Updated the UPGRADE.txt and CHANGES file stating that CDR records will not be explicity
written unless cdr.conf exists and is configured.
(closes issue #17373)
Reported by: wdoekes
Tested by: pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Add the ability to announce a call to an endpoint when there are no B
channels available. A call waiting call is a SETUP message with no B
channel selected.
Relevant specification: EN 300 056, EN 300 057, EN 300 058
For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
* Returns "0" if there is a B channel associated with the call.
* Returns "1" if no B channel is associated with the call. The call is
either on hold or is a call waiting call.
If you are going to allow incoming call waiting calls then you need to use
CHANNEL(no_media_path) do determine if you must drop a call to accept the
new call.
Review: https://reviewboard.asterisk.org/r/568/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added ability to send and receive ETSI Explicit Call Transfer (ECT)
messages to eliminate tromboned calls.
Note: Asterisk already supported initiating the transfer of calls to
eliminate tromboned calls to libpri so there was nothing to do for the
asterisk portion.
Review: https://reviewboard.asterisk.org/r/520/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Queries from the PBX core come in 3's. Caching avoids the additional
performance penalty from those two additional queries hitting the database.
(closes issue #16521)
Reported by: tilghman
Patches:
20091229__issue16521.diff.txt uploaded by tilghman (license 14)
Tested by: Hubguru, tilghman
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266238 65c4cc65-6c06-0410-ace0-fbb531ad65f3
directmediapermit/directmediadeny support to restrict which peers can do
directmedia based on ip address. In some networks not all phones are fully
routed, i.e. not all phones can ping each other. This patch adds a way to
restrict directmedia for certain peers between certain networks.
(closes issue #16645)
Reported by: raarts
Patches:
directmediapermit.patch uploaded by raarts (license 937)
Tested by: raarts
Review: https://reviewboard.asterisk.org/r/467/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Added the keyword 'all' to the 'channel hangup request' CLI command
so that you can request all channels to be hungup without having to
restart Asterisk.
(closes issue #16009)
Reported by: moy
Patches:
hangup-all-rev-221688.patch uploaded by moy (license 222)
Tested by: moy, russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r263769 | jpeeler | 2010-05-18 13:54:58 -0500 (Tue, 18 May 2010) | 10 lines
Modify directory name reading to be interrupted with operator or pound escape.
In the case of accidentally entering the wrong first three letters for the
reading, users could be very frustrated if the name listing is very long. This
allows interrupting the reading by pressing 0 or #. 0 will attempt to execute
a configured operator (o) extension and # will exit and proceed in the
dialplan.
ABE-2200
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the following in-conference admin DTMF features:
*81 - Roll call (or simply user count if INTROUSER isn't enabled)
*82 - Eject all non-admins
*83 - Mute/unmute all non-admins
*84 - Start recording the conference on the fly
FWIW, this code uses newly recorded prompts.
(closes issue #16379)
Reported by: rfinnie
Patches:
meetme-enhancements-232771-v1.patch uploaded by rfinnie (license 940)
modified slightly by me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@260757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
From Review Board:
There are two interrelated changes here.
First, there is the introduction of func_srv. This adds two new read-only
dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the
ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV
records instead. In order to facilitate this work, I added a couple of new API
calls to srv.h. ast_srv_get_record_count tells the number of records returned
by an SRV lookup. This number is calculated at the time of the SRV lookup.
ast_srv_get_nth_record allows one to get a numbered SRV record.
Second, there is the modification to chan_sip that allows one to specify a
hostname or IP address (along with a port) to send an outgoing INVITE to when
dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV
records and then use the host and port from the results to dial via a specific
host instead of what is configured in sip.conf.
Review: https://reviewboard.asterisk.org/r/608
SWP-1200
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously only configurable globally. A unit test has also been written to
provide protection against parse failures for supported mailbox options.
(closes issue #16864)
Reported by: kobaz
Patches:
voicemail2.patch uploaded by kobaz (license 834)
Review: https://reviewboard.asterisk.org/r/555/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
users expect them to work.
'core set debug' and 'core set verbose' can optionally change the
level for a specific filename; however, this is actually for a
specific source file name, not the module that source file is included
in. With examples like chan_sip, chan_iax2, chan_misdn and others
consisting of multiple source files, this will not lead to the
behavior that users expect. If they want to set the debug level for
chan_sip, they want it set for all of chan_sip, and not to have to
also set it for reqresp_parser and other files that comprise the
chan_sip module.
This patch changes this functionality to be module-name based instead
of file-name based.
To make this work, some Makefile modifications were required to ensure
that the AST_MODULE definition is present in each object file produced
for each module as well.
Review: https://reviewboard.asterisk.org/r/574/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The PITCH_SHIFT function can be used on a channel to independently
modify the pitch of both rx and tx audio streams. Now you can
improve your conference calls by assigning a random pitch effect
to everyone entering a meetme room, or just make your day more
interesting by making your co-workers sound funny. These are just
some of the numerious practical uses for this function. Enjoy!
https://reviewboard.asterisk.org/r/526/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
New config parameter "reportalarms" added in chan_dahdi.conf which supports the
following possible values:
"channels": report each channel alarms (current behavior, default for backward compatibility)
"spans": report an "SpanAlarm" event when the span of any configured channel is alarmed
"all": report channel and span alarms (aggregated behavior)
"none": do not report any alarms
(closes issue #16709)
Reported by: nahuelgreco
Patches:
chan_dahdi.c.reportalarms.patch uploaded by nahuelgreco (license 162)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The 'p' option allows the PickupChan app to pickup
a ringing phone by looking for the first match to a
partial channel name rather than requiring a full match.
(closes issue #16613)
Reported by: syspert
Patches:
pickipbycallid.patch uploaded by syspert (license 938)
pickupbycallerid_v2.patch uploaded by dvossel (license 671)
Tested by: dvossel, syspert
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
VMSayName that will play the recorded name of the voicemail user if it exists,
otherwise will play the mailbox number. A unit test has been written to verify
correct functionality called test_voicemail_vmsayname.
(closes issue #14973)
Reported by: ghjm
Review: https://reviewboard.asterisk.org/r/530/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The new JabberStatus event gives a concise view of the status change to the AMI
clients. Thanks fiddur!
(closes issue #16760)
Reported by: fiddur
Patches:
244498.2.diff uploaded by fiddur (license 678)
Tested by: fiddur, phsultan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247500 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Current support for regex matching was previously only available on the group.
Also, error reporting for regex failures has been added. In addition to this
feature enhancement a unit test has been written to check the regular expression
logic to ensure the count operation is working as expected.
(closes issue #16642)
Reported by: kobaz
Patches:
groupmatch2.patch uploaded by kobaz (license 834)
Review: https://reviewboard.asterisk.org/r/503/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This feature allows for parkinglots to be created dynamically within
the dialplan. Thanks to all who were involved with getting this patch
written and tested!
(closes issue #15135)
Reported by: IgorG
Patches:
features.dynamic_park.v3.diff uploaded by IgorG (license 20)
2009090400_dynamicpark.diff.txt uploaded by mvanbaak (license 7)
dynamic_parkinglot.diff uploaded by dvossel (license 671)
Tested by: eliel, IgorG, acunningham, mvanbaak, zktech
Review: https://reviewboard.asterisk.org/r/352/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247248 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The options are: parkedcallparking, parkedcallhangup, parkedcallrecording, and
parkedcalltransfers. Previously these options were only available for the
default parking lot.
(closes issue #16641)
Reported by: bluecrow76
Patches:
asterisk-1.6.2.1-features.c.diff uploaded by bluecrow76 (license 270)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244598 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Previously, we would parse GOSUB_RESULT, but not actually do anything with it.
Also, allow GOSUB_RETVAL to be inherited back across a peer/master channel.
(closes issue #16687)
Reported by: bklang
Patches:
app_dial-preserve-gosub_retval.patch uploaded by bklang (license 919)
(with modifications)
(closes issue #16686)
Reported by: bklang
Patches:
app_dial-respect-gosub_result.patch uploaded by bklang (license 919)
(with modifications)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@244393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In asterisk.conf, transmit_silence_during_record has been removed
in favor of using only the transmit_silence option. The
transmit_silence_during_record option remains a valid option in
asterisk.conf, but has been removed from the sample config and
noted in CHANGES.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Part of the work done for connected line was to add an optional
argument to the 'f' option to allow for the connected party information
of the outgoing channel to be set to the argument provided. This was
overlooked during the merge of the work to trunk and is being added
back now. The CHANGES file has also been updated to note this change.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The option is global and currently the acceptable values as noted in the sample
config are accept or deny.
(closes issue #15228)
Reported by: lp0
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
