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2007-11-12Doxygen fixes.Jason Parker
Also fix a common typo I kept seeing (arguement) in various files. Closes issue #11222, patch by snuffy (with arguement > argument by me). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89202 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
2007-11-12Merged revisions 89173 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | 1 line if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89179 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12Merged revisions 89172 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) | 1 line added restart all interfaces Restart_Indicator, to automatically send a RESTART after the L2 of a PTP Port comes up. Also fixed some places where we have send a RELEASE without need for it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89178 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12Fix building on FreeBSD by including/not including some headers.Joshua Colp
(closes issue #11218) Reported by: ys Patches: trunk89169.diff uploaded by ys (license 281) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12Merged revisions 89171 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) | 1 line fixed a state/event issue with overlapdial=yes when no extension matched. removed the general sending of a RELEASE_COMPLETE when we receive a RELEASE, this is done by mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 (when using the NT mode at all) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12Merged revisions 89170 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89170 | crichter | 2007-11-12 10:57:23 +0100 (Mo, 12 Nov 2007) | 1 line fixed the support for CW and therefore for the reject_cause option. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12Merged revisions 89169 via svnmerge from Christian Richter
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | 1 line aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89174 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
2007-11-08convert this code to a more efficient idiomKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08Fix missed conversion to linkedlists macro changeTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89113 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08initialize a variable to silence compiler.Luigi Rizzo
The type of warnings emitted depends on the optimization level, at the lower levels the compiler doesn't always understand what the programmer has in mind. In this case I could not understand it either. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08improve linked-list macros in two ways:Kevin P. Fleming
- the *_CURRENT macros no longer need the list head pointer argument - add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89106 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
2007-11-07Merged revisions 89095 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4 lines If callerid is configured in sip.conf use that for checking the presence of an extension in the dialplan. (closes issue #11185) Reported by: spditner ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89096 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-07Merged revisions 89090 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov 2007) | 6 lines This patch makes it possible for SIP phones to dial extensions defined with '#' characters in extensions.conf AND maintain their escaped characters when forming URI's (closes issue #10681, reported by cahen, patched by me, code review by file) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-07Minor change so chan_h323 builds again.Joshua Colp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-07Provide the ability to directly manipulate the TON/NPI bits in the dialstring.Tilghman Lesher
Reported by: thetatag Patch by: thetatag/stevens/tilghman Closes issue #5331 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Commit some cleanups to the format type code.Tilghman Lesher
- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits. - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution. (This doesn't affect anything immediately, until another codec has wb support.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Add some more locking as well as API update for libss7 for new transport typesMatthew Fredrickson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06"show application <foo>" changes for clarity.Mark Michelson
(closes issue #11171, reported and patched by blitzrage) Many thanks! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Allow gtalk and jingle to use TLS connections again.Jason Parker
Closes issue #9972 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Merged revisions 89032 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 lines Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable. (closes issue #11006) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Merged revisions 88994 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6 lines Fix improbable but possible memory leaks in chan_zap. (closes issue #11166) Reported by: eliel Patches: chan_zap.c.patch uploaded by eliel (license 64) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Update chan_agent documentation. Change a | to , as that is now the required ↵Joshua Colp
way. (closes issue #11167) Reported by: eliel Patches: chan_agent.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88974 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Set up detection of IP_PKTINFO in autoconf for chan_unistimTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06convert uses of LOG_DEBUG to use ast_debug()Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88937 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Add jitterbuffer support to chan_unistim.Russell Bryant
(closes issue #11168) Reported by: IgorG Patches: unistimjb-88863-1.patch uploaded by IgorG (license 20) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06Merged revisions 88805 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | 12 lines After seeing crashes related to channel variables, I went looking around at the ways that channel variables are handled. In general, they were not handled in a thread-safe way. The channel _must_ be locked when reading or writing from/to the channel variable list. What I have done to improve this situation is to make pbx_builtin_setvar_helper() and friends lock the channel when doing their thing. Asterisk API calls almost all lock the channel for you as necessary, but this family of functions did not. (closes issue #10923, reported by atis) (closes issue #11159, reported by 850t) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06explain that the host environment must be used to build gentone;Luigi Rizzo
Remove unset variables, they would be misleading. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05Merged revisions 88768 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) | 8 lines When traversing the list of channel variables here in transmit_invite(), the asterisk channel must be locked, as this data may change at any time. (I have seen numerous reports of crashes related to the handling of channel variables. There are a couple of issues on the bug tracker related to it, but it has also been noted on IRC and mailing lists. So, I am finding and fixing some places where channel variables are handled improperly.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05Merged revisions 88765 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) | 2 lines Fix up some indentation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88766 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05Merged revisions 88671 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7 lines If a SIP channel is put on hold multiple times do not keep incrementing the onHold value. (closes issue #11085) Reported by: francesco_r Tested by: blitzrage (closes issue #10474) Reported by: acennami ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05Merged revisions 88585 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11163) ........ r88585 | qwell | 2007-11-05 11:19:41 -0600 (Mon, 05 Nov 2007) | 4 lines Make sure we destroy the config structure on configuration failure. Issue 11163, patch by eliel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88586 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05Fix memory leaks and deadlocks in chan_unistim.Joshua Colp
(closes issue #11158) Reported by: eliel Patches: chan_unistim.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88510 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-04Simplify the implementation and the API for stringfields;Luigi Rizzo
details and examples are in include/asterisk/stringfields.h. Not applicable to older branches except for 1.4 which will receive a fix for the routines that free memory pools. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02fix some issues with crashing on unload, when it didn't completely load cleanlyRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02Convert the CLI commands to the new formatRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02Merge the code from asterisk/team/group/chan_unistim:Russell Bryant
This introduces a new channel driver, chan_unistim, that supports the Unistim VoIP protocol for Nortel phones. The following models have been confirmed to work: i2002, i2004 and i2050. (closes issue #8864) Reported by: c_hans Patches: chan_unistim.patch uploaded by c (license 304) ustm_no_conf.diff uploaded by junky (license 177) Tested by: c_hans, dbowerman, math, junky, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02Merged revisions 88366 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4 lines Make subscribecontext behave as advertised. It will now look for the presence of a hint in the given context (be it subscribecontext or context). (closes issue #10702) Reported by: slavon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-02Merged revisions 88328 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6 lines If an INFO request within a dialog is received with a content length of 0 simply send back a 200 OK. It is valid to do this and the remote side is probably using it to make sure the signalling is still alive. (closes issue #5747) Reported by: chandi Patches: infofix-81430-1.patch uploaded by IgorG (license 20) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-01Remove traces of gnutls, since we no longer use/need it.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-01Merged revisions 88078 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88078 | qwell | 2007-11-01 11:21:22 -0500 (Thu, 01 Nov 2007) | 4 lines Make sure we set the poll fds to NULL after free()ing it. Part of issue 11017, patch by tzafrir. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-01Change some uses of free() to ast_free(). (No functional differences.)Russell Bryant
(closes issue #11138) Reported by: eliel Patches: pbx_dundi.c.patch uploaded by eliel (license 64) chan_sip.c.patch uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-01Janitor: use ast_free to pair calls of ast_malloc and ast_callocTilghman Lesher
Reported by: eliel Patch by: eliel Closes issue #11135 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-31Merged revisions 87906 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11130) (closes issue #11132) ........ r87906 | qwell | 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines Don't try to allocate memory that we're just going to re-allocate later anyways. Issues 11130 and 11132, patch by eliel. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-30Merged revisions 87686 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87686 | russell | 2007-10-30 16:19:09 -0500 (Tue, 30 Oct 2007) | 11 lines Merge the changes from team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a race condition related to the handling of POKEing peers. Essentially, a reference to a peer is held by the scheduler when there are pending callbacks, but the reference count didn't reflect it. So, it was possible for a peer to hit a reference count of zero and have its destructor begin to be called at the same time that the scheduler thread ran a POKE related callback. If that happened, a crash would likely occur. (closes issue #11082, closes issue #11094) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-30Merged revisions 87650 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87650 | qwell | 2007-10-30 15:29:41 -0500 (Tue, 30 Oct 2007) | 1 line Only try to clean out h323/ if the h323/Makefile exists. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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