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2009-06-10Blocked revisions 199958 via svnmergemmichelson0-0/+0
........ r199958 | mmichelson | 2009-06-10 15:15:48 -0500 (Wed, 10 Jun 2009) | 6 lines Only try to use the invite_branch on outgoing INVITEs with auth credentials. I have added a comment to the code to help ease understanding of the logic here as well. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199967 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-10Merged revisions 199857 via svnmerge from seanbright1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines __WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199858 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-09Merged revisions 199818 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r199818 | dvossel | 2009-06-09 15:47:57 -0500 (Tue, 09 Jun 2009) | 11 lines CLI NOTIFY sending wrong transport type. SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer. (closes issue #15283) Reported by: jthurman Patches: sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614) Tested by: jthurman, dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199821 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-09Blocked revisions 199743 via svnmergedvossel0-0/+0
........ r199743 | dvossel | 2009-06-09 11:22:04 -0500 (Tue, 09 Jun 2009) | 11 lines module load priority This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized. The lower the value, the higher the priority. The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set. If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority on load. Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty. (closes issue #15191) Reported by: alecdavis Tested by: dvossel Review: https://reviewboard.asterisk.org/r/262/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199746 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-08Merged revisions 199630 via svnmerge from seanbright1-4/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines Increase the size of our thread stack on 64 bit processors. We were setting the stack size for each thread to 240KB regardless of architecture, which meant that in some scenarios we actually had less available stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we calculate the stack size we reserve based on the platform's __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 bit -> 1008KB (that's right, we're ready for 128 bit processors) Patch typed by me but written by several members of #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes issue #14932) Reported by: jpiszcz Patches: 06052009_issue14932.patch uploaded by seanbright (license 71) Tested by: seanbright ........ r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the stack size calculation just introduced. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199632 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-08Blocked revisions 199588 via svnmergemmichelson0-0/+0
........ r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (Mon, 08 Jun 2009) | 9 lines Fix a deadlock that could occur when setting rtp stats on SIP calls. (closes issue #15143) Reported by: cristiandimache Patches: 15143.patch uploaded by mmichelson (license 60) Tested by: cristiandimache ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199589 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-05Merged revisions 199298 via svnmerge from dvossel1-19/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r199298 | dvossel | 2009-06-05 16:21:22 -0500 (Fri, 05 Jun 2009) | 21 lines Merged revisions 199297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199301 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-05Merged revisions 199227 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (Fri, 05 Jun 2009) | 14 lines Correct "dahdi show channels" output when specifying a group. Since a DAHDI channel may belong to multiple groups, we need to use a bitwise and instead of equivalence to determine whether to display the channel information. (closes issue #15248) Reported by: gentian Patches: 15248.patch uploaded by mmichelson (license 60) Tested by: gentian ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199228 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-04Merged revisions 199139 via svnmerge from dvossel1-1/+17
https://origsvn.digium.com/svn/asterisk/trunk ................ r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines Merged revisions 199138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199142 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-04Merged revisions 199051 via svnmerge from seanbright3-0/+96
https://origsvn.digium.com/svn/asterisk/trunk ................ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199052 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-03Blocked revisions 198958 via svnmergeseanbright0-0/+0
................ r198958 | seanbright | 2009-06-03 16:49:11 -0400 (Wed, 03 Jun 2009) | 17 lines Blocked revisions 198957 via svnmerge ........ r198957 | seanbright | 2009-06-03 16:39:10 -0400 (Wed, 03 Jun 2009) | 11 lines Fix a possible crash in pbx_spool. We were trying to reference members of a struct that had previously been freed. This patch makes sure that we free the struct after it has been removed from the spooler queue. (closes issue #15072) Reported by: garlew Patches: spool.diff uploaded by garlew (license 376) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198959 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-03Merged revisions 198856 via svnmerge from dvossel3-6/+111
https://origsvn.digium.com/svn/asterisk/trunk ........ r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines Generic call forward api, ast_call_forward() The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198889 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-02Merged revisions 198824 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines fixes issue with channels not going down after transfer Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop. (closes issue #15216) Reported by: oxymoron Tested by: dvossel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198825 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-02Merged revisions 198791 via svnmerge from file2-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines Correct documentation for the register line, specifically where the domain should be specified. (closes issue #14367) Reported by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198792 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-01Merged revisions 198626 via svnmerge from tilghman1-1/+9
https://origsvn.digium.com/svn/asterisk/trunk ........ r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01 Jun 2009) | 2 lines Add information for new meetme realtime fields ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198627 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-30Merged revisions 198375 via svnmerge from seanbright1-4/+14
https://origsvn.digium.com/svn/asterisk/trunk ........ r198375 | seanbright | 2009-05-30 16:11:33 -0400 (Sat, 30 May 2009) | 13 lines Properly terminate the receive buffer before sending to iksemel. aji_io_recv takes the maximum number of bytes to read (instead of the total buffer size), so we have to subtract 1 from our buffer size. Without this, when we receive packets that are larger than our buffer, iksemel will choke and things get wonky. (closes issue #15232) Reported by: lp0 Patches: 05302009_res_jabber.c.patch uploaded by seanbright (license 71) Tested by: seanbright, lp0 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198384 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-30Merged revisions 198371 via svnmerge from seanbright1-8/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r198371 | seanbright | 2009-05-30 15:38:58 -0400 (Sat, 30 May 2009) | 19 lines Merged revisions 198370 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines Properly terminate AMI JabberSend response messages. The response message (either Error or Success) needs an extra trailing \r\n after the fields to inform the client that the message is complete. (closes issue #14876) Reported by: srt Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71) asterisk_14876.patch uploaded by srt (license 378) trunk-14876-2.diff uploaded by phsultan (license 73) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198372 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-30Merged revisions 198312 via svnmerge from russell1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r198312 | russell | 2009-05-29 22:43:23 -0500 (Fri, 29 May 2009) | 12 lines Merged revisions 198311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) | 5 lines Fix a crash that occurred when MWI SMDI messages expired. (closes issue #14561) Reported by: cmoss28 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198313 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-30Merged revisions 198285 via svnmerge from seanbright1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May 2009) | 15 lines Merged revisions 198251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we treat a missing one. (closes issue #15056) Reported by: p_lindheimer Patches: 05292009_bug15056.diff uploaded by seanbright (license 71) Tested by: p_lindheimer ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198292 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-29Blocked revisions 198088 via svnmergejpeeler0-0/+0
........ r198088 | jpeeler | 2009-05-29 14:19:51 -0500 (Fri, 29 May 2009) | 9 lines New signaling module to handle analog operations in chan_dahdi This branch splits all the analog signaling logic out of chan_dahdi.c into sig_analog.c. Functionality in theory should not change at all. As noted in the code, there is still some unused code remaining that will be cleaned up in a later commit. Review: https://reviewboard.asterisk.org/r/253/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198141 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-29Merged revisions 198072 via svnmerge from mnicholson3-2/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r198072 | mnicholson | 2009-05-29 14:04:24 -0500 (Fri, 29 May 2009) | 21 lines Merged revisions 198068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 | mnicholson | 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) | 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198073 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-29Merged revisions 198000 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r198000 | seanbright | 2009-05-29 14:15:15 -0400 (Fri, 29 May 2009) | 15 lines Merged revisions 197998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197998 | seanbright | 2009-05-29 14:14:12 -0400 (Fri, 29 May 2009) | 8 lines Fix 'make config' target for Slackware. There was a missing semi-colon after the echo statement in the Makefile that was causing problems for some users. Fix suggested by reporter. (closes issue #15225) Reported by: pdavis ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@198001 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Update MixMonitor documentation.lmadsen1-0/+4
Updated the MixMonitor documentation for the 'b' option so that it is more obvious that you must not optimize awat the Local channel when using this option. (issue #14829) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197896 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Blocked revisions 197738 via svnmergetwilson0-0/+0
........ r197738 | twilson | 2009-05-28 14:57:18 -0500 (Thu, 28 May 2009) | 19 lines Add Calendaring support for Asterisk This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS Exchange calendars. Exchange support has only been tested on Exchange Server 2k3 and does not support forms-based authentication at this time (patches *very* welcome). Exchange support is also currently missing the ability to return a list of a meting's attendees (again, patches are very, very welcome). Features include: Querying a calendar for events over a specific time range Checking a calendar's busy status via the dialplan Writing calendar events via the dialplan (CalDAV and Exchange only) Handling calendar event notifications through the dialplan (closes issue #14771) Tested by: lmadsen, twilson, Shivaprakash Review: https://reviewboard.asterisk.org/r/58 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197779 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Merged revisions 197621 via svnmerge from eliel1-2/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r197621 | eliel | 2009-05-28 12:01:48 -0400 (Thu, 28 May 2009) | 19 lines Merged revisions 197562 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197562 | eliel | 2009-05-28 11:21:32 -0400 (Thu, 28 May 2009) | 13 lines Use the address we already know when reloading a peer with nat=yes. If we already have an address for a peer, and we are reloading the sip configuration, try to use that address to contact the peer, instead of getting it from the Contact. (closes issue #15194) Reported by: ibc Patches: sip.patch uploaded by eliel (license 64) Tested by: manwe ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197704 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Merged revisions 197697 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r197697 | file | 2009-05-28 15:45:11 -0300 (Thu, 28 May 2009) | 2 lines Fix a bug where the trunkmtu setting was not set to the default value of 1240 on load but was on reload. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197698 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28'iax show peer blah' now outputs whether or not peer 'blah' is in trunk mode ↵dvossel1-0/+1
or not. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197622 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Merged revisions 197606 via svnmerge from mmichelson3-4/+109
https://origsvn.digium.com/svn/asterisk/trunk ................ r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines Recorded merge of revisions 197588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines Allow for media to arrive from an alternate source when responding to a reinvite with 491. When we receive a SIP reinvite, it is possible that we may not be able to process the reinvite immediately since we have also sent a reinvite out ourselves. The problem is that whoever sent us the reinvite may have also sent a reinvite out to another party, and that reinvite may have succeeded. As a result, even though we are not going to accept the reinvite we just received, it is important for us to not have problems if we suddenly start receiving RTP from a new source. The fix for this is to grab the media source information from the SDP of the reinvite that we receive. This information is passed to the RTP layer so that it will know about the alternate source for media. Review: https://reviewboard.asterisk.org/r/252 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197615 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Merged revisions 197543 via svnmerge from mmichelson3-1/+16
https://origsvn.digium.com/svn/asterisk/trunk ................ r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines Merged revisions 197537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines Add flags to chanspy audiohook so that audio stays in sync. There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197544 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Merged revisions 197538 via svnmerge from file1-1/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r197538 | file | 2009-05-28 11:51:43 -0300 (Thu, 28 May 2009) | 5 lines Fix a bug in stringfields where it did not actually free the pools of memory. (closes issue #15074) Reported by: pj ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197539 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28Merged revisions 197467 via svnmerge from file1-1/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r197467 | file | 2009-05-28 10:47:45 -0300 (Thu, 28 May 2009) | 15 lines Merged revisions 197466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197466 | file | 2009-05-28 10:44:58 -0300 (Thu, 28 May 2009) | 8 lines Fix a bug where the flag indicating the presence of rport would get overwritten by the nat setting. The presence of rport is now stored as a separate flag. Once the dialog is setup and authenticated (or it passes through unauthenticated) the proper nat flag is set. (closes issue #13823) Reported by: dimas ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197469 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-28issue #15155 and issue #15156 from trunkghenry4-13/+255
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197439 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-27Merged revisions 197260 via svnmerge from seanbright1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r197260 | seanbright | 2009-05-27 16:08:16 -0400 (Wed, 27 May 2009) | 6 lines Use bash explicitly when calling build_tools/mkpkgconfig from the Makefile. Since we use bashisms in build_tools/mkpkgconfig, we should call on bash explicitly when running from the Makefile, otherwise we get errors during a 'make install.' (closes issue #15209) Reported by: seandarcy ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197261 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-27Recorded merge of revisions 197209 via svnmerge from tilghman1-4/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r197209 | tilghman | 2009-05-27 14:20:56 -0500 (Wed, 27 May 2009) | 12 lines Recorded merge of revisions 197194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) | 5 lines Use a different determinator on whether to print the delimiter, since leading fields may be blank. (closes issue #15208) Reported by: ramonpeek Patch by me, though inspired in part by a patch from ramonpeek ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197244 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-27Fix broken attended transfersjpeeler2-5/+7
The bridge was terminating immediately after the attended transfer was completed. The problem was because upon reentering ast_channel_bridge nexteventts was checked to see if it was set and if so could possibly return AST_BRIDGE_COMPLETE. (closes issue #15183) Reported by: andrebarbosa Tested by: andrebarbosa, tootai, loloski git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197126 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-27Merged revisions 197089 via svnmerge from seanbright5-9/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r197089 | seanbright | 2009-05-27 12:07:57 -0400 (Wed, 27 May 2009) | 6 lines Fix references to /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf in the sample configuration files. (closes issue #15207) Reported by: seandarcy ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197090 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-27Change register format example to match wording.lmadsen1-1/+1
This change does not affect any other 1.6 branches as they have already been updated for other changes, which uses the word 'domain' as I have here. (closes issue #15204) Reported by: okrief git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197088 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-27Fixes merge issue during r196454.dvossel1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@197086 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-27Merged revisions 196988 via svnmerge from seanbright1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r196988 | seanbright | 2009-05-27 09:02:54 -0400 (Wed, 27 May 2009) | 9 lines Display an error message when chan_alsa fails to load due to a missing or inaccessible configuration file. Before this change, when chan_alsa failed to load due to a missing or inaccessible configuration file, no message would be displayed. With this change, when chan_alsa fails to load due to a missing or inaccessible configuration file, a message will be displayed. (closes issue #14760) Reported by: Nick_Lewis Patches: chan_alsa.c-confload.patch uploaded by Nick (license 657) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@196989 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-26Merged revisions 196843 via svnmerge from russell1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r196843 | russell | 2009-05-26 13:20:57 -0500 (Tue, 26 May 2009) | 16 lines Merged revisions 196826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) | 9 lines Resolve a file handle leak. The frames here should have always been freed. However, out of luck, there was never any memory leaked. However, after file streams became reference counted, this code would leak the file stream for the file being read. (closes issue #15181) Reported by: jkroon ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@196865 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-26Merged revisions 196721 via svnmerge from file1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r196721 | file | 2009-05-26 10:43:13 -0300 (Tue, 26 May 2009) | 7 lines Fix a bug where the sip unregister CLI command did not completely unregister the peer. (closes issue #15118) Reported by: alecdavis Patches: chan_sip_unregister.diff2.txt uploaded by alecdavis (license 585) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@196722 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-26Merged revisions 196658 via svnmerge from file1-4/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r196658 | file | 2009-05-26 10:06:50 -0300 (Tue, 26 May 2009) | 14 lines Merged revisions 196657 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196657 | file | 2009-05-26 10:06:09 -0300 (Tue, 26 May 2009) | 7 lines Remove some bash specific stuff from safe_asterisk. (closes issue #10812) Reported by: paravoid Patches: safe_asterisk_bashism.diff uploaded by tzafrir (license 46) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@196659 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-22Merged revisions 196416 via svnmerge from dvossel2-32/+120
https://origsvn.digium.com/svn/asterisk/trunk ........ r196416 | dvossel | 2009-05-22 16:09:45 -0500 (Fri, 22 May 2009) | 19 lines SIP set outbound transport type from Registration In sip.conf the transport option allows for the configuration of what transport types (udp, tcp, and tls) a peer will accept, but only the first type listed was used for outbound connections. This patch changes this. Now the default transport type is only used until the peer registers. When registration takes place the transport type is parsed out of the Contact header. If the Contact header's transport type is equal to one that the peer supports, the peer's default transport type for outbound connections is set to match the Contact header's type. If the Contact header's transport type is not present, then the peer's default transport type is set to match the one the peer registered with. When a peer unregisters or the registration expires, the default transport type for that peer is reset. (closes issue #12282) Reported by: rjain Patches: reg_patch_1.diff uploaded by dvossel (license 671) Tested by: dvossel (closes issue #14727) Reported by: pj Patches: reg_patch_3.diff uploaded by dvossel (license 671) Tested by: pj, dvossel Review: https://reviewboard.asterisk.org/r/249/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@196454 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-22Merged revisions 196117 via svnmerge from file1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r196117 | file | 2009-05-22 10:56:47 -0300 (Fri, 22 May 2009) | 12 lines Merged revisions 196116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196116 | file | 2009-05-22 10:54:17 -0300 (Fri, 22 May 2009) | 5 lines Fix a bug where using immediate with mISDN caused a cause code of 16 to get sent back instead of 1 if the 's' extension did not exist. (closes issue #12286) Reported by: lmamane ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@196118 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-21Merged revisions 195995 via svnmerge from dvossel1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r195995 | dvossel | 2009-05-21 14:11:49 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195991 | dvossel | 2009-05-21 14:04:56 -0500 (Thu, 21 May 2009) | 14 lines Sign problem calculating timestamp for iax frame leads to no audio on the receiving peer. There are rare cases in which a frame's delivery timestamp is slightly less than the iax2_pvt's offset. This causes the pvt's timestamp to be a small negative number, but since the timestamp value is unsigned it looks like a huge positive number. This patch checks for this negative case and sets the ms to zero. A similar check is already done right below this one in the 'else' statement. (closes issue #15032) Reported by: guillecabeza Patches: chan_iax2.c.patch_timestamp uploaded by guillecabeza (license 380) Tested by: guillecabeza (closes issue #14216) Reported by: Andrey Sofronov ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@195997 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-21Merged revisions 195882 via svnmerge from mnicholson2-0/+42
https://origsvn.digium.com/svn/asterisk/trunk ................ r195882 | mnicholson | 2009-05-21 10:33:55 -0500 (Thu, 21 May 2009) | 20 lines Merged revisions 195881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 | mnicholson | 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) | 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@195894 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-20Merged revisions 195839 via svnmerge from tilghman1-14/+13
https://origsvn.digium.com/svn/asterisk/trunk ........ r195839 | tilghman | 2009-05-20 18:30:05 -0500 (Wed, 20 May 2009) | 3 lines If a variable had a blank value upon the initial setting, then it would do nothing. Identified by Dmitry Andrianov via private email, fixed by me. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@195840 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-20Merged revisions 195698 via svnmerge from file1-5/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r195698 | file | 2009-05-20 14:33:02 -0300 (Wed, 20 May 2009) | 12 lines Merged revisions 195688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195688 | file | 2009-05-20 14:30:25 -0300 (Wed, 20 May 2009) | 5 lines Fix some code that wrongly assumed a pointer would always be non-NULL when dealing with CDRs after a bridge. (closes issue #15079) Reported by: barryf ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@195702 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-20Merged revisions 195636 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r195636 | file | 2009-05-20 14:14:42 -0300 (Wed, 20 May 2009) | 12 lines Merged revisions 195635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5 lines Fix a bug where the MeetMe option 'D' did not actually prompt for the pin. (closes issue #15050) Reported by: pmhaddad ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@195637 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-19Merged revisions 195521 via svnmerge from tilghman1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r195521 | tilghman | 2009-05-19 15:16:01 -0500 (Tue, 19 May 2009) | 14 lines Merged revisions 195520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009) | 7 lines Ensure thread keys are initialized before attempting to access them. (closes issue #14889) Reported by: jaroth Patches: app_voicemail.c.patch uploaded by msirota (license 758) Tested by: msirota, BlargMaN ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@195522 f38db490-d61c-443f-a65b-d21fe96a405b
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