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2007-11-05Merged revisions 88768 via svnmerge from russell1-2/+8
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: http://svn.digium.com/svn/asterisk/trunk@88769 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05Merged revisions 88765 via svnmerge from russell1-26/+26
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: http://svn.digium.com/svn/asterisk/trunk@88766 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05Merged revisions 88671 via svnmerge from file1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@88673 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05Merged revisions 88585 via svnmerge from qwell1-0/+2
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: http://svn.digium.com/svn/asterisk/trunk@88586 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-05Fix memory leaks and deadlocks in chan_unistim.file1-37/+53
(closes issue #11158) Reported by: eliel Patches: chan_unistim.c.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88510 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-04Simplify the implementation and the API for stringfields;rizzo2-23/+23
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: http://svn.digium.com/svn/asterisk/trunk@88454 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-02fix some issues with crashing on unload, when it didn't completely load cleanlyrussell1-3/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88409 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-02Convert the CLI commands to the new formatrussell1-78/+116
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88408 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-02Merge the code from asterisk/team/group/chan_unistim:russell1-0/+5628
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: http://svn.digium.com/svn/asterisk/trunk@88368 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-02Merged revisions 88366 via svnmerge from file1-18/+18
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: http://svn.digium.com/svn/asterisk/trunk@88367 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-02Merged revisions 88328 via svnmerge from file1-0/+5
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: http://svn.digium.com/svn/asterisk/trunk@88329 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01Remove traces of gnutls, since we no longer use/need it.qwell2-20/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88184 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01Merged revisions 88078 via svnmerge from qwell1-1/+3
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: http://svn.digium.com/svn/asterisk/trunk@88079 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01Change some uses of free() to ast_free(). (No functional differences.)russell1-1/+1
(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: http://svn.digium.com/svn/asterisk/trunk@88077 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01Janitor: use ast_free to pair calls of ast_malloc and ast_calloctilghman1-4/+4
Reported by: eliel Patch by: eliel Closes issue #11135 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88008 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-31Merged revisions 87906 via svnmerge from qwell2-6/+1
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: http://svn.digium.com/svn/asterisk/trunk@87907 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-30Merged revisions 87686 via svnmerge from russell1-38/+96
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: http://svn.digium.com/svn/asterisk/trunk@87687 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-30Merged revisions 87650 via svnmerge from qwell1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@87651 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-29Merged revisions 87342 via svnmerge from file1-1/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87342 | file | 2007-10-29 14:20:28 -0300 (Mon, 29 Oct 2007) | 6 lines Fix issue where if both sides of the dialog cancelled the dialog at the same time chan_sip could kepe retransmitting a response for no reason. (closes issue #9566) Reported by: atca_pres Patches: bug9566.patch uploaded by oej ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87343 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-29Add autoconf checks for extra suppserv definitions that are not present in ↵file1-0/+8
releases yet. chan_misdn should now build against the latest release. (closes issue #11103) Reported by: IgorG git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87325 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-27Add Circuit Group Queury message codemattf1-0/+36
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87232 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-27Make sure we turn on the DSP when we answer the callmattf1-1/+9
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87231 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-26Correctly use defined return values in (some) load_module functions.qwell4-7/+7
(issue #11096) Patches: chan_agent.c.patch uploaded by eliel (license 64) chan_local.c.patch uploaded by eliel (license 64) chan_features.c.patch uploaded by eliel (license 64) chan_zap.c.patch uploaded by eliel (license 64) res_monitor.c.patch uploaded by eliel (license 64) res_realtime.c.patch uploaded by eliel (license 64) res_crypto.c.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87202 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-24Merged revisions 86982 via svnmerge from qwell1-8/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11079) ........ r86982 | qwell | 2007-10-24 15:56:47 -0500 (Wed, 24 Oct 2007) | 5 lines Correctly respect hidecalleridname configuration option. Simplify code slightly in the process. Issue 11079, reported by ddv2005 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86983 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former ↵qwell16-133/+133
didn't make much sense git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86820 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22Merged revisions 86756 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86756 | file | 2007-10-22 13:35:22 -0300 (Mon, 22 Oct 2007) | 4 lines After reading online I have confirmed that Record-Route headers should be copied to 1xx responses as well. (closes issue #10113) Reported by: makoto ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86757 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22There is a really fun game that you can play before committing code,russell1-1/+1
and it's called "make". :) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86749 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22resetinterval defaulting to something other than 'never' doesn't seem to ↵kpfleming1-2/+1
accomplish any good and causes problems for plenty of people... git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86697 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22Merged revisions 86598 via svnmerge from crichter2-15/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86598 | crichter | 2007-10-22 11:21:15 +0200 (Mo, 22 Okt 2007) | 1 line we send DISCONNECT instead of RELEASE/RELEASE_COMPLETE if the dialplan does not match after an overlap call. Also added out_cause=1 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86617 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22started to add some basic support for supplementary services like ↵crichter4-82/+169
CallForwarding and so forth git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86616 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19Add better support for blocking and unblocking of CICs (#10965)mattf1-13/+25
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86549 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19Convert NEW_CLI to AST_CLI.qwell16-133/+133
Closes issue #11039, as suggested by seanbright. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19More changes to NEW_CLI.qwell4-940/+1326
Also fixes a few cli messages and some minor formatting. (closes issue #11001) Reported by: seanbright Patches: newcli.1.patch uploaded by seanbright (license 71) newcli.2.patch uploaded by seanbright (license 71) newcli.4.patch uploaded by seanbright (license 71) newcli.5.patch uploaded by seanbright (license 71) newcli.6.patch uploaded by seanbright (license 71) newcli.7.patch uploaded by seanbright (license 71) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86534 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19Merged revisions 86471 via svnmerge from file1-8/+12
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86471 | file | 2007-10-19 12:33:49 -0300 (Fri, 19 Oct 2007) | 6 lines Fix two issues with domains and transfers. If a port was given in the hostname it was treated as part of the hostname. If domains were configured but external domains were not enabled all transfers would be considered remote. (closes issue #11027) Reported by: ramonpeek Patches: 11027-1.diff uploaded by ramonpeek (license 266) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86472 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19Merged revisions 86469 via svnmerge from file1-2/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86469 | file | 2007-10-19 12:08:12 -0300 (Fri, 19 Oct 2007) | 4 lines Set port number in received as information for registrations as well. (closes issue #11028) Reported by: brad-x ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86470 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-18Fixing a segfault from tab-completing a "zap restart" CLI command.mmichelson1-0/+2
(patch made by seanbright, pointed out in #asterisk-dev on IRC) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86350 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-17Merged revisions 86149 via svnmerge from russell1-1/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86149 | russell | 2007-10-17 12:57:45 -0500 (Wed, 17 Oct 2007) | 8 lines If Asterisk is in the middle of shutting down, respond to OPTIONS with 503 Unavailable. (closes issue #10994) Reported by: eserra Patches: sip-options-503.patch uploaded by eserra (license 45) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86150 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-17Merged revisions 86117 via svnmerge from file1-1/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86117 | file | 2007-10-17 13:58:03 -0300 (Wed, 17 Oct 2007) | 4 lines Whoops, forgot to remove the original sip_scheddestroy. (closes issue #11010) Reported by: vadim ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86118 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-17Allow chan_usbradio to compile again.qwell2-4/+6
Closes issue #11014, patch by seanbright. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86104 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-17Change dependency for chan_usbradio to asound. Let's keep everything uniform.file1-1/+1
(closes issue #11013) Reported by: seanbright git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86067 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-17Merged revisions 86063 via svnmerge from file1-2/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86063 | file | 2007-10-17 12:06:36 -0300 (Wed, 17 Oct 2007) | 4 lines Don't schedule dialog destruction if a MESSAGE is received using an existing dialog. (closes issue #11010) Reported by: vadim ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86064 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-16Don't hangup the call for SS7 if we get an alarmmattf1-14/+20
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85957 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-16This fixes SIP subscriptions in trunk.russell1-1/+0
Don't improperly memset() over an ast_str. This was leftover from before it got changed to use ast_str. (closes issue #11003, reported by pj) (closes issue #10770, reported by yehavi) (patched by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85944 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-16Fix CLI help outputphsultan2-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85787 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-16Added two CLI functions, taken from chan_gtalk :phsultan1-50/+107
- jingle reload ; - jingle show channels. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85778 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-16Make an audio path under the following call configuration :phsultan1-11/+18
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2 Modifications : - set bridge type to partial ; - process media candidates from the remote peer properly. Now we have Jingle audio, at least between two Asterisk Jingle clients. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85777 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-15Switch dundi to new tos config format.qwell5-19/+0
Remove old unused defines for old style. Closes issue 10860, patch by IgorG. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85764 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-15Merged revisions 85604 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r85604 | russell | 2007-10-15 11:54:57 -0500 (Mon, 15 Oct 2007) | 6 lines Make the default for the srvlookup option to be yes. It doesn't really make sense for it to default to off. The default configuration file has it on, and proper RFC behavior, as indicated by a comment in the code, is for it to be on. So, let's have it on by default to make lives easier. (closes issue #10954, suggested by jtodd) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85605 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-15Allow RTP structure registrationphsultan1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85555 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-12Trying to finish the last of the charge_number patch up #10916mattf1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85526 f38db490-d61c-443f-a65b-d21fe96a405b
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