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385
ChangeLog
385
ChangeLog
@@ -1,3 +1,388 @@
|
||||
2015-01-06 Asterisk Development Team <asteriskteam@digium.com>
|
||||
|
||||
* Certified Asterisk 13.1-cert1-rc1 Released.
|
||||
|
||||
2015-01-06 19:53 +0000 [r430245] Scott Griepentrog <sgriepentrog@digium.com>
|
||||
|
||||
* /, main/bridge_basic.c: bridge: avoid leaking channel during
|
||||
blond transfer pt2 A blond transfer to a failed destination, when
|
||||
followed by a recall attempt, lead to a leak of the reference to
|
||||
the destination channel. In addition to correcting the regression
|
||||
on the previous attempt (r429826) this fixes the leak and two
|
||||
additional reference leaks on failures of bridge_import.
|
||||
ASTERISK-24513 #close Review:
|
||||
https://reviewboard.asterisk.org/r/4302/ ........ Merged
|
||||
revisions 430199 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
|
||||
revisions 430200 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
2014-12-24 15:27 +0000 [r430085-430094] Matthew Jordan <mjordan@digium.com>
|
||||
|
||||
* res/res_agi.c, /: res/res_agi: Make Verbose message for 'stream
|
||||
file' match other playbacks The Verbose message displayed when a
|
||||
file is played back via 'stream file' was formatted differently
|
||||
than other playbacks: * It didn't include the channel name * It
|
||||
didn't include the channel language It does, however, include the
|
||||
playback offset as well as any escape digits. That information
|
||||
was kept; however, this patch updates the formatting to more
|
||||
closely match the Verbose messages displayed when a file is
|
||||
played back by 'control stream file', Playback, ControlPlayback,
|
||||
or any other file playback operation. ........ Merged revisions
|
||||
429519 from http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* /, res/res_pjsip.c,
|
||||
contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py
|
||||
(added): res_pjsip: Backport missing commits for user_eq_phone
|
||||
This backports the following from trunk, which were missed:
|
||||
r427257 | file | 2014-11-04 16:31:16 -0600 (Tue, 04 Nov 2014) | 2
|
||||
lines res_pjsip: Allow + at the beginning of a phone number when
|
||||
user_eq_phone is enabled. r427259 | file | 2014-11-04 16:51:32
|
||||
-0600 (Tue, 04 Nov 2014) | 2 lines res_pjsip: Apply the
|
||||
'user_eq_phone' setting to the To header as well. It also adds
|
||||
the Alembic script for the option. ASTERISK-24643 ........ Merged
|
||||
revisions 430092 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* /, tests/test_stasis_channels.c: Stasis: Update unittest for
|
||||
channel snapshots This adjusts the unit test for channel
|
||||
snapshots to take the new language key into account. ........
|
||||
Merged revisions 429352 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* res/res_pjsip_keepalive.c (added), res/res_pjsip/config_global.c,
|
||||
/, configs/samples/pjsip.conf.sample, CHANGES, res/res_pjsip.c,
|
||||
include/asterisk/res_pjsip.h: res_pjsip_keepalive: Add runtime
|
||||
configurable keepalive module for connection-oriented transports.
|
||||
Note that this is backport from trunk of r425825. This change
|
||||
adds a module which is configurable using the keep_alive_interval
|
||||
setting in the global section that will send a CRLF keep alive to
|
||||
all active connection-oriented transports at the provided
|
||||
interval. This is useful because it can help keep connections
|
||||
open through NATs. This functionality also exists within PJSIP
|
||||
but can not be controlled at runtime and requires recompiling it.
|
||||
Review: https://reviewboard.asterisk.org/r/4084/ ASTERISK-24644
|
||||
#close ........ Merged revisions 430084 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* include/asterisk/res_pjsip.h, /,
|
||||
res/res_pjsip/pjsip_configuration.c, res/res_pjsip_caller_id.c,
|
||||
CHANGES, res/res_pjsip.c: res_pjsip: Add 'user_eq_phone' option
|
||||
to add a 'user=phone' parameter when applicable. Note that this
|
||||
is a backport of r425804 from trunk. This change adds a
|
||||
configuration option which adds a 'user=phone' parameter if the
|
||||
user portion of the request URI or the From URI is determined to
|
||||
be a number. Review: https://reviewboard.asterisk.org/r/4073/
|
||||
ASTERISK-24643 #close ........ Merged revisions 430083 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
2014-12-22 21:22 +0000 [r430030-430046] Richard Mudgett <rmudgett@digium.com>
|
||||
|
||||
* /, main/bridge_basic.c: DTMF atxfer: Setup recall channels as if
|
||||
the transferee initiated the call. After the initial DTMF atxfer
|
||||
call attempt to the transfer target fails to answer during a
|
||||
blonde transfer, the recall callback channels do not get setup
|
||||
with information from the initial transferrer channel. As a
|
||||
result, the recall callback to the transferrer does not have
|
||||
callid, channel variables, datastores, accountcode, peeraccount,
|
||||
COLP, and CLID setup. A similar situation happens with the recall
|
||||
callback to the transfer target but it is less visible. The
|
||||
recall callback to the transfer target does not have callid,
|
||||
channel variables, datastores, accountcode, peeraccount, and COLP
|
||||
setup. * Added missing information to the recall callback
|
||||
channels before initiating the call. callid, channel variables,
|
||||
datastores, accountcode, peeraccount, COLP, and CLID * Set callid
|
||||
of the transferrer channel on the DTMF atxfer controller thread
|
||||
attended_transfer_monitor_thread(). * Added missing channel
|
||||
unlocks and props unref to off nominal paths in
|
||||
attended_transfer_properties_alloc(). ASTERISK-23841 #close
|
||||
Reported by: Richard Mudgett Review:
|
||||
https://reviewboard.asterisk.org/r/4259/ ........ Merged
|
||||
revisions 430034 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* /, main/logger.c, include/asterisk/_private.h, main/asterisk.c:
|
||||
queue_log: Post QUEUESTART entry when Asterisk fully boots. The
|
||||
QUEUESTART log entry has historically acted like a fully booted
|
||||
event for the queue_log file. When the QUEUESTART entry was
|
||||
posted to the log was broken by the change made by
|
||||
ASTERISK-15863. * Made post the QUEUESTART queue_log entry when
|
||||
Asterisk fully boots. This restores the intent of that log entry
|
||||
and happens after realtime has had a chance to load. AST-1444
|
||||
#close Reported by: Denis Martinez Review:
|
||||
https://reviewboard.asterisk.org/r/4282/ ........ Merged
|
||||
revisions 430009 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
|
||||
revisions 430010 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
2014-12-22 18:35 +0000 [r430007-430008] bebuild <bebuild@localhost>:
|
||||
|
||||
* res/res_pjsip/pjsip_options.c, /: Multiple revisions
|
||||
429128,429246 ........ r429128 | kmoore | 2014-12-09 08:00:50
|
||||
-0600 (Tue, 09 Dec 2014) | 12 lines PJSIP: Stagger outbound
|
||||
qualifies This change staggers initiation of outbound qualify
|
||||
(OPTIONS) attempts to reduce instantaneous server load and
|
||||
prevent network congestion. Review:
|
||||
https://reviewboard.asterisk.org/r/4246/ ASTERISK-24342 #close
|
||||
Reported by: Richard Mudgett ........ Merged revisions 429127
|
||||
from http://svn.asterisk.org/svn/asterisk/branches/12 ........
|
||||
r429246 | kmoore | 2014-12-10 07:14:56 -0600 (Wed, 10 Dec 2014) |
|
||||
8 lines PJSIP: Fix assert on initial mass qualify This fixes the
|
||||
MWI test regressions caused by r429127 and ensures that contacts
|
||||
have non-zero qualify_frequency before attempting scheduling.
|
||||
........ Merged revisions 429245 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
|
||||
revisions 429128,429246 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* main/manager.c, /: Prevent possible race condition on dual
|
||||
redirect of channels in the same bridge. The
|
||||
AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT flag was created to prevent
|
||||
bridges from prematurely acting on orphaned channels in bridges.
|
||||
The problem with the AMI redirect action was that it was setting
|
||||
this flag on channels based on the presence of a PBX, not whether
|
||||
the channel was in a bridge. Whether a channel has a PBX is
|
||||
irrelevant, so the condition has been altered to check if the
|
||||
channel is in a bridge. ASTERISK-24536 #close Reported by Niklas
|
||||
Larsson Review: https://reviewboard.asterisk.org/r/4268 ........
|
||||
Merged revisions 429741 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
2014-12-19 21:52 +0000 [r429855-429892] bebuild <bebuild@localhost>:
|
||||
|
||||
* res/res_ari_channels.c, res/ari/resource_channels.h, /,
|
||||
rest-api/api-docs/channels.json, res/ari/resource_channels.c,
|
||||
CHANGES: ari: Add support for specifying an originator channel
|
||||
when originating. If an originator channel is specified when
|
||||
originating a channel the linked ID of it will be applied to the
|
||||
newly originated outgoing channel. This allows an association to
|
||||
be made between the two so it is known that the originator has
|
||||
dialed the originated channel. ASTERISK-24552 #close Reported by:
|
||||
Matt Jordan Review: https://reviewboard.asterisk.org/r/4243/
|
||||
........ Merged revisions 429153 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* main/stasis_channels.c, rest-api/api-docs/channels.json,
|
||||
res/ari/ari_model_validators.c, main/manager_channels.c,
|
||||
res/ari/ari_model_validators.h, /: ARI/AMI: Include language in
|
||||
standard channel snapshot output The channel "language" was
|
||||
already part of a channel snapshot, however is was not sent out
|
||||
over AMI or ARI. This patch makes it so the channel "language" is
|
||||
included in the appropriate AMI or ARI events. ASTERISK-24553
|
||||
#close Reported by: Matt Jordan Review:
|
||||
https://reviewboard.asterisk.org/r/4245/ ........ Merged
|
||||
revisions 429204 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
|
||||
revisions 429206 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* res/res_pjsip_session.c, /: res_pjsip_session: Fix issue where a
|
||||
declined media stream in a re-INVITE would fail SDP negotiation.
|
||||
In the past the SDP negotiation within res_pjsip_session was made
|
||||
more tolerant of certain situations. The only case where SDP
|
||||
negotiation will fail is when a major error occurs during
|
||||
negotiation. Receiving an already declined media stream is not
|
||||
considered a major error. When producing the local SDP the logic
|
||||
took this into account so on the initial INVITE the declined
|
||||
media stream did not cause an SDP negotiation failure.
|
||||
Unfortunately the logic for handling media streams with a handler
|
||||
did not mirror this logic and considered an already declined
|
||||
media stream an error and thus failed the SDP negotiation. This
|
||||
change makes the logic between both situations match so only
|
||||
under major errors will the SDP negotiation fail. ASTERISK-24607
|
||||
#close Reported by: Matt Jordan Review:
|
||||
https://reviewboard.asterisk.org/r/4254/ ........ Merged
|
||||
revisions 429407 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* include/asterisk/format.h, main/format.c, /, main/codec.c: media:
|
||||
Fix crash when determining sample count of a frame during
|
||||
shutdown. When shutting down Asterisk the codecs are cleaned up.
|
||||
As a result anything attempting to get a codec based on ID or
|
||||
details will find that no codec exists. This currently occurs
|
||||
when determining the sample count of a frame. This code did not
|
||||
take this situation into account. This change fixes this by
|
||||
getting the codec directly from the format and eliminates the
|
||||
lookup. This is both faster and also provides a guarantee that
|
||||
the codec will exist and will be valid. ASTERISK-24604 #close
|
||||
Reported by: Matt Jordan Review:
|
||||
https://reviewboard.asterisk.org/r/4260/ ........ Merged
|
||||
revisions 429497 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* /, res/res_pjsip_outbound_registration.c: Prevent potential
|
||||
infinite outbound authentication loops in registration. Prior to
|
||||
this patch, Asterisk would always respond to 401 responses to
|
||||
registration attempts by trying to provide a registration with
|
||||
authentication credentials. Even if subsequent attempts were
|
||||
rejected with 401 responses, Asterisk would continue this
|
||||
behavior. If authentication credentials were incorrect, this
|
||||
could continue forever. With this patch, we keep track of whether
|
||||
we have attempted authentication on an outbound registration
|
||||
attempt. If we already have, we don not try again until the next
|
||||
attempt. This prevents the infinite loop scenario. Review:
|
||||
https://reviewboard.asterisk.org/r/4273 ........ Merged revisions
|
||||
429761 from http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* res/res_pjsip_outbound_publish.c, /: res_pjsip_outbound_publish:
|
||||
stack overflow when using non-default sorcery wizard When using a
|
||||
non-default sorcery wizard (in this instance realtime) for
|
||||
outbound publishes Asterisk will crash after a stack overflow
|
||||
occurs due to the code infinitely recursing. The fix entails
|
||||
removing the outbound publish state dependency from the outbound
|
||||
publish sorcery object and instead keeping an in memory container
|
||||
that can be used to lookup the state when needed. ASTERISK-24514
|
||||
#close Reported by: Mark Michelson Review:
|
||||
https://reviewboard.asterisk.org/r/4178/ ........ Merged
|
||||
revisions 429175 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* /, res/res_pjsip_sdp_rtp.c: PJSIP: Allow use of 'inactive'
|
||||
streams for hold This allows use of the 'inactive' stream
|
||||
direction identifier to be used for hold where 'sendonly' is
|
||||
normally used. Some Seimens phones use 'inactive' and this change
|
||||
allows music on hold to operate properly. Review:
|
||||
https://reviewboard.asterisk.org/r/4252/ Reported by: Steve Pitts
|
||||
........ Merged revisions 429432 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
|
||||
revisions 429433 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* include/asterisk/res_pjsip_session.h, /,
|
||||
res/res_pjsip_session.exports.in, channels/chan_pjsip.c,
|
||||
res/res_pjsip_session.c: res_pjsip_session: Delay sending BYE if
|
||||
a re-INVITE transaction is in progress. Given the scenario where
|
||||
a PJSIP channel is in a native RTP bridge with direct media and
|
||||
the channel is then hung up the code will currently re-INVITE the
|
||||
channel back to Asterisk and send a BYE at the same time. Many
|
||||
SIP implementations dislike this greatly. This change makes it so
|
||||
that if a re-INVITE transaction is in progress the BYE is queued
|
||||
to occur after the completion of the transaction (be it through
|
||||
normal means or a timeout). Review:
|
||||
https://reviewboard.asterisk.org/r/4248/ ........ Merged
|
||||
revisions 429409 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* channels/chan_pjsip.c, /: chan_pjsip: Race between channel answer
|
||||
and bridge setup when using direct media When direct media is
|
||||
enabled and a pjsip channel is answered a race would occur
|
||||
between the handling of the answer and bridge setup. Sometimes
|
||||
the media negotiation would take place after the native bridge
|
||||
was setup. This resulted in a NULL media address, which in turn
|
||||
resulted in Asterisk using its address as the remote media
|
||||
address when sending a reinvite. This patch makes the chan_pjsip
|
||||
answer handler synchronous thus alleviating the race condition
|
||||
(the bridge won't start setting things up until after it
|
||||
returns). ASTERISK-24563 #close Reported by: Steve Pitts Review:
|
||||
https://reviewboard.asterisk.org/r/4257/ ........ Merged
|
||||
revisions 429477 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* /, channels/chan_sip.c, include/asterisk/rtp_engine.h,
|
||||
res/res_rtp_asterisk.c, main/rtp_engine.c: Direct Media calls
|
||||
within private network sometimes get one way audio When endpoints
|
||||
with direct_media enabled, behind a firewall (Asterisk on a
|
||||
separate network) and were bridged sometimes Asterisk would send
|
||||
the ip address of the firewall in the sdp to one of the phones in
|
||||
the reinvite resulting in one way audio. When sending the
|
||||
reinvite Asterisk will retrieve the media address from the
|
||||
associated rtp instance, but if frames were being read this can
|
||||
be overwritten with another address (in this case the
|
||||
firewall's). This patch ensures that Asterisk uses the original
|
||||
device address when using direct media. ASTERISK-24563 Reported
|
||||
by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4216/
|
||||
........ Merged revisions 429195 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged
|
||||
revisions 429196 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* channels/pjsip/dialplan_functions.c, /: Ensure the correct value
|
||||
is returned for CHANNEL(pjsip, secure) Prior to this patch, we
|
||||
were using the PJSIP dialog's secure flag to determine if a
|
||||
secure transport was being used. Unfortunately, the dialog's
|
||||
secure flag was only set if a SIPS URI were in use, as required
|
||||
by RFC 3261 sections 12.1.1 and 12.1.2. What we're interested in
|
||||
is not dialog security, but transport security. This code change
|
||||
switches to a model where we use the dialog's target URI to
|
||||
determine what transport would be used to communicate, and then
|
||||
check if that transport is secure. AST-1450 #close Reported by
|
||||
John Bigelow Review: https://reviewboard.asterisk.org/r/4277
|
||||
........ Merged revisions 429739 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* channels/chan_dahdi.c, /: chan_dahdi: Don't ignore setvar when
|
||||
using configuration section scheme. When the configuration
|
||||
section scheme of chan_dahdi.conf is used (keyword dahdichan
|
||||
instead of channel) all setvar= options are completely ignored.
|
||||
No variable defined this way appears in the created DAHDI
|
||||
channels. * Move the clearing of setvar values to after the
|
||||
deferred processing of dahdichan. AST-1378 #close Reported by:
|
||||
Guenther Kelleter Patch by: Guenther Kelleter ........ Merged
|
||||
revisions 429825 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
|
||||
revisions 429829 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* /, include/asterisk/lock.h, main/lock.c: DEBUG_THREADS: Fix
|
||||
regression and lock tracking initialization problems. This patch
|
||||
started with David Lee's patch at
|
||||
https://reviewboard.asterisk.org/r/2826/ and includes a
|
||||
regression fix introduced by the ASTERISK-22455 patch. The
|
||||
initialization of a mutex's lock tracking structure was not
|
||||
protected in a critical section. This is fine for any mutex that
|
||||
is explicitly initialized, but a static mutex may have its lock
|
||||
tracking double initialized if multiple threads attempt the first
|
||||
lock simultaneously. * Added a global mutex to properly serialize
|
||||
initialization of the lock tracking structure. The painful global
|
||||
lock can be mitigated by adding a double checked lock flag as
|
||||
discussed on the original review request. * Defer lock tracking
|
||||
initialization until first use. * Don't be "helpful" and
|
||||
initialize an uninitialized lock when DEBUG_THREADS is enabled.
|
||||
Debug code is not supposed to fix or change normal code behavior.
|
||||
We don't need a lock initialization race that would force a
|
||||
re-setup of lock tracking. Lock tracking already handles
|
||||
initialization on first use. * Properly handle allocation
|
||||
failures of the lock tracking structure. * No need to initialize
|
||||
tracking data in __ast_pthread_mutex_destroy() just to turn
|
||||
around and destroy it. The regression introduced by
|
||||
ASTERISK-22455 is the result of manipulating a pthread_mutex_t
|
||||
struct outside of the pthread library code. The pthread_mutex_t
|
||||
struct seems to have a global linked list pointer member that can
|
||||
get changed by other threads. Therefore, saving and restoring the
|
||||
contents of a pthread_mutex_t struct is a bad thing. Thanks to
|
||||
Thomas Airmont for finding this obscure regression. * Don't
|
||||
overwrite the struct ast_lock_track.reentr_mutex member to
|
||||
restore tracking data in __ast_cond_wait() and
|
||||
__ast_cond_timedwait(). The pthread_mutex_t struct must be
|
||||
treated as a read-only opaque variable. Miscellaneous other items
|
||||
fixed by this patch: * Match ast_suspend_lock_info() with
|
||||
ast_restore_lock_info() in __ast_cond_timedwait(). * Made some
|
||||
uninitialized lock sanity checks return EINVAL and try a
|
||||
DO_THREAD_CRASH. * Fix bad canlog initialization expressions.
|
||||
ASTERISK-24614 #close Reported by: Thomas Airmont Review:
|
||||
https://reviewboard.asterisk.org/r/4247/ Review:
|
||||
https://reviewboard.asterisk.org/r/2826/ ........ Merged
|
||||
revisions 429539 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged
|
||||
revisions 429540 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* res/res_pjsip_pubsub.c, /: Activate persistent subscriptions when
|
||||
they are recreated. Prior to this change, recreating persistent
|
||||
subscriptions would create the subscription but would not
|
||||
activate it. This led to subscriptions being listed in the "NULL"
|
||||
state by diagnostics and not sending NOTIFYs when expected.
|
||||
Review: https://reviewboard.asterisk.org/r/4261 ........ Merged
|
||||
revisions 429571 from
|
||||
http://svn.asterisk.org/svn/asterisk/branches/13
|
||||
|
||||
* asterisk-13.1.0-summary.html (removed),
|
||||
asterisk-13.1.0-summary.txt (removed), /: Update properties;
|
||||
remove old summaries
|
||||
|
||||
* / (added): Create Certified Asterisk 13.1 branch
|
||||
|
||||
2014-12-15 Asterisk Development Team <asteriskteam@digium.com>
|
||||
|
||||
* Asterisk 13.1.0 Released.
|
||||
|
||||
222
certified-asterisk-13.1-cert1-rc1-summary.html
Normal file
222
certified-asterisk-13.1-cert1-rc1-summary.html
Normal file
@@ -0,0 +1,222 @@
|
||||
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN" http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
|
||||
<html xmlns="http://www.w3.org/1999/xhtml">
|
||||
<head><meta http-equiv="Content-Type" content="text/html; charset=iso-8859-1" /><title>Release Summary - certified-asterisk-13.1-cert1-rc1</title></head>
|
||||
<body>
|
||||
<h1 align="center"><a name="top">Release Summary</a></h1>
|
||||
<h3 align="center">certified-asterisk-13.1-cert1-rc1</h3>
|
||||
<h3 align="center">Date: 2015-01-06</h3>
|
||||
<h3 align="center"><asteriskteam@digium.com></h3>
|
||||
<hr/>
|
||||
<h2 align="center">Table of Contents</h2>
|
||||
<ol>
|
||||
<li><a href="#summary">Summary</a></li>
|
||||
<li><a href="#contributors">Contributors</a></li>
|
||||
<li><a href="#issues">Closed Issues</a></li>
|
||||
<li><a href="#commits">Other Changes</a></li>
|
||||
<li><a href="#diffstat">Diffstat</a></li>
|
||||
</ol>
|
||||
<hr/>
|
||||
<a name="summary"><h2 align="center">Summary</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This release includes new features. For a list of new features that have been included with this release, please see the CHANGES file inside the source package. Since this is new major release, users are encouraged to do extended testing before upgrading to this version in a production environment.</p>
|
||||
<p>The data in this summary reflects changes that have been made since the previous release, certified-asterisk-13.1.0.</p>
|
||||
<hr/>
|
||||
<a name="contributors"><h2 align="center">Contributors</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were closed by commits that went into this release.</p>
|
||||
<table width="100%" border="0">
|
||||
<tr>
|
||||
<td width="33%"><h3>Coders</h3></td>
|
||||
<td width="33%"><h3>Testers</h3></td>
|
||||
<td width="33%"><h3>Reporters</h3></td>
|
||||
</tr>
|
||||
<tr valign="top">
|
||||
<td>
|
||||
16 bebuild<br/>
|
||||
5 mjordan<br/>
|
||||
1 rmudgett<br/>
|
||||
1 sgriepentrog<br/>
|
||||
</td>
|
||||
<td>
|
||||
</td>
|
||||
<td>
|
||||
6 mjordan<br/>
|
||||
3 rmudgett<br/>
|
||||
2 kharwell<br/>
|
||||
1 mmichelson<br/>
|
||||
1 pnlarsson<br/>
|
||||
</td>
|
||||
</tr>
|
||||
</table>
|
||||
<hr/>
|
||||
<a name="issues"><h2 align="center">Closed Issues</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p>
|
||||
<h3>Category: Channels/chan_pjsip</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24536">ASTERISK-24536</a>: AMI redirect with PJSIP fails to move extra channel<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430007">430007</a><br/>
|
||||
Reporter: pnlarsson<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<h3>Category: Core/Bridging</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24536">ASTERISK-24536</a>: AMI redirect with PJSIP fails to move extra channel<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430007">430007</a><br/>
|
||||
Reporter: pnlarsson<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<h3>Category: Core/Bridging/bridge_basic</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24513">ASTERISK-24513</a>: Local channel apparently leaked in off-nominal DTMF attended transfer<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430245">430245</a><br/>
|
||||
Reporter: mmichelson<br/>
|
||||
Coders: sgriepentrog<br/>
|
||||
<br/>
|
||||
<h3>Category: Core/CodecInterface</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24604">ASTERISK-24604</a>: res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429871">429871</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<h3>Category: Core/General</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24614">ASTERISK-24614</a>: Deadlock when DEBUG_THREADS compiler flag enabled<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429859">429859</a><br/>
|
||||
Reporter: rmudgett<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<h3>Category: Core/ManagerInterface</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24536">ASTERISK-24536</a>: AMI redirect with PJSIP fails to move extra channel<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430007">430007</a><br/>
|
||||
Reporter: pnlarsson<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24553">ASTERISK-24553</a>: ARI/AMI: Include language in standard channel snapshot output<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429891">429891</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<h3>Category: Features</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23841">ASTERISK-23841</a>: DTMF atxfer doesn't set CallerID for the recall calls to the transferrer.<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430046">430046</a><br/>
|
||||
Reporter: rmudgett<br/>
|
||||
Coders: rmudgett<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_ari</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24552">ASTERISK-24552</a>: ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429892">429892</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24553">ASTERISK-24553</a>: ARI/AMI: Include language in standard channel snapshot output<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429891">429891</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_ari_channels</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24552">ASTERISK-24552</a>: ARI: Allow associating a channel as an initiator of an Origination for record keeping purposes<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429892">429892</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_pjsip</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24342">ASTERISK-24342</a>: PJSIP: Qualifying endpoints attempts to do them all at the same time.<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430008">430008</a><br/>
|
||||
Reporter: rmudgett<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24643">ASTERISK-24643</a>: res_pjsip: Add user=phone option<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430085">430085</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: mjordan<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_pjsip_keepalive</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24644">ASTERISK-24644</a>: res_pjsip_keepalive: Add keepalive module for connection-oriented transports.<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430086">430086</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: mjordan<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_pjsip_outbound_registration</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24514">ASTERISK-24514</a>: res_pjsip_outbound_registration: stack overflow when using non-default sorcery wizard<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429866">429866</a><br/>
|
||||
Reporter: kharwell<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_pjsip_session</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24607">ASTERISK-24607</a>: res_pjsip_session: re-INVITE with declined media streams results in 488<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429890">429890</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<h3>Category: Resources/res_rtp_asterisk</h3><br/>
|
||||
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24604">ASTERISK-24604</a>: res_rtp_asterisk: Crash during restart due to race condition in accessing codec in stored ast_frame and codec core<br/>
|
||||
Revision: <a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429871">429871</a><br/>
|
||||
Reporter: mjordan<br/>
|
||||
Coders: bebuild<br/>
|
||||
<br/>
|
||||
<hr/>
|
||||
<a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well.</p>
|
||||
<table width="100%" border="1">
|
||||
<tr><td><b>Revision</b></td><td><b>Author</b></td><td><b>Summary</b></td><td><b>Issues Referenced</b></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429855">429855</a></td><td>bebuild</td><td>Create Certified Asterisk 13.1 branch</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429856">429856</a></td><td>bebuild</td><td>Update properties; remove old summaries</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429858">429858</a></td><td>bebuild</td><td>Activate persistent subscriptions when they are recreated.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429862">429862</a></td><td>bebuild</td><td>Direct Media calls within private network sometimes get one way audio</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429864">429864</a></td><td>bebuild</td><td>res_pjsip_session: Delay sending BYE if a re-INVITE transaction is in progress.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429865">429865</a></td><td>bebuild</td><td>PJSIP: Allow use of 'inactive' streams for hold</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=429869">429869</a></td><td>bebuild</td><td>Prevent potential infinite outbound authentication loops in registration.</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430090">430090</a></td><td>mjordan</td><td>Stasis: Update unittest for channel snapshots</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430093">430093</a></td><td>mjordan</td><td>res_pjsip: Backport missing commits for user_eq_phone</td>
|
||||
<td></td></tr><tr><td><a href="http://svn.digium.com/view/asterisk/certified/branches/13.1?view=revision&revision=430094">430094</a></td><td>mjordan</td><td>res/res_agi: Make Verbose message for 'stream file' match other playbacks</td>
|
||||
<td></td></tr></table>
|
||||
<hr/>
|
||||
<a name="diffstat"><h2 align="center">Diffstat Results</h2></a>
|
||||
<center><a href="#top">[Back to Top]</a></center><br/><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p>
|
||||
<pre>
|
||||
.version | 2
|
||||
CHANGES | 21
|
||||
ChangeLog | 4
|
||||
asterisk-13.1.0-rc2-summary.html | 64 -
|
||||
asterisk-13.1.0-rc2-summary.txt | 95 -
|
||||
channels/chan_dahdi.c | 15
|
||||
channels/chan_pjsip.c | 34
|
||||
channels/chan_sip.c | 4
|
||||
channels/pjsip/dialplan_functions.c | 6
|
||||
configs/samples/pjsip.conf.sample | 3
|
||||
contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py | 30
|
||||
include/asterisk/_private.h | 1
|
||||
include/asterisk/format.h | 11
|
||||
include/asterisk/lock.h | 47
|
||||
include/asterisk/res_pjsip.h | 18
|
||||
include/asterisk/res_pjsip_session.h | 8
|
||||
include/asterisk/rtp_engine.h | 82 +
|
||||
main/asterisk.c | 2
|
||||
main/bridge_basic.c | 118 ++
|
||||
main/codec.c | 5
|
||||
main/format.c | 5
|
||||
main/lock.c | 570 ++++------
|
||||
main/logger.c | 42
|
||||
main/manager.c | 22
|
||||
main/manager_channels.c | 2
|
||||
main/rtp_engine.c | 38
|
||||
main/stasis_channels.c | 5
|
||||
res/ari/ari_model_validators.c | 16
|
||||
res/ari/ari_model_validators.h | 1
|
||||
res/ari/resource_channels.c | 214 +++
|
||||
res/ari/resource_channels.h | 4
|
||||
res/res_agi.c | 5
|
||||
res/res_ari_channels.c | 14
|
||||
res/res_pjsip.c | 56
|
||||
res/res_pjsip/config_global.c | 19
|
||||
res/res_pjsip/pjsip_configuration.c | 1
|
||||
res/res_pjsip/pjsip_options.c | 19
|
||||
res/res_pjsip_caller_id.c | 18
|
||||
res/res_pjsip_keepalive.c | 267 ++++
|
||||
res/res_pjsip_outbound_publish.c | 563 ++++++---
|
||||
res/res_pjsip_outbound_registration.c | 9
|
||||
res/res_pjsip_pubsub.c | 7
|
||||
res/res_pjsip_sdp_rtp.c | 3
|
||||
res/res_pjsip_session.c | 38
|
||||
res/res_pjsip_session.exports.in | 1
|
||||
res/res_rtp_asterisk.c | 3
|
||||
rest-api/api-docs/channels.json | 21
|
||||
tests/test_stasis_channels.c | 2
|
||||
48 files changed, 1671 insertions(+), 864 deletions(-)
|
||||
</pre><br/>
|
||||
<hr/>
|
||||
</body>
|
||||
</html>
|
||||
298
certified-asterisk-13.1-cert1-rc1-summary.txt
Normal file
298
certified-asterisk-13.1-cert1-rc1-summary.txt
Normal file
@@ -0,0 +1,298 @@
|
||||
Release Summary
|
||||
|
||||
certified-asterisk-13.1-cert1-rc1
|
||||
|
||||
Date: 2015-01-06
|
||||
|
||||
<asteriskteam@digium.com>
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Table of Contents
|
||||
|
||||
1. Summary
|
||||
2. Contributors
|
||||
3. Closed Issues
|
||||
4. Other Changes
|
||||
5. Diffstat
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Summary
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This release includes new features. For a list of new features that have
|
||||
been included with this release, please see the CHANGES file inside the
|
||||
source package. Since this is new major release, users are encouraged to
|
||||
do extended testing before upgrading to this version in a production
|
||||
environment.
|
||||
|
||||
The data in this summary reflects changes that have been made since the
|
||||
previous release, certified-asterisk-13.1.0.
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Contributors
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This table lists the people who have submitted code, those that have
|
||||
tested patches, as well as those that reported issues on the issue tracker
|
||||
that were resolved in this release. For coders, the number is how many of
|
||||
their patches (of any size) were committed into this release. For testers,
|
||||
the number is the number of times their name was listed as assisting with
|
||||
testing a patch. Finally, for reporters, the number is the number of
|
||||
issues that they reported that were closed by commits that went into this
|
||||
release.
|
||||
|
||||
Coders Testers Reporters
|
||||
16 bebuild 6 mjordan
|
||||
5 mjordan 3 rmudgett
|
||||
1 rmudgett 2 kharwell
|
||||
1 sgriepentrog 1 mmichelson
|
||||
1 pnlarsson
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Closed Issues
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This is a list of all issues from the issue tracker that were closed by
|
||||
changes that went into this release.
|
||||
|
||||
Category: Channels/chan_pjsip
|
||||
|
||||
ASTERISK-24536: AMI redirect with PJSIP fails to move extra channel
|
||||
Revision: 430007
|
||||
Reporter: pnlarsson
|
||||
Coders: bebuild
|
||||
|
||||
Category: Core/Bridging
|
||||
|
||||
ASTERISK-24536: AMI redirect with PJSIP fails to move extra channel
|
||||
Revision: 430007
|
||||
Reporter: pnlarsson
|
||||
Coders: bebuild
|
||||
|
||||
Category: Core/Bridging/bridge_basic
|
||||
|
||||
ASTERISK-24513: Local channel apparently leaked in off-nominal DTMF
|
||||
attended transfer
|
||||
Revision: 430245
|
||||
Reporter: mmichelson
|
||||
Coders: sgriepentrog
|
||||
|
||||
Category: Core/CodecInterface
|
||||
|
||||
ASTERISK-24604: res_rtp_asterisk: Crash during restart due to race
|
||||
condition in accessing codec in stored ast_frame and codec core
|
||||
Revision: 429871
|
||||
Reporter: mjordan
|
||||
Coders: bebuild
|
||||
|
||||
Category: Core/General
|
||||
|
||||
ASTERISK-24614: Deadlock when DEBUG_THREADS compiler flag enabled
|
||||
Revision: 429859
|
||||
Reporter: rmudgett
|
||||
Coders: bebuild
|
||||
|
||||
Category: Core/ManagerInterface
|
||||
|
||||
ASTERISK-24536: AMI redirect with PJSIP fails to move extra channel
|
||||
Revision: 430007
|
||||
Reporter: pnlarsson
|
||||
Coders: bebuild
|
||||
|
||||
ASTERISK-24553: ARI/AMI: Include language in standard channel snapshot
|
||||
output
|
||||
Revision: 429891
|
||||
Reporter: mjordan
|
||||
Coders: bebuild
|
||||
|
||||
Category: Features
|
||||
|
||||
ASTERISK-23841: DTMF atxfer doesn't set CallerID for the recall calls to
|
||||
the transferrer.
|
||||
Revision: 430046
|
||||
Reporter: rmudgett
|
||||
Coders: rmudgett
|
||||
|
||||
Category: Resources/res_ari
|
||||
|
||||
ASTERISK-24552: ARI: Allow associating a channel as an initiator of an
|
||||
Origination for record keeping purposes
|
||||
Revision: 429892
|
||||
Reporter: mjordan
|
||||
Coders: bebuild
|
||||
|
||||
ASTERISK-24553: ARI/AMI: Include language in standard channel snapshot
|
||||
output
|
||||
Revision: 429891
|
||||
Reporter: mjordan
|
||||
Coders: bebuild
|
||||
|
||||
Category: Resources/res_ari_channels
|
||||
|
||||
ASTERISK-24552: ARI: Allow associating a channel as an initiator of an
|
||||
Origination for record keeping purposes
|
||||
Revision: 429892
|
||||
Reporter: mjordan
|
||||
Coders: bebuild
|
||||
|
||||
Category: Resources/res_pjsip
|
||||
|
||||
ASTERISK-24342: PJSIP: Qualifying endpoints attempts to do them all at the
|
||||
same time.
|
||||
Revision: 430008
|
||||
Reporter: rmudgett
|
||||
Coders: bebuild
|
||||
|
||||
ASTERISK-24643: res_pjsip: Add user=phone option
|
||||
Revision: 430085
|
||||
Reporter: mjordan
|
||||
Coders: mjordan
|
||||
|
||||
Category: Resources/res_pjsip_keepalive
|
||||
|
||||
ASTERISK-24644: res_pjsip_keepalive: Add keepalive module for
|
||||
connection-oriented transports.
|
||||
Revision: 430086
|
||||
Reporter: mjordan
|
||||
Coders: mjordan
|
||||
|
||||
Category: Resources/res_pjsip_outbound_registration
|
||||
|
||||
ASTERISK-24514: res_pjsip_outbound_registration: stack overflow when using
|
||||
non-default sorcery wizard
|
||||
Revision: 429866
|
||||
Reporter: kharwell
|
||||
Coders: bebuild
|
||||
|
||||
Category: Resources/res_pjsip_session
|
||||
|
||||
ASTERISK-24607: res_pjsip_session: re-INVITE with declined media streams
|
||||
results in 488
|
||||
Revision: 429890
|
||||
Reporter: mjordan
|
||||
Coders: bebuild
|
||||
|
||||
Category: Resources/res_rtp_asterisk
|
||||
|
||||
ASTERISK-24604: res_rtp_asterisk: Crash during restart due to race
|
||||
condition in accessing codec in stored ast_frame and codec core
|
||||
Revision: 429871
|
||||
Reporter: mjordan
|
||||
Coders: bebuild
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Commits Not Associated with an Issue
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This is a list of all changes that went into this release that did not
|
||||
directly close an issue from the issue tracker. The commits may have been
|
||||
marked as being related to an issue. If that is the case, the issue
|
||||
numbers are listed here, as well.
|
||||
|
||||
+------------------------------------------------------------------------+
|
||||
| Revision | Author | Summary | Issues |
|
||||
| | | | Referenced |
|
||||
|----------+---------+--------------------------------------+------------|
|
||||
| 429855 | bebuild | Create Certified Asterisk 13.1 | |
|
||||
| | | branch | |
|
||||
|----------+---------+--------------------------------------+------------|
|
||||
| 429856 | bebuild | Update properties; remove old | |
|
||||
| | | summaries | |
|
||||
|----------+---------+--------------------------------------+------------|
|
||||
| 429858 | bebuild | Activate persistent subscriptions | |
|
||||
| | | when they are recreated. | |
|
||||
|----------+---------+--------------------------------------+------------|
|
||||
| 429862 | bebuild | Direct Media calls within private | |
|
||||
| | | network sometimes get one way audio | |
|
||||
|----------+---------+--------------------------------------+------------|
|
||||
| | | res_pjsip_session: Delay sending BYE | |
|
||||
| 429864 | bebuild | if a re-INVITE transaction is in | |
|
||||
| | | progress. | |
|
||||
|----------+---------+--------------------------------------+------------|
|
||||
| 429865 | bebuild | PJSIP: Allow use of 'inactive' | |
|
||||
| | | streams for hold | |
|
||||
|----------+---------+--------------------------------------+------------|
|
||||
| | | Prevent potential infinite outbound | |
|
||||
| 429869 | bebuild | authentication loops in | |
|
||||
| | | registration. | |
|
||||
|----------+---------+--------------------------------------+------------|
|
||||
| 430090 | mjordan | Stasis: Update unittest for channel | |
|
||||
| | | snapshots | |
|
||||
|----------+---------+--------------------------------------+------------|
|
||||
| 430093 | mjordan | res_pjsip: Backport missing commits | |
|
||||
| | | for user_eq_phone | |
|
||||
|----------+---------+--------------------------------------+------------|
|
||||
| | | res/res_agi: Make Verbose message | |
|
||||
| 430094 | mjordan | for 'stream file' match other | |
|
||||
| | | playbacks | |
|
||||
+------------------------------------------------------------------------+
|
||||
|
||||
----------------------------------------------------------------------
|
||||
|
||||
Diffstat Results
|
||||
|
||||
[Back to Top]
|
||||
|
||||
This is a summary of the changes to the source code that went into this
|
||||
release that was generated using the diffstat utility.
|
||||
|
||||
.version | 2
|
||||
CHANGES | 21
|
||||
ChangeLog | 4
|
||||
asterisk-13.1.0-rc2-summary.html | 64 -
|
||||
asterisk-13.1.0-rc2-summary.txt | 95 -
|
||||
channels/chan_dahdi.c | 15
|
||||
channels/chan_pjsip.c | 34
|
||||
channels/chan_sip.c | 4
|
||||
channels/pjsip/dialplan_functions.c | 6
|
||||
configs/samples/pjsip.conf.sample | 3
|
||||
contrib/ast-db-manage/config/versions/371a3bf4143e_add_user_eq_phone_option_to_pjsip.py | 30
|
||||
include/asterisk/_private.h | 1
|
||||
include/asterisk/format.h | 11
|
||||
include/asterisk/lock.h | 47
|
||||
include/asterisk/res_pjsip.h | 18
|
||||
include/asterisk/res_pjsip_session.h | 8
|
||||
include/asterisk/rtp_engine.h | 82 +
|
||||
main/asterisk.c | 2
|
||||
main/bridge_basic.c | 118 ++
|
||||
main/codec.c | 5
|
||||
main/format.c | 5
|
||||
main/lock.c | 570 ++++------
|
||||
main/logger.c | 42
|
||||
main/manager.c | 22
|
||||
main/manager_channels.c | 2
|
||||
main/rtp_engine.c | 38
|
||||
main/stasis_channels.c | 5
|
||||
res/ari/ari_model_validators.c | 16
|
||||
res/ari/ari_model_validators.h | 1
|
||||
res/ari/resource_channels.c | 214 +++
|
||||
res/ari/resource_channels.h | 4
|
||||
res/res_agi.c | 5
|
||||
res/res_ari_channels.c | 14
|
||||
res/res_pjsip.c | 56
|
||||
res/res_pjsip/config_global.c | 19
|
||||
res/res_pjsip/pjsip_configuration.c | 1
|
||||
res/res_pjsip/pjsip_options.c | 19
|
||||
res/res_pjsip_caller_id.c | 18
|
||||
res/res_pjsip_keepalive.c | 267 ++++
|
||||
res/res_pjsip_outbound_publish.c | 563 ++++++---
|
||||
res/res_pjsip_outbound_registration.c | 9
|
||||
res/res_pjsip_pubsub.c | 7
|
||||
res/res_pjsip_sdp_rtp.c | 3
|
||||
res/res_pjsip_session.c | 38
|
||||
res/res_pjsip_session.exports.in | 1
|
||||
res/res_rtp_asterisk.c | 3
|
||||
rest-api/api-docs/channels.json | 21
|
||||
tests/test_stasis_channels.c | 2
|
||||
48 files changed, 1671 insertions(+), 864 deletions(-)
|
||||
|
||||
----------------------------------------------------------------------
|
||||
@@ -8,7 +8,7 @@ Create Date: 2014-10-13 13:46:24.474675
|
||||
|
||||
# revision identifiers, used by Alembic.
|
||||
revision = '371a3bf4143e'
|
||||
down_revision = '10aedae86a32'
|
||||
down_revision = 'eb88a14f2a'
|
||||
|
||||
from alembic import op
|
||||
import sqlalchemy as sa
|
||||
|
||||
@@ -703,3 +703,9 @@ ALTER TABLE ps_endpoints ADD COLUMN media_encryption_optimistic ENUM('yes','no')
|
||||
|
||||
UPDATE alembic_version SET version_num='eb88a14f2a';
|
||||
|
||||
-- Running upgrade eb88a14f2a -> 371a3bf4143e
|
||||
|
||||
ALTER TABLE ps_endpoints ADD COLUMN user_eq_phone ENUM('yes','no');
|
||||
|
||||
UPDATE alembic_version SET version_num='371a3bf4143e';
|
||||
|
||||
|
||||
@@ -984,7 +984,17 @@ ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (media_encryption_opt
|
||||
|
||||
/
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a')
|
||||
-- Running upgrade eb88a14f2a -> 371a3bf4143e
|
||||
|
||||
ALTER TABLE ps_endpoints ADD user_eq_phone VARCHAR(3 CHAR)
|
||||
|
||||
/
|
||||
|
||||
ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (user_eq_phone IN ('yes', 'no'))
|
||||
|
||||
/
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('371a3bf4143e')
|
||||
|
||||
/
|
||||
|
||||
|
||||
@@ -733,7 +733,11 @@ DROP TYPE sip_directmedia_values;
|
||||
|
||||
ALTER TABLE ps_endpoints ADD COLUMN media_encryption_optimistic yesno_values;
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a');
|
||||
-- Running upgrade eb88a14f2a -> 371a3bf4143e
|
||||
|
||||
ALTER TABLE ps_endpoints ADD COLUMN user_eq_phone yesno_values;
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('371a3bf4143e');
|
||||
|
||||
COMMIT;
|
||||
|
||||
|
||||
@@ -982,7 +982,17 @@ ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (media_encryption_opt
|
||||
|
||||
GO
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('eb88a14f2a');
|
||||
-- Running upgrade eb88a14f2a -> 371a3bf4143e
|
||||
|
||||
ALTER TABLE ps_endpoints ADD user_eq_phone VARCHAR(3) NULL;
|
||||
|
||||
GO
|
||||
|
||||
ALTER TABLE ps_endpoints ADD CONSTRAINT yesno_values CHECK (user_eq_phone IN ('yes', 'no'));
|
||||
|
||||
GO
|
||||
|
||||
INSERT INTO alembic_version (version_num) VALUES ('371a3bf4143e');
|
||||
|
||||
GO
|
||||
|
||||
|
||||
Reference in New Issue
Block a user