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2007-12-10Removing some LOG_DEBUG itemsOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-10Merged revisions 92158 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92158 | oej | 2007-12-10 15:04:44 +0100 (MÃ¥n, 10 Dec 2007) | 16 lines Avoid reinvite race situations with two Asterisks trying to reinvite each other in 1.4 and trunk. This patch implements support for the 491 error code that Asterisk 1.4 generates on situations where we get an incoming INVITE and already has one in progress. Thanks to mavetju for reporting and to Raj Jain for an excellent explanation of the problem. Patch by myself. Tested with 8 Asterisk servers connected to each other in a training network. Closes issue #10481 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-07Fix a small typo in a comment.Jason Parker
Closes issue #11490 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-06Merged revisions 91439 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4 lines Add support for accepting and sending T.38 in the initial INVITE. (closes issue #9402) Reported by: thdei ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91440 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-05Rename "username" to "defaultuser" to match with "defaultip".Olle Johansson
"Username" still works, but is deprecated. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-05Remove the cseqs from "sip show channel" and make more place for the call ID.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04Add manager action 'sipshowregistry'.Jason Parker
Closes issue #11464, patch by eliel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-30Merged revisions 90269 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6 lines Fix locking issues under one legged replaces scenarios. (closes issue #11420) Reported by: irroot Patches: chan_sip_oneleg.patch uploaded by irroot (license 52) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27remove a duplicate manager eventRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27Starting to merge changes from the "moremanager" branch. Documentation willOlle Johansson
follow. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89702 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-27The following patch with updates for trunk. Works much better in trunk.Olle Johansson
Also by accident fixed a bad typo by a previous committer, which actually made video calls not work fully... Merged revisions 89630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines If we get a codec offer using a well-known payload type, but using it for another codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89698 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26Rename "limitonpeer" to "counteronpeer" since the call-limit is deprecated.Olle Johansson
Both still works in this version. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26Formatting, doxygenificationOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26Formatting changes, cleaning up some codeOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26Start using Doxygen groupings to group variables and defines.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26Instead of printing out one codec in sip show channels print out all of the ↵Joshua Colp
native ones (this is for video). (closes issue #11366) Reported by: ovi git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89573 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-25Typo (someone needs to test compile before committing his changes)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89560 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-25More doxygen changesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-25HousekeepingOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-25Formatting, doxygen updatesOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-25- Deprecate "call-limit" in chan_sip. No other channel driver enforces ↵Olle Johansson
call-limits and we now have the groupcount system to implement call-limits in the dialplan. You can use the "setvar" option in realtime/sip.conf to set limits per device. - Implement "callcounter" as a new option to enable the call counting we need to report device status to queue, manager and SIP subscriptions. The call counter setting is now enabled in the code by setting the device call-limit to 999. When we remove the call limit, we can simply enable this with a boolean setting. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-25Housekeeping...Olle Johansson
- Fix typo in chan_sip - Remove changes to caller ID structure, moving it to branch (russellb) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-23set rtpmap video info according to what is read from SDP;Luigi Rizzo
make the format explicit in a debug message; print the audio instead of aggregated peer capability in a debugging msg. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21closes issue #11285, where an unload of a module that creates a dialplan ↵Steve Murphy
context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21remove another set of redundant #include "asterisk/options.h"Luigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20move asterisk/paths.h outside asterisk.h and into those filesLuigi Rizzo
who really need it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20Fix sip show history.Olle Johansson
Closes issue #11312 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20Change terminology a bit for CLI commands handling SIP ↵Olle Johansson
channels/calls/dialogs/whatever. Closes issue #11312 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89444 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19Changed the "busy-level" option in sip.conf to "busylevel" to be more parallelMark Michelson
with the SIPPEER() argument of the same name. The deprecation procedure is not being used here since this is a trunk-only option. (closes issue #11307, reported by pj, patched by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19Change delimiter of SIPPEER to be comma (instead of pipe) and further ↵Tilghman Lesher
deprecate the old ':' delimiter Reported by: pj Patch by: tilghman Closes issue #11305 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89429 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyLuigi Rizzo
were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19Adding busy-level to the SIP_PEER() dialplan function. Olle Johansson
With this, you can control the peer in the dialplan, so you avoid placing outbound calls when the device has reached busy-level. Reported by pj. Closes bug #11180 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19Make some notes about a problem I found with the OPTIONs handler while ↵Olle Johansson
working with the bug tracker. Since we don't authenticate devices (peers/users) on OPTIONS we don't have the proper context set for the user/peer. However, we might not want to process an authentication for every OPTIONS, so we could have a config option for this, "optionsforceok" to always answer 200 OK on the request and not check device or destination, nor add a SDP. If Asterisk sends the OPTIONs request, it doesn't care about the reply. Some devices use OPTIONs to discover capabilities, since we should answer like an INVITE from the device and we need to support that properly too, which we don't today. So much to do :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-17fix breakage induced by previous mistakeLuigi Rizzo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89382 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16remove redundant #include "asterisk/compat.h",Luigi Rizzo
but make sure that asterisk/compiler.h is included everywhere git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89336 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16Start untangling header inclusion in a way that does not affectLuigi Rizzo
build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15And file said... let trunk build again! Accomplished by some more ↵Joshua Colp
constification, and marking a function in chan_sip as purposely unused until it is fixed up. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15Always relying on the responses when crossing NAT's are not a goodOlle Johansson
solution, it breaks communication. Rizzo - you need to implement a configuration option for this code. It's good, but maybe should be off by default. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89285 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15Merged revisions 89281 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6 lines Don't send re-invites during pending INVITE transactions. Patch by one47 - thanks! Closes issue #9305 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89283 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15Merged revisions 89280 via svnmerge from Olle Johansson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5 lines Improve support for multipart messages. Code by gasparz, changes by me (mostly formatting). Thanks, gasparz! Closes issue #10947 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15Exit early instead of deciding to exit after processing the message.Olle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89279 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15Add support for application/dtmf SIP INFO dtmf handling. Yep, anotherOlle Johansson
way of handling DTMF in SIP. Totally undocumented, but implemented in enough devices so we have to support it. Code by sergee, small changes by oej. Closes issue #11049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-14make the 'name' and 'value' fields in ast_variable const char *Luigi Rizzo
This prevents modifying the strings in the stored variables, and catched a few instances where this was actually done. Given the differences between trunk and 1.4 (and the fact that this is effectively an API change) it is better to fix 1.4 independently. These are chan_sip.c::sip_register() chan_skinny.c:: near line 2847 config.c:: near line 1774 logger.c::make_components() res_adsi.c:: near line 1049 I may have missed some instances for modules that do not build here. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-13- Convert initialization of a struct to C99 style instead of GNU styleRussell Bryant
- Fix a minor spelling error in a comment git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-13Merged revisions 89246 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007) | 2 lines If we set a value for qualify, we should actually pay attention to it, instead of overriding the value ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89247 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12Merged revisions 89184 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007) | 5 lines Fix two cases of memory corruption caused by background threads. Reported by: atis Patch by: tilghman Fixes issue #10923 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08Merged revisions 89119 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov 2007) | 7 lines Rework of the commit I made yesterday to use the already built-in ast_uri_decode function as opposed to my home-rolled one. Also added comments. Thanks to oej for pointing me in the right direction ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08Merged revisions 89101 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4 lines Do not add a sip: to the beginning of the To URI unless needed. (closes issue #10756) Reported by: goestelecom ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08Merged revisions 89099 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6 lines Improve the devicestate logic for multiple devices. If any are available then the extension is considered available. (closes issue #10164) Reported by: nic_bellamy Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08Merged revisions 89097 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8 lines Add support for allowing one outgoing transaction. This means if a response comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support. (closes issue #10946) Reported by: flefoll (closes issue #10915) Reported by: ramonpeek (closes issue #9567) Reported by: atca_pres ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89098 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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