faked DAHDI events no longer worked when
that happened. * Updated the DAHDI AMI action documentation for
the following actions: DAHDITransfer, DAHDIHangup,
DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and
DAHDIRestart. * Made use sscanf() instead of atoi() for better
error checking of the DAHDIChannel header string. JIRA AST-620
JIRA SWP-3616 ........ ................
2011-08-12 16:32 +0000 [r331660] Terry Wilson <twilson@digium.com>
* /, tests/test_netsock2.c: Merged revisions 331659 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r331659 | twilson | 2011-08-12 11:31:21 -0500
(Fri, 12 Aug 2011) | 11 lines Merged revisions 331658 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331658 | twilson | 2011-08-12 11:30:26 -0500 (Fri, 12 Aug 2011)
| 4 lines Fix netsock2 multiple zero-expansion test Remove
erroneous single bracket. ........ ................
2011-08-12 16:22 +0000 [r331657] Kinsey Moore <kmoore@digium.com>
* /, main/logger.c: Merged revisions 331654 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331654 | kmoore | 2011-08-12 11:21:37 -0500
(Fri, 12 Aug 2011) | 19 lines Merged revisions 331649 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331649 | kmoore | 2011-08-12 11:20:25 -0500 (Fri, 12 Aug 2011) |
12 lines Logger does not warn of failure to open logging channels
Currently, logger only prints an error message to stderr when it
fails to open a logger channel where many users will not see it
because the logger lock is held. The alternative provided by this
patch is to log the error to all attached consoles in the hopes
that it will be easier to see. Additionally, this patch prevents
the failed logger channel from being added to the list where it
would silently fail on each call to the Asterisk logger. (closes
issue ASTERISK-16231) Review:
https://reviewboard.asterisk.org/r/1338 ........ ................
2011-08-11 21:55 +0000 [r331580] Jason Parker <jparker@digium.com>
* apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331579
via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331579 | qwell | 2011-08-11 16:54:54 -0500
(Thu, 11 Aug 2011) | 13 lines Merged revisions 331578 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) |
6 lines Use proper values for 64-bit option flags. Also, reusing
bits es no bueno, so change the value of a duplicate. (issue
ASTERISK-18239) ........ ................
2011-08-11 21:44 +0000 [r331577] Richard Mudgett <rmudgett@digium.com>
* /, funcs/func_shell.c: Merged revisions 331576 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331576 | rmudgett | 2011-08-11 16:42:21 -0500
(Thu, 11 Aug 2011) | 16 lines Merged revisions 331575 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011)
| 9 lines Segfault in shell_helper in func_shell.c. The return
value of popen() was not checked for failure to open. (closes
issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael Myles
Tested by: rmudgett ........ ................
2011-08-10 22:24 +0000 [r331519] Kinsey Moore <kmoore@digium.com>
* /, channels/chan_sip.c: Merged revisions 331518 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331518 | kmoore | 2011-08-10 17:23:49 -0500
(Wed, 10 Aug 2011) | 17 lines Merged revisions 331517 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) |
10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify
sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing
the additional ref just before the invite and adding an unref
following it corrects the issue as seen via REF_DEBUG. The unref
existed in a distant revision and it appears as though the wrong
ref operation was removed. (closes issue ASTERISK-18091) Review:
https://reviewboard.asterisk.org/r/1332/ ........
................
2011-08-10 20:51 +0000 [r331419-331463] Richard Mudgett <rmudgett@digium.com>
* /, main/logger.c: Merged revisions 331462 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331462 | rmudgett | 2011-08-10 15:41:35 -0500
(Wed, 10 Aug 2011) | 37 lines Merged revisions 331461 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331461 | rmudgett | 2011-08-10 15:29:59 -0500 (Wed, 10 Aug 2011)
| 30 lines Output of queue log not started until logger reloaded.
ASTERISK-15863 caused a regression with queue logging. The output
of the queue log is not started until the logger configuration is
reloaded. * Queue log initialization is completely delayed until
the first message is posted to the queue log system. Including
the initial opening of the queue log file. * Fixed rotate_file()
ROTATE strategy to give the file just rotated out to the
configured exec function after rotate. Just like the other
strategies. * Fixed logger reload to always post the queue reload
entry instead of just if there is a queue log file. * Refactored
some code to eliminate some redundancy and to reduce stack
utilization. (closes issue ASTERISK-17036) JIRA SWP-2952 Reported
by: Juan Carlos Valero Patches: jira_asterisk_17036_v1.8.patch
(license #5621) patch uploaded by rmudgett Tested by: rmudgett
(closes issue ASTERISK-18208) Reported by: Christian Pinedo
Review: https://reviewboard.asterisk.org/r/1333/ ........
................
* /, main/features.c: Merged revisions 331420 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r331420 | rmudgett | 2011-08-10 14:07:53 -0500 (Wed, 10 Aug 2011)
| 2 lines Make sure feature_request_and_dial() initializes
outstate if passed in. ........
* /, main/features.c, CHANGES: Merged revisions 331418 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r331418 | rmudgett | 2011-08-10 13:25:08 -0500 (Wed, 10 Aug 2011)
| 6 lines Revert -r318141. It was a band-aid that only partially
fixed parking. A better fix is on reviewboard review 1358. (issue
ASTERISK-17374) ........
2011-08-10 15:45 +0000 [r331371] Jonathan Rose <jrose@digium.com>
* channels/chan_sip.c, CHANGES: SIP display-name needed to be empty
for Avaya IP500 In order to address a compatability issue with
certain features on certain devices which rely on display name
content to change behavior, initreqprep in chan_sip.c has been
changed to no longer substitute cid_number into the display name
when cid_name isn't present. Instead, it will send no display
name in that case. (closes issue ASTERISK-16198) Reported by:
Walter Doekes Review: https://reviewboard.asterisk.org/r/1341/
2011-08-10 13:49 +0000 [r331317] Kinsey Moore <kmoore@digium.com>
* main/manager.c, /: Merged revisions 331316 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331316 | kmoore | 2011-08-10 08:48:41 -0500
(Wed, 10 Aug 2011) | 15 lines Merged revisions 331315 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331315 | kmoore | 2011-08-10 08:47:46 -0500 (Wed, 10 Aug 2011) |
8 lines AMI action ModuleReload returns Error if Module: missing
or empty An empty string was not being checked for properly
causing identification of the module to be reloaded to fail and
return an Error with message "No such module." (closes issue
AST-616) ........ ................
2011-08-09 23:17 +0000 [r331266] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c, /, channels/chan_sip.c, main/features.c,
channels/chan_iax2.c, apps/app_parkandannounce.c: Merged
revisions 331265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331265 | rmudgett | 2011-08-09 18:12:49 -0500
(Tue, 09 Aug 2011) | 22 lines Merged revisions 331248 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011)
| 15 lines Misc minor items found in code. * Add some reentrancy
protection in pbx.c when creating the contexts_table hash table.
* Fix inverted test in chan_sip.c conditional code. * Fix
uninitialized variable and use of the wrong variable in
chan_iax2.c. * Fix test of return value in app_parkandannounce.c.
Explicitly testing for -1 is bad if the function does not
actually return that value when it fails. * Fixup some comments
and add some curly braces in features.c. ........
................
2011-08-09 17:12 +0000 [r331202] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c, /,
addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooq931.c:
Merged revisions 331147,331200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331147 | may | 2011-08-09 20:16:55 +0400 (Tue,
09 Aug 2011) | 11 lines Merged revisions 331146 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4
lines move ast_cond_signal for admitted call after all data
filled/freed clear all log channels by pointed number not only
first free allocated callToken in ooh323_answer ........
................ r331200 | may | 2011-08-09 20:36:39 +0400 (Tue,
09 Aug 2011) | 9 lines Setup IP proto version for call in GK mode
Added additional check for IP semantics before parse destination
by ast_parse_args due to it can parse numeric as IP. (closes
issue ASTERISK-18218) Reported by: slesru Patch:
ASTERISK-18218.patch ................
2011-08-09 17:08 +0000 [r331201] Kinsey Moore <kmoore@digium.com>
* funcs/func_enum.c, UPGRADE.txt, main/enum.c: Allow ENUM query
functions to report lookup errors The ENUM dialplan functions do
not report DNS query errors properly. It is useful to
differentiate between failed query (e.g. non-existent domain) vs.
no data records of the appropriate type. This is required to make
overlapped dialing work. (closes issue ASTERISK-13769) Review:
https://reviewboard.asterisk.org/r/1355/ Patch-by: Timo Teras
2011-08-09 16:02 +0000 [r331140-331144] Jason Parker <jparker@digium.com>
* /, doc/asterisk.8: Merged revisions 331143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331143 | qwell | 2011-08-09 10:59:54 -0500
(Tue, 09 Aug 2011) | 9 lines Merged revisions 331142 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug
2011) | 1 line Regenerate asterisk man page from sgml. ........
................
* /, doc/asterisk.8, configs/asterisk.conf.sample,
configs/voicemail.conf.sample, doc/asterisk.sgml: Merged
revisions 331139 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r331139 | qwell | 2011-08-09 10:50:07 -0500
(Tue, 09 Aug 2011) | 19 lines Merged revisions 306999 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011)
| 12 lines Documentation Updates Note default polling setting in
voicemail.conf Add missing config to asterisk.conf Update manpage
(issue #16505) Reported by: tzafrir Patches:
asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46)
Tested by: lathama, tzafrir ........ ................
* doc/asterisk.8, configs/asterisk.conf.sample,
configs/voicemail.conf.sample, doc/asterisk.sgml: Merged
revisions 331138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r331138 | qwell | 2011-08-09 10:47:20 -0500 (Tue, 09 Aug 2011) |
1 line Revert merge of r306999, due to merge conflict. ........
2011-08-08 22:59 +0000 [r331042-331098] Terry Wilson <twilson@digium.com>
* /, UPGRADE.txt, CHANGES, include/asterisk/manager.h: Merged
revisions 331097 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011)
| 5 lines Bump the AMI protocol version to 1.2 As a result of
converting Unlink events that were missed in the AMI 1.1 update
to Bridge events, the AMI protocol version is being incremented.
........
* main/channel.c, /, CHANGES: Merged revisions 331041 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011)
| 6 lines Replace AMI Unlink events with Bridge events A previous
update converted some of the Link and Unlink events to Bridge
events, but a couple of Unlink events were missed. This patch
rectifies the situation. (closes issues ASTERISK-17455) ........
2011-08-08 20:54 +0000 [r331000-331040] Kinsey Moore <kmoore@digium.com>
* /, res/res_musiconhold.c: Merged revisions 331039 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r331039 | kmoore | 2011-08-08 15:53:30 -0500
(Mon, 08 Aug 2011) | 18 lines Merged revisions 331038 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08 Aug 2011) |
11 lines In-queue MOH stops after a periodic announcement If the
seek value is past the end of file when resuming G.722 MOH, MOH
will cease to function for the duration of the MOH session
through all starts and stops until saved state is cleared.
Adjusting the code to guarantee a single valid read (which is
already assumed) fixes the bug. (closes issue ASTERISK-18077)
Review: https://reviewboard.asterisk.org/r/1328/ Tested-by:
Jonathan Rose <jrose@digium.com> ........ ................
* configs/queues.conf.sample, apps/app_queue.c: Log queue member
name when state_interface is set for ADDMEMBER and REMOVEMEMBER
events app_queue logs the events ADDMEMBER and REMOVEMEMBER with
the agent field set to the interface value rather than the
membername value when a member is added with a state_interface
value set. However all other member related queue events are
logged with the membername when a state_interface is set. This
patch makes these fields optionally more consistent and correct.
(closes issue ASTERISK-14769) Review:
https://reviewboard.asterisk.org/r/1286 Patch-by: Jamuel Starkey
Tested-by: Kinsey Moore <kmoore@digium.com>
* apps/app_queue.c: app_queue: Add StateInterface to output of
"queue show" and "QueueStatus" This patch adds the
state_interface of the queue member struct to the output of
"queue show" (CLI command) and "QueueStatus" (AMI action) when
displaying relevant queue member information. For the AMI event
message the variable StateInterface has been added. (closes issue
ASTERISK-18071) Review: https://reviewboard.asterisk.org/r/1300/
Patch-by: Jamuel Starkey
2011-08-05 15:57 +0000 [r330941] David Vossel <dvossel@digium.com>
* /, codecs/codec_resample.c: Merged revisions 330940 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r330940 | dvossel | 2011-08-05 10:53:49 -0500 (Fri, 05 Aug 2011)
| 2 lines The slin resampler is no longer dependent on an
external library, but the dependency was not removed correctly.
........
2011-08-05 08:47 +0000 [r330903] Alexandr Anikin <may@telecom-service.ru>
* addons/ooh323c/src/ooGkClient.c, /,
addons/ooh323c/src/ooCmdChannel.c: Merged revisions 330899 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r330899 | may | 2011-08-05 11:38:28 +0400 (Fri,
05 Aug 2011) | 11 lines Merged revisions 330827 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4
lines change gk client behaivour on rrq/grq failures to setup
timers and next tries after timeout instead of complete failure
in the ooh323 stack ........ ................
2011-08-04 20:53 +0000 [r330845] Terry Wilson <twilson@digium.com>
* /, configure, configure.ac: Merged revisions 330844 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r330844 | twilson | 2011-08-04 15:51:23 -0500
(Thu, 04 Aug 2011) | 11 lines Merged revisions 330843 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r330843 | twilson | 2011-08-04 15:29:19 -0500 (Thu, 04 Aug 2011)
| 4 lines Make libsrtp instructions more explicit when linking
fails (closes issue ASTERISK-18139) ........ ................
2011-08-03 15:16 +0000 [r330707-330764] Kinsey Moore <kmoore@digium.com>
* /, main/Makefile: Merged revisions 330763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r330763 | kmoore | 2011-08-03 10:15:26 -0500
(Wed, 03 Aug 2011) | 16 lines Merged revisions 330762 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r330762 | kmoore | 2011-08-03 10:14:36 -0500 (Wed, 03 Aug 2011) |
9 lines editing files in main/editline does not ensure rebuild of
libedit.a When editing a source file in main/editline, the build
system does not rebuild libedit.a and uses the already existing
one instead. Adding a PHONY to CHECK_SUBDIR fixes this problem.
(closes issue ASTERISK-16221) Patch-by: Walter Doekes ........
................
* channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions
330706 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r330706 | kmoore | 2011-08-03 08:39:06 -0500
(Wed, 03 Aug 2011) | 17 lines Merged revisions 330705 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) |
10 lines Call pickup broken for DAHDI channels when beginning
with # The call pickup feature did not work on DAHDI devices for
anything other than feature codes beginning with * since all
feature codes in chan_dahdi were originally hard-coded to begin
with *. This patch is also applied to chan_dahdi.c to fix this
bug with radio modes. (closes issue AST-621) Review:
https://reviewboard.asterisk.org/r/1336/ ........
................
2011-08-02 20:54 +0000 [r330650] Kevin P. Fleming <kpfleming@digium.com>
* /, res/res_jabber.c: Merged revisions 330649 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r330649 | kpfleming | 2011-08-02 15:52:44 -0500
(Tue, 02 Aug 2011) | 9 lines Merged revisions 330648 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r330648 | kpfleming | 2011-08-02 15:51:56 -0500 (Tue, 02
Aug 2011) | 2 lines Convert an error message to actually be
helpful. ........ ................
2011-08-02 16:19 +0000 [r330577-330593] David Vossel <dvossel@digium.com>
* /, channels/chan_iax2.c: Merged revisions 330586 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r330586 | dvossel | 2011-08-02 11:17:59 -0500
(Tue, 02 Aug 2011) | 15 lines Merged revisions 330581 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011)
| 8 lines Fixes crash in chan_iax2. Fixes crash in chan_iax2
resulting from an edge case in the way control frames are queued
during calltoken negotiation is complete. (closes issue
ASTERISK-17610) Reported by: mgrobecker ........ ................
* /, channels/chan_sip.c: Merged revisions 330579 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r330579 | dvossel | 2011-08-02 11:08:57 -0500
(Tue, 02 Aug 2011) | 9 lines Merged revisions 330578 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r330578 | dvossel | 2011-08-02 11:07:02 -0500 (Tue, 02
Aug 2011) | 2 lines Optimization to buffer initialization fix.
........ ................
* /, channels/chan_sip.c: Merged revisions 330576 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r330576 | dvossel | 2011-08-02 10:55:36 -0500
(Tue, 02 Aug 2011) | 12 lines Merged revisions 330575 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r330575 | dvossel | 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011)
| 5 lines Fixes uninitialized string buffer in log message.
(closes issue ASTERISK-17200) Reported by: lmadsen ........
................
2011-08-01 15:24 +0000 [r330435] Kinsey Moore <kmoore@digium.com>
* /, main/say.c: Merged revisions 330434 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r330434 | kmoore | 2011-08-01 10:23:29 -0500
(Mon, 01 Aug 2011) | 16 lines Merged revisions 330433 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r330433 | kmoore | 2011-08-01 10:22:10 -0500 (Mon, 01 Aug 2011) |
9 lines Incorrect playback for Spanish in some circumstances When
you say the time in spanish and it is 01:00 - 01:59 or 13:00 -
13:59 you must use female pronunciation "1F". The function
"say_date_with_format_es" does not take this in account. (closes
ASTERISK-15016) Patch-by: Luis Jimenez ........ ................
2011-07-31 00:19 +0000 [r330370-330379] Richard Mudgett <rmudgett@digium.com>
* main/astobj2.c: Fixed compiler warning and a couple prototype
mismatches.
* main/channel.c, /: Merged revisions 330369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r330369 | rmudgett | 2011-07-30 18:57:56 -0500
(Sat, 30 Jul 2011) | 11 lines Merged revisions 330368 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011)
| 4 lines Remove some redundant locking code in
ast_do_masquerade(). Also updated some comments. ........
................
2011-07-30 15:54 +0000 [r330313] Gregory Nietsky <gregory@distrotech.co.za>
* main/channel.c, /: Merged revisions 330312 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r330312 | irroot | 2011-07-30 17:34:41 +0200
(Sat, 30 Jul 2011) | 15 lines Merged revisions 330311 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) |
9 lines prevent double masqurading channels when one is been hung
up and deadlock avoidance is used. There is a race condition in
ast_do_masquerade / ast_hangup (at least) Reported by me signed
off by schmidts with input from David Vossel Review:
https://reviewboard.asterisk.org/r/1323/ ........
................
2011-07-29 19:34 +0000 [r330273] Russell Bryant <russell@russellbryant.com>
* include/asterisk/astobj2.h, tests/test_astobj2.c,
channels/chan_iax2.c, main/astobj2.c: astobj2: Avoid using
temporary objects + ao2_find() with OBJ_POINTER. There is a
fairly common pattern making its way through the code base where
we put a temporary object on the stack so we can call ao2_find()
with OBJ_POINTER. The purpose is so that it can be passed into
the object hash function. However, this really seems like a hack
and potentially error prone. This patch is a first stab at
approach to avoid having to do that. It adds a new flag, OBJ_KEY,
which can be used instead of OBJ_POINTER in these situations.
Then, the hash function can know whether it was given an object
or some custom data to hash. The patch also changes some uses of
ao2_find() for iax2_user and iax2_peer objects to reflect how
OBJ_KEY would be used. So long, and thanks for all the fish.
Review: https://reviewboard.asterisk.org/r/1184/
2011-07-29 17:20 +0000 [r330205-330221] Sean Bright <sean@malleable.com>
* /, formats/format_wav.c: Merged revisions 330217 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r330217 | seanbright | 2011-07-29 13:19:42 -0400
(Fri, 29 Jul 2011) | 9 lines Merged revisions 330213 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r330213 | seanbright | 2011-07-29 13:18:56 -0400 (Fri,
29 Jul 2011) | 2 lines Correct the check for O_RDONLY. ........
................
* /, formats/format_wav.c: Merged revisions 330204 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r330204 | seanbright | 2011-07-29 12:58:40 -0400
(Fri, 29 Jul 2011) | 9 lines Merged revisions 330203 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r330203 | seanbright | 2011-07-29 12:58:08 -0400 (Fri,
29 Jul 2011) | 2 lines Only write to wav files that were opened
to be written to. ........ ................
2011-07-29 05:27 +0000 [r330163] Paul Belanger <paul.belanger@polybeacon.com>
* /, apps/app_confbridge.c: Merged revisions 330162 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r330162 | pabelanger | 2011-07-29 01:25:18 -0400 (Fri, 29 Jul
2011) | 4 lines Fix typo pointed out on #asterisk Thanks notten
........
2011-07-28 21:46 +0000 [r330109] Terry Wilson <twilson@digium.com>
* /, main/term.c: Merged revisions 330108 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r330108 | twilson | 2011-07-28 16:44:31 -0500
(Thu, 28 Jul 2011) | 9 lines Merged revisions 330107 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r330107 | twilson | 2011-07-28 16:42:41 -0500 (Thu, 28
Jul 2011) | 2 lines Make console colors work for
TERM=xterm-256color ........ ................
2011-07-28 17:16 +0000 [r330052] Richard Mudgett <rmudgett@digium.com>
* /, channels/sig_pri.c: Merged revisions 330051 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r330051 | rmudgett | 2011-07-28 12:10:37 -0500
(Thu, 28 Jul 2011) | 29 lines Merged revisions 330050 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
................ r330050 | rmudgett | 2011-07-28 12:04:24 -0500
(Thu, 28 Jul 2011) | 22 lines Merged revisions 330033 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
.......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu,
28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and
outgoing call legs of a data call are using different formats:
a-law, u-law. When the call is bridged, the media stream is run
through translation to convert the media formats. The translation
is bad for data calls. * Make incoming call that does not
explicitly specify u-law or a-law use the DAHDI channel's default
law. The outgoing call always uses the default law from the DAHDI
channel. (closes issue ABE-2800) Patches:
jira_abe_2800_companding.patch (license #5621) patch uploaded by
rmudgett .......... ................ ................
2011-07-28 15:46 +0000 [r329996] Jason Parker <jparker@digium.com>
* /, channels/chan_sip.c: Merged revisions 329995 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r329995 | qwell | 2011-07-28 10:45:49 -0500
(Thu, 28 Jul 2011) | 13 lines Merged revisions 329994 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r329994 | qwell | 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) |
6 lines Fix a SIP transfer deadlock. The locking in this function
is very scary. There are like 6 structs involved. (closes issue
AST-470) ........ ................
2011-07-28 15:30 +0000 [r329993] Matthew Nicholson <mnicholson@digium.com>
* /, res/res_fax.c: Merged revisions 329992 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r329992 | mnicholson | 2011-07-28 10:28:21 -0500
(Thu, 28 Jul 2011) | 13 lines Merged revisions 329991 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r329991 | mnicholson | 2011-07-28 10:26:56 -0500 (Thu, 28 Jul
2011) | 6 lines check for CONFIG_STATUS_FILE_INVALID when loading
the res_fax config file Patch by: tzafrir Reported by: tzafrir
(closes issue ASTERISK-18161) ........ ................
2011-07-28 13:04 +0000 [r329897-329953] Sean Bright <sean@malleable.com>
* configs/confbridge.conf.sample, /: Merged revisions 329952 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
........ r329952 | seanbright | 2011-07-28 09:03:58 -0400 (Thu,
28 Jul 2011) | 4 lines The default conf-usermenu says that '8'
can be used to leave the conference, so put that in the sample
user menu. '5' is supposed to extend the conference, but there
doesn't appear to be a concept of that in the menu actions.
........
* /, apps/app_confbridge.c: Merged revisions 329950 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10 ........
r329950 | seanbright | 2011-07-28 08:43:55 -0400 (Thu, 28 Jul
2011) | 1 line Correct the spelling of 'conference.' ........
* /, channels/chan_sip.c: Merged revisions 329896 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r329896 | seanbright | 2011-07-28 07:35:27 -0400
(Thu, 28 Jul 2011) | 9 lines Merged revisions 329895 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r329895 | seanbright | 2011-07-28 07:34:33 -0400 (Thu,
28 Jul 2011) | 2 lines Make the output of Externhost in 'sip show
settings' more consistent. ........ ................
2011-07-27 21:22 +0000 [r329835-329856] Jonathan Rose <jrose@digium.com>
* main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES:
reverting 329840 due to failing tests. Going to change this
feature to be purely optional.
* main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Adds cdr
logging of calls resulting in CONGESTION Applies a patch made a
long time ago by alecdavis which adds a CDR feature for logging
calls that failed due to congestion. (closes issue #15907)
Reported by: alecdavis Patches: cdr_congestion.diff.txt uploaded
by alecdavis (license #5546) Review:
https://reviewboard.asterisk.org/r/454/
2011-07-27 19:19 +0000 [r329775] Sean Bright <sean@malleable.com>
* /, Makefile.moddir_rules: Merged revisions 329771 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r329771 | seanbright | 2011-07-27 15:18:47 -0400
(Wed, 27 Jul 2011) | 15 lines Merged revisions 329767 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r329767 | seanbright | 2011-07-27 15:17:46 -0400 (Wed, 27 Jul
2011) | 8 lines Explicitly sort the module list so that the
menuselect lists are sorted. (closes ASTERISK-18141) Reported by:
Richard Miller Patches: sort-order.diff uploaded by seanbright
(License #5060) Tested by: leifmadsen ........ ................
2011-07-27 18:12 +0000 [r329711] Jonathan Rose <jrose@digium.com>
* /, configs/indications.conf.sample: Merged revisions 329710 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r329710 | jrose | 2011-07-27 13:11:07 -0500
(Wed, 27 Jul 2011) | 14 lines Merged revisions 329709 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r329709 | jrose | 2011-07-27 13:10:30 -0500 (Wed, 27 Jul 2011) |
8 lines Fix New Zealand indications profile based on
http://www.telepermit.co.nz/TNA102.pdf (closes issue
ASTERISK-16263) Reported by: richardf Patches:
nz-indications.patch uploaded by richardf (License #6015)
........ ................
2011-07-27 15:26 +0000 [r329671] Sean Bright <sean@malleable.com>
* /, main/loader.c: Merged revisions 329670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r329670 | seanbright | 2011-07-27 11:25:53 -0400 (Wed, 27 Jul
2011) | 2 lines Sort the module list so that 'module show' is
alphabetical. ........
2011-07-27 04:27 +0000 [r329615] Tilghman Lesher <tilghman@meg.abyt.es>
* /, cdr/cdr_odbc.c: Merged revisions 329614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r329614 | tilghman | 2011-07-26 23:25:26 -0500
(Tue, 26 Jul 2011) | 13 lines Merged revisions 329613 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r329613 | tilghman | 2011-07-26 23:23:46 -0500 (Tue, 26 Jul 2011)
| 6 lines Duration and billsec are swapped in high resolution
time. Closes ASTERISK-18024 Patches:
20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003)
........ ................
2011-07-26 14:27 +0000 [r329530-329564] Jonathan Rose <jrose@digium.com>
* /, apps/app_voicemail.c: Merged revisions 329538 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/10
................ r329538 | jrose | 2011-07-26 09:19:34 -0500
(Tue, 26 Jul 2011) | 11 lines Merged revisions 329529 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r329529 | jrose | 2011-07-26 09:04:55 -0500 (Tue, 26 Jul 2011) |
5 lines Changes sound file for prepend "then-press-pound" to
"vm-then-pound" which is the same prompt, only it turned out
"then-press-pound" was part of extra sounds. Also, vm is more
appropriate anyway. ........ ................
* include/asterisk/app.h, /, configs/voicemail.conf.sample,
main/app.c, apps/app_voicemail.c: Merged revisions 329528 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r329528 | jrose | 2011-07-26 08:52:34 -0500
(Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) |
17 lines Fixes some voicemail forwarding behavior based around
prepend mode. Formerly, prepend forwarding would have the user
record a message with no useful prompt and an expectation for the
user to push a button on the phone when finished recording. If a
length of silence was detected instead, the recording would be
canceled and the user would re-enter the voicemail forwarding
menu. Subsequent time-outs in prepend recording would also bug
out in the sense that they would write over the original message
and get sent to the recipient regardless of whether they timed
out or were accepted. This patch fixes this issue and adds a
prompt which will be played after a timeout informing the user
that they needed to press a button. Currently, the sound files
that we have are somewhat inadquate for this, so after the call
we simply have Allison say "Please try again. Then press pound."
which actually relies on two separate sound files. Just one would
be more appropriate. reporter: Vlad Povorozniuc Review:
https://reviewboard.asterisk.org/r/1327/ ........
................
2011-07-25 19:57 +0000 [r329473] Paul Belanger <paul.belanger@polybeacon.com>
* /, main/enum.c: Merged revisions 329472 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r329472 | pabelanger | 2011-07-25 15:55:33 -0400
(Mon, 25 Jul 2011) | 9 lines Merged revisions 329471 via svnmerge
from https://origsvn.digium.com/svn/asterisk/branches/1.8
........ r329471 | pabelanger | 2011-07-25 15:49:40 -0400 (Mon,
25 Jul 2011) | 2 lines Decrease verbose messages to debug, to
help clean up CLI. ........ ................
2011-07-25 14:07 +0000 [r329391-329432] Gregory Nietsky <gregory@distrotech.co.za>
* include/asterisk/dsp.h, main/dsp.c: dsp_process was enhanced to
work with alaw and ulaw in addition to slin. noticed that some
functions could be refactored here it is. Reported by: irroot
Tested by: irroot, mnicholson Review:
https://reviewboard.asterisk.org/r/1304/
* channels/chan_sip.c, channels/sip/include/sip.h: Remove
lastmsgssent from sip it has not been working since 1.6 Clean up
the return values to be consistant not currently used Add doxygen
returns MWI Event is sent on Register (closes issue
ASTERISK-17866) Reported by: one47 Tested by: irroot, mvanbaak
Review: https://reviewboard.asterisk.org/r/1172/
2011-07-22 21:15 +0000 [r329332-329335] Richard Mudgett <rmudgett@digium.com>
* main/pbx.c, /: Merged revisions 329334 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10 ........
r329334 | rmudgett | 2011-07-22 16:14:22 -0500 (Fri, 22 Jul 2011)
| 1 line Make use less redundant loop construct for iterating
over hints. ........
* main/pbx.c, /: Merged revisions 329331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r329331 | rmudgett | 2011-07-22 15:43:07 -0500
(Fri, 22 Jul 2011) | 55 lines Merged revisions 329299 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011)
| 48 lines Deadlocks dealing with dialplan hints during reload.
There are two remaining different deadlocks reported dealing with
dialplan hints. The deadlock in ASTERISK-17666 is caused by
invalid locking order in ast_remove_hint(). The hints container
must be locked before the hint object. The deadlock in
ASTERISK-17760 is caused by a catch-22 situation in
handle_statechange(). The deadlock is caused by not having the
conlock before calling the watcher callbacks. Unfortunately,
having that lock causes a different deadlock as reported in
ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made
handle_statechange() no longer call the watcher callbacks holding
any locks that matter. * Made hint ao2 destructor do the watcher
callbacks for extension deactivation to guarantee that they get
called. * Fixed hint reference leak in ast_add_hint() if the
callback container constructor failed. * Fixed hint reference
leak in complete_core_show_hint() for every hint it found for CLI
tab completion. * Adjusted locking in
ast_merge_contexts_and_delete() for safety. * Added
context_merge_lock to prevent ast_merge_contexts_and_delete() and
handle_statechange() from interfering with each other. * Fixed
ast_change_hint() not taking into account that the extension is
used for the hash key. (closes issue ASTERISK-17666) Reported by:
irroot Tested by: irroot JIRA SWP-3318 (closes issue
ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA
SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/
........ ................
2011-07-21 20:26 +0000 [r329258] Russell Bryant <russell@russellbryant.com>
* channels/chan_dahdi.c, /, main/features.c,
include/asterisk/netsock2.h, CHANGES, channels/sig_pri.c,
include/asterisk/rtp_engine.h: Merged revisions 329257 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
........ r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21
Jul 2011) | 2 lines s/1.10/10.0/ ........
2011-07-21 18:06 +0000 [r329146-329205] Richard Mudgett <rmudgett@digium.com>
* channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged
revisions 329204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r329204 | rmudgett | 2011-07-21 13:05:18 -0500
(Thu, 21 Jul 2011) | 13 lines Merged revisions 329203 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011)
| 6 lines Document parkinglot in chan_dahdi.conf.sample. *
Document existing feature in chan_dahdi.conf.sample. * Remove
some dead code related to the parkinglot option. ........
................
* /, apps/app_directed_pickup.c: Merged revisions 329200 via
svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10
................ r329200 | rmudgett | 2011-07-21 12:32:02 -0500
(Thu, 21 Jul 2011) | 24 lines Merged revisions 329199 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011)
| 17 lines Update PickupChan documentation. The PickupChan uses
the ampersand as the argument separator. Was documented as:
PickupChan(channel[,channel2[,...][,options]]) Fixed
documentation to:
PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options])
This is a continuation of ASTERISK-17494 for v1.8 and later.
(closes issue ASTERISK-18144) Reported by: Erik Smith Patches:
pickupchan_ducumentation-v2.patch (License #6263) patch uploaded
by Erik Smith Tested by: Erik Smith ........ ................
* /, main/features.c: Merged revisions 329145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10
................ r329145 | rmudgett | 2011-07-21 11:52:17 -0500
(Thu, 21 Jul 2011) | 16 lines Merged revisions 329144 via
svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........
r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011)
| 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed
more times than we've locked! This appears to be a leftover from
when ast_channel was converted to ao2 objects. Simply removed the
extraneous unlock. (closes issue ASTERISK-17772) ........
................
2011-07-21 16:22 +0000 [r329106-329130] Jason Parker <jparker@digium.com>
* UPGRADE-1.10.txt (removed), UPGRADE-10.txt (added), UPGRADE.txt:
Fix UPGRADE.txt files for Asterisk 10.
* /: Remove another 2.0 property.
2011-07-21 16:05 +0000 [r329105] Russell Bryant <russell@russellbryant.com>
* /: Fix merge properties to reflect Asterisk 10 branch
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