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${ noResults }
20144 Commits (d348c9aa1ea5771b51ad9883420cb41a1fc27b3e)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Tilghman Lesher
|
7e204048fc |
Only report swap on platforms which can examine those statistics
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266337 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Vossel
|
185f73d8b1 |
fixes crash when creation of UDPTL fails
(closes issue #17264) Reported by: falves11 Patches: issue_17264_reviewboard_fix.diff uploaded by dvossel (license 671) issue_17264_1.6.2_reviewboard_fix.diff uploaded by dvossel (license 671) Tested by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Terry Wilson
|
9fdbc5618d |
More build fixes for ical/neon and res_calendar_ews
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266289 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jeff Peeler
|
a370bc1522 |
fix compile error
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266240 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
05a1ec8abc |
Cache query results for one second.
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 |
16 years ago |
|
Tilghman Lesher
|
fb80119b87 |
Merged revisions 266142 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r266142 | tilghman | 2010-05-26 16:11:44 -0500 (Wed, 26 May 2010) | 14 lines Use sigaction for signals which should persist past the initial trigger, not signal. If you call signal() in a Solaris signal handler, instead of just resetting the signal handler, it causes the signal to refire, because the signal is not marked as handled prior to the signal handler being called. This effectively causes Solaris to immediately exceed the threadstack in recursive signal handlers and crash. (closes issue #17000) Reported by: rmcgilvr Patches: 20100526__issue17000.diff.txt uploaded by tilghman (license 14) Tested by: rmcgilvr ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266146 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Mark Michelson
|
70a1bf3142 |
Remove redundant ast_conntected_line_free call.
This wouldn't cause any problems, but it's certainly not needed either. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266098 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
529a87ce7d |
Remove unrelated MOH change from previous commit.
Thanks Kevin! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266094 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
8999372c33 |
Fix misspelling of macro args.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
77a96c5a93 |
do all sip registry parsing before transmit_register
This patch breaks up every part of the sip registry string during config parsing and removes all parsing from transmit_register(). Thanks to Nick_Lewis for contributing this patch! (closes issue #14331) Reported by: Nick_Lewis Patches: chan_sip.c-domparse.patch uploaded by Nick Lewis (license 657) chan_sip.c.patch uploaded by Nick Lewis (license 657) chan_sip.c.domainparse3.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse4.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse5.patch uploaded by Nick Lewis (license 657) nicklewispatch.diff uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel Review: https://reviewboard.asterisk.org/r/628/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
07500098c1 |
fixes failed SIP Directed pickup resulting in dead channel
(closes issue #17339) Reported by: one47 Patches: sip_magic_pickup2 uploaded by one47 (license 23) Tested by: one47, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
7c02ee4f91 |
Blocked revisions 266004 via svnmerge
........ r266004 | mmichelson | 2010-05-26 13:21:10 -0500 (Wed, 26 May 2010) | 12 lines Make AgentComplete message more consistent. At times, the "Member" field was not specified during the event. It's there now. (closes issue #15638) Reported by: elbriga Patches: patchAppQueueAgentComplete.diff uploaded by elbriga (license 482) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266005 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
35025c16d0 |
Merged revisions 265910 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265910 | tilghman | 2010-05-26 11:21:00 -0500 (Wed, 26 May 2010) | 7 lines Not finding rows in the DB does not rise to the level of a warning. (closes issue #17062) Reported by: drookie Patches: 20100525__issue17062.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265923 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
5c9fdd8666 |
Construct socket name, according to the Postgres docs, and document as such.
(closes issue #17392) Reported by: dps Patches: 20100525__issue17392.diff.txt uploaded by tilghman (license 14) Tested by: dps git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265894 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
822423adb8 |
.......
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
83ca84d9bd |
Re-enable "always" option for videosupport option in sip.conf.
(closes issue #17016) Reported by: twilson Patches: 17016.patch uploaded by mmichelson (license 60) Tested by: devmod git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265842 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Terry Wilson
|
f1503b9e1d |
Ensure that libneon > 0.29.0 is installed for res_calendar_ews
This uses a modified version of pabelanger's patch that checks for NTLM support instead, which was added in 0.29.0 which is what is required for res_calendar_ews. (closes issue #17391) Reported by: loloski Patches: issue17391.patch.v2 uploaded by pabelanger (license 224) Tested by: twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265793 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
|
a7498ae02e |
Use configure to determine the prefixes and include directories properly.
This ensures cross-platform compatibility, even among Linux distributions, which don't always put headers in the same place. (closes issue #17391) Reported by: loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265747 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
|
01db62bdbe |
Properly use peer's outboundproxy for outbound REGISTERs.
The logic used in transmit_register to get the outboundproxy for a peer was flawed since this value would be overridden shortly afterwards when create_addr was called. In addition, this also fixes some logic used when parsing users.conf so that the peer name is placed in the internally-generated register string so that an outboundproxy set in the Asterisk GUI will be used for outbound REGISTERs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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170fdad7df |
Blocked revisions 265613 via svnmerge
........ r265613 | dvossel | 2010-05-25 12:11:40 -0500 (Tue, 25 May 2010) | 8 lines fixes build issue with zaptel (closes issue #17394) Reported by: aragon Patches: half_buffer_fix.diff uploaded by dvossel (license 671) Tested by: aragon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265614 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthew Nicholson
|
9ed82007f1 |
Merged revisions 265610 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670. (closes issue #17334) Reported by: jvandal Patches: queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested by: aragon, jvandal ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265611 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Richard Mudgett
|
838ce15e20 |
Memory leak in connected line data when SIP blond transfer done.
The handling of the control subclass AST_CONTROL_READ_ACTION frame leaked connected line string memory in __ast_read(). Also in __ast_read() the frame type switch should not have had a case for AST_CONTROL_READ_ACTION. AST_CONTROL_READ_ACTION is not a frame type. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265608 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Tzafrir Cohen
|
f14e56249a |
Typos: 'succesful' (lintian)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265525 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Terry Wilson
|
0390dae08d |
Merge the rest of the FullyBooted patch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265467 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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cba378d847 |
Allow SendDTMF to play digits to a specified channel.
Patch supplied by reporter was modified to use autoservice and prevent a potential channel ref leak but is otherwise as the reporter uploaded it. (closes issue #17182) Reported by: rcasas Patches: app_senddtmf.c.patch_trunk uploaded by rcasas (license 641) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265453 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
c5be9d2247 |
Print openh323 log to the Asterisk console.
(closes issue #17109) Reported by: under Patches: logstream.diff uploaded by under (license 914) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265451 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
|
f908dd984c |
Allow type=user SIP endpoints to be loaded properly from realtime.
(closes issue #16021) Reported by: Guggemand Patches: realtime-type-fix.patch uploaded by Guggemand (license 897) (altered by me slightly to avoid ref leaks) Tested by: Guggemand git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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4e38beb960 |
Make app_rpt.c able to compile again.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265367 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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David Vossel
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94de6f3cb8 |
reverses incorrect logic introduced by r243200
The decoding of the replace_id did not need to be broken up in this instance. This was brought to my attention again because it caused a segfault when the from or to tags were not present in the "Replaces" header. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265366 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Terry Wilson
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3c9a8ebadb |
Add the FullyBooted AMI event
It is possible to connect to the manager interface before all Asterisk modules are loaded. To ensure that an application does not send AMI actions that might require a module that has not yet loaded, the application can listen for the FullyBooted manager event. It will be sent upon connection if all modules have been loaded, or as soon as loading is complete. The event: Event: FullyBooted Privilege: system,all Status: Fully Booted Review: https://reviewboard.asterisk.org/r/639/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265320 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Terry Wilson
|
880cde12ac |
Calendaring support for Exchange Server 2007+ via EWS
This commit adds support for calendaring with Exchange Server 2007+ via Exchange Web Services. Full write support and for querying attendees. Many thanks to Jan Kaláb for the feature. (closes issue #17022) Reported by: pitel Patches: res_calendar_ews.c uploaded by pitel (license 1008) Tested by: pitel, twilson Review: https://reviewboard.asterisk.org/r/557/ Review: https://reviewboard.asterisk.org/r/668/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
6f998f06af |
On systems with a LOT of RAM, a signed integer sometimes printed negative.
(closes issue #16837) Reported by: jlpedrosa Patches: 20100504__issue16837.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265316 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
David Vossel
|
fdb698ca2b |
fixes segfault when using generic plc
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265273 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Alexandr Anikin
|
45895dc320 |
small changes to avoiding 'freeing unused memory...'
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
|
ba8e183938 |
Channel initialization failure causes crashes.
__ast_channel_alloc_ap() has several points in the initialization of a new channel structure where it could fail. Since the channel structure is now an ao2 object, the destructor callback needs to be able to handle clean up when the structure setup is incomplete. Problems corrected: 1) Failing to setup the alertpipe would not unreference the structure but free it directly. Doing this to an ao2_object is very bad. 2) File descriptors need to be initialized to -1 before a construction failure could occur so the destructor will not close unopened descriptors. 3) The destructor needs to check that the string field has been initialized before using any string field values. Crashes expected. 4) The destructor should not notify devstate if the device name is empty. It is a waste of cycles and a couple ERROR log messages are generated. Review: https://reviewboard.asterisk.org/r/675/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265174 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
1225ee831c |
Merged revisions 265089 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines Don't hang up on a queue caller if the file we attempt to play does not exist. This also fixes a documentation mistake in file.h that made my original attempt to correct this problem not work correctly. (closes issue #17061) Reported by: RoadKill ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
ee60cb8d41 |
Be sure to set the sin_family on the proxy when allocating.
(closes issue #17157) Reported by: stuarth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265087 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
f30952c577 |
Merged revisions 264999 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264999 | mmichelson | 2010-05-21 11:53:53 -0500 (Fri, 21 May 2010) | 3 lines Fix grammatical error in comment. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265000 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Mark Michelson
|
73e8c7572e |
Merged revisions 264996 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264996 | mmichelson | 2010-05-21 11:28:34 -0500 (Fri, 21 May 2010) | 32 lines Allow ast_safe_sleep to defer specific frames until after the sleep has concluded. From reviewboard Background: A Digium customer discovered a somewhat odd bug. The setup is that parties A and B are bridged, and party A places party B on hold. While party B is listening to hold music, he mashes a bunch of DTMF. Party A takes party B off hold while this is happening, but party B continues to hear hold music. I could reproduce this about 1 in 5 times. The issue: When DTMF features are enabled and a user presses keys, the channel that the DTMF is streamed to is placed in an ast_safe_sleep for 100 ms, the duration of the emulated tone. If an AST_CONTROL_UNHOLD frame is read from the channel during the sleep, the frame is dropped. Thus the unhold indication is never made to the channel that was originally placed on hold. The fix: Originally, I discussed with Kevin possible ways of fixing the specific problem reported. However, we determined that the same type of problem could happen in other situations where ast_safe_sleep() is used. Using autoservice as a model, I modified ast_safe_sleep_conditional() to defer specific frame types so they can be re-queued once the sleep has finished. I made a common function for determining if a frame should be deferred so that there are not two identical switch blocks to maintain. Review: https://reviewboard.asterisk.org/r/674/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264997 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
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0a63e3fa10 |
Log spandsp's fax debug output to the FAX logger level.
Review: https://reviewboard.asterisk.org/r/658 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264953 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Terry Wilson
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fe9c315171 |
Take dup'd code for directmedia ACLs and make utility func
The same code was repeated in lots of different places, so I made a utility fuction for it. This should make the merge in the v6-new branch easier. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
|
43991ce806 |
Merged revisions 264820 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r264820 | rmudgett | 2010-05-20 18:23:21 -0500 (Thu, 20 May 2010) | 6 lines ast_callerid_parse() had a path that left name uninitialized. Several callers of ast_callerid_parse() do not initialize the name parameter before calling thus there is the potential to use an uninitialized pointer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264828 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
815d7bfe44 |
Let ExtensionState resolve dynamic hints.
(closes issue #16623) Reported by: tilghman Patches: 20100116__issue16623.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264779 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Tilghman Lesher
|
a5bee137f9 |
Error message fix.
(closes issue #17356) Reported by: kenner Patches: app_stack.c.diff uploaded by kenner (license 1040) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264752 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
|
dafb48fe09 |
Avoid crash in generic CC agent init if caller name or number is NULL.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264711 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
|
3d1f005fed |
Dial and queue connected line update macro not always run when expected.
The connected line update macro would not get run if the connected line number string was empty. The number could be empty if the connected line update did not update a number but the name. It should be run if there was an AST_CONTROL_CONNECTED_LINE frame received for pending dials and queues. Renamed and added some more comments for some confusing identifiers directly connected to the related code. Also fixed a memory leak in app_queue. Review: https://reviewboard.asterisk.org/r/669/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264669 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Terry Wilson
|
c7303d840e |
Add support for direct media ACLs
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 |
16 years ago |
Kevin P. Fleming
|
25b6ac55d8 |
Ignore pre-processed source files generated during DONT_OPTIMIZE dev-mode builds.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264540 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Kevin P. Fleming
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2aa0c11679 |
Correct 'all logger levels' patch to work properly.
Nick Lewis pointed out that the patch as committed wouldn't actually include dynamic logger levels, which was missed by the other reviewers. Thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Mark Michelson
|
6bb45831eb |
Fix transcode_via_sln option with SIP calls and improve PLC usage.
From reviewboard: The problem here is a bit complex, so try to bear with me... It was noticed by a Digium customer that generic PLC (as configured in codecs.conf) did not appear to actually be having any sort of benefit when packet loss was introduced on an RTP stream. I reproduced this issue myself by streaming a file across an RTP stream and dropping approx. 5% of the RTP packets. I saw no real difference between when PLC was enabled or disabled when using wireshark to analyze the RTP streams. After analyzing what was going on, it became clear that one of the problems faced was that when running my tests, the translation paths were being set up in such a way that PLC could not possibly work as expected. To illustrate, if packets are lost on channel A's read stream, then we expect that PLC will be applied to channel B's write stream. The problem is that generic PLC can only be done when there is a translation path that moves from some codec to SLINEAR. When I would run my tests, I found that every single time, read and write translation paths would be set up on channel A instead of channel B. There appeared to be no real way to predict which channel the translation paths would be set up on. This is where Kevin swooped in to let me know about the transcode_via_sln option in asterisk.conf. It is supposed to work by placing a read translation path on both channels from the channel's rawreadformat to SLINEAR. It also will place a write translation path on both channels from SLINEAR to the channel's rawwriteformat. Using this option allows one to predictably set up translation paths on all channels. There are two problems with this, though. First and foremost, the transcode_via_sln option did not appear to be working properly when I was placing a SIP call between two endpoints which did not share any common formats. Second, even if this option were to work, for PLC to be applied, there had to be a write translation path that would go from some format to SLINEAR. It would not work properly if the starting format of translation was SLINEAR. The one-line change presented in this review request in chan_sip.c fixed the first issue for me. The problem was that in sip_request_call, the jointcapability of the outbound channel was being set to the format passed to sip_request_call. This is nativeformats of the inbound channel. Because of this, when ast_channel_make_compatible was called by app_dial, both channels already had compatibly read and write formats. Thus, no translation path was set up at the time. My change is to set the jointcapability of the sip_pvt created during sip_request_call to the intersection of the inbound channel's nativeformats and the configured peer capability that we determined during the earlier call to create_addr. Doing this got the translation paths set up as expected when using transcode_via_sln. The changes presented in channel.c fixed the second issue for me. First and foremost, when Asterisk is started, we'll read codecs.conf to see the value of the genericplc option. If this option is set, and ast_write is called for a frame with no data, then we will attempt to fill in the missing samples for the frame. The implementation uses a channel datastore for maintaining the PLC state and for creating a buffer to store PLC samples in. Even when we receive a frame with data, we'll call plc_rx so that the PLC state will have knowledge of the previous voice frame, which it can use as a basis for when it comes time to actually do a PLC fill-in. So, reviewers, now I ask for your help. First off, there's the one line change in chan_sip that I have put in. Is it right? By my logic it seems correct, but I'm sure someone can tell me why it is not going to work. This is probably the change I'm least concerned about, though. What concerns me much more is the set of changes in channel.c. First off, am I even doing it right? When I run tests, I can clearly see that when PLC is activated, I see a significant increase in RTP traffic where I would expect it to be. However, in my humble opinion, the audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to me than when no PLC is used at all. I need someone to review the logic I have used to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm sure someone can point out somewhere where I've done something incorrectly. As I was writing this review request up, I decided to give the code a test run under valgrind, and I find that for some reason, calls to plc_rx are causing some invalid reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around a bit to see why that is the case. If it's obvious to someone reviewing, speak up! Finally, I have one other proposal that is not reflected in my code review. Since without transcode_via_sln set, one cannot predict or control where a translation path will be up, it seems to me that the current practice of using PLC only when transcoding to SLINEAR is not useful. I recommend that once it has been determined that the method used in this code review is correct and works as expected, then the code in translate.c that invokes PLC should be removed. Review: https://reviewboard.asterisk.org/r/622/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |