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2009-04-09Blocked revisions 187269 via svnmergekpfleming0-0/+0
........ r187269 | kpfleming | 2009-04-08 22:44:27 -0400 (Wed, 08 Apr 2009) | 5 lines add a dedicated log channel for modules to be able report security-related events, so that they can be fed into external processes for analysis and possible mitigation efforts (inspired by this evening's Toronto Asterisk Users Group meeting and previous dicussions amongst various community members) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@187271 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Blocked revisions 187211 via svnmergejpeeler0-0/+0
........ r187211 | jpeeler | 2009-04-08 16:00:39 -0500 (Wed, 08 Apr 2009) | 20 lines Add timer for features so that backup bridge config can go away The biggest change done here was elimination of the backup_config for use with features. Previously, the bridging code upon detecting a feature would set the start time of the bridge to the start time of the feature. Then after the feature had either expired or timed out the start time would be reset to the true bridge start time from the backup_config. Now, the time differences are calculated with respect to the newly added feature_start_time timeval instead. There should be no behavior changes from the previous functionality aside from the bridge timing being unaffected by either valid or partial feature matches. Previously the timing would be increased by the length of time configured for featuredigittimeout, which was probably never noticed. (closes issue #14503) Reported by: KNK Tested by: jpeeler Review: http://reviewboard.digium.com/r/179/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@187213 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Blocked revisions 187138 via svnmergemmichelson0-0/+0
................ r187138 | mmichelson | 2009-04-08 14:18:10 -0500 (Wed, 08 Apr 2009) | 13 lines Blocked revisions 187135 via svnmerge ........ r187135 | mmichelson | 2009-04-08 14:16:49 -0500 (Wed, 08 Apr 2009) | 8 lines Fix a crash due to too few arguments to RetryDial. (closes issue #14852) Reported by: junky Patches: retry_fix.diff uploaded by junky (license 177) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@187143 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Merged revisions 187046 via svnmerge from mmichelson1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r187046 | mmichelson | 2009-04-08 11:52:20 -0500 (Wed, 08 Apr 2009) | 16 lines Merged revisions 187045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r187045 | mmichelson | 2009-04-08 11:52:03 -0500 (Wed, 08 Apr 2009) | 10 lines Fix a small logical error when loading moh classes. We were unconditionally incrementing the number of mohclasses registered. However, we should actually only increment if the call to moh_register was successful. While this probably has never caused problems, I noticed it and decided to fix it anyway. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@187048 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Merged revisions 186985 via svnmerge from mmichelson1-6/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r186985 | mmichelson | 2009-04-08 10:27:41 -0500 (Wed, 08 Apr 2009) | 30 lines Merged revisions 186984 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186984 | mmichelson | 2009-04-08 10:26:46 -0500 (Wed, 08 Apr 2009) | 24 lines Make a couple of changes with regards to a new message printed in ast_read(). "ast_read() called with no recorded file descriptor" is a new message added after a bug was discovered. Unfortunately, it seems there are a bunch of places that potentially make such calls to ast_read() and trigger this error message to be displayed. This commit does two things to help to make this message appear less. First, the message has been downgraded to a debug level message if dev mode is not enabled. The message means a lot more to developers than it does to end users, and so developers should take an effort to be sure to call ast_read only when a channel is ready to be read from. However, since this doesn't actually cause an error in operation and is not something a user can easily fix, we should not spam their console with these messages. Second, the message has been moved to after the check for any pending masquerades. ast_read() being called with no recorded file descriptor should not interfere with a masquerade taking place. This could be seen as a simple way of resolving issue #14723. However, I still want to try to clear out the existing ways of triggering this message, since I feel that would be a better resolution for the issue. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186987 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Merged revisions 186899 via svnmerge from tilghman1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r186899 | tilghman | 2009-04-08 00:06:22 -0500 (Wed, 08 Apr 2009) | 2 lines Add lastms to the require API call. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186900 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Merged revisions 186842 via svnmerge from mmichelson2-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r186842 | mmichelson | 2009-04-07 19:09:28 -0500 (Tue, 07 Apr 2009) | 14 lines Merged revisions 186841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186841 | mmichelson | 2009-04-07 19:09:04 -0500 (Tue, 07 Apr 2009) | 8 lines Fix a few typos of the word "frequency." (closes issue #14842) Reported by: jvandal Patches: frequency-typo.diff uploaded by jvandal (license 413) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186844 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-08Merged revisions 186837 via svnmerge from mmichelson1-3/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r186837 | mmichelson | 2009-04-07 19:01:49 -0500 (Tue, 07 Apr 2009) | 7 lines Fix bad merge from fix for issue 13867. (closes issue #14686) Reported by: davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186839 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-07Merged revisions 186833 via svnmerge from mmichelson1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r186833 | mmichelson | 2009-04-07 18:50:56 -0500 (Tue, 07 Apr 2009) | 15 lines Merged revisions 186832 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186832 | mmichelson | 2009-04-07 18:49:49 -0500 (Tue, 07 Apr 2009) | 8 lines Set the AST_FEATURE_WARNING_ACTIVE flag when a p2p bridge returns AST_BRIDGE_RETRY. Without this flag set, warning sounds will not be properly played to either party of the bridge. (closes issue #14845) Reported by: adomjan ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186835 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-07Merged revisions 186799 via svnmerge from tilghman1-12/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r186799 | tilghman | 2009-04-07 17:23:46 -0500 (Tue, 07 Apr 2009) | 10 lines Merged revisions 186775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186775 | tilghman | 2009-04-07 17:16:50 -0500 (Tue, 07 Apr 2009) | 3 lines Fix Macro documentation to match current (and intended) behavior. (See -dev mailing list) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186806 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-07Merged revisions 186720 via svnmerge from mmichelson1-3/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r186720 | mmichelson | 2009-04-07 15:46:18 -0500 (Tue, 07 Apr 2009) | 12 lines Merged revisions 186719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186719 | mmichelson | 2009-04-07 15:43:49 -0500 (Tue, 07 Apr 2009) | 6 lines Ensure that \r\n is printed after the ActionID in an OriginateResponse. (closes issue #14847) Reported by: kobaz ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186722 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-06Blocked revisions 186620 via svnmergemmichelson0-0/+0
........ r186620 | mmichelson | 2009-04-06 11:06:25 -0500 (Mon, 06 Apr 2009) | 3 lines Silly svn. These files didn't get merged over in the merge of the issue8824 branch. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186622 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-06Blocked revisions 186566 via svnmergemmichelson0-0/+0
................ r186566 | mmichelson | 2009-04-06 08:57:39 -0500 (Mon, 06 Apr 2009) | 8 lines Blocked revisions 186565 via svnmerge ........ r186565 | mmichelson | 2009-04-06 08:54:41 -0500 (Mon, 06 Apr 2009) | 3 lines Revert commit 186445 because it causes the build to fail when IMAP_STORAGE is used. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186568 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03Blocked revisions 186525 via svnmergemmichelson0-0/+0
........ r186525 | mmichelson | 2009-04-03 17:41:46 -0500 (Fri, 03 Apr 2009) | 22 lines This commit introduces COLP/CONP and Redirecting party information into Asterisk. The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186527 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03Merged revisions 186461 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r186461 | kpfleming | 2009-04-03 15:20:01 -0500 (Fri, 03 Apr 2009) | 11 lines Merged revisions 186458 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186458 | kpfleming | 2009-04-03 15:19:20 -0500 (Fri, 03 Apr 2009) | 5 lines Fix a bug where DAHDI/Zaptel channels would not properly switch formats when requested Don't offer AST_FORMAT_SLINEAR on DAHDI/Zaptel channels... while it could provide a slight performance benefit, the translation core in Asterisk has some flaws when a channel driver offers multiple raw formats. this fix is much simpler than fixing the translation core to solve that issue (although that will be done later). ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186466 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03Merged revisions 186444,186447 via svnmerge from tilghman2-24/+111
https://origsvn.digium.com/svn/asterisk/trunk ................ r186444 | tilghman | 2009-04-03 14:30:34 -0500 (Fri, 03 Apr 2009) | 14 lines Merged revisions 186415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186415 | tilghman | 2009-04-03 14:06:58 -0500 (Fri, 03 Apr 2009) | 7 lines Distinguish in a sent email between simple sends and forwards. (closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen ........ ................ r186447 | tilghman | 2009-04-03 14:59:55 -0500 (Fri, 03 Apr 2009) | 9 lines Merged revisions 186445 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186445 | tilghman | 2009-04-03 14:56:48 -0500 (Fri, 03 Apr 2009) | 2 lines Found a conflict in the last commit, due to multiple targets ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186448 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03Merged revisions 186379 via svnmerge from dvossel1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r186379 | dvossel | 2009-04-03 11:29:47 -0500 (Fri, 03 Apr 2009) | 4 lines audio_audiohook_write_list() did not correctly update sample size after ast_translate. audio_audiohook_write_list() did not take into account that the sample size may change after translation depending on if the original frame is is 8khz or 16khz. the sample size is now updated after translating to reflect this possibility. This caused the audio on the receiving end to sound terrible. Thanks to jcolp and mmichelson for helping me work this out. (issue AST-197) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186381 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03Merged revisions 186321 via svnmerge from file1-7/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r186321 | file | 2009-04-03 12:52:50 -0300 (Fri, 03 Apr 2009) | 12 lines Merged revisions 186320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186320 | file | 2009-04-03 12:48:56 -0300 (Fri, 03 Apr 2009) | 5 lines Fix a problem with the crypto variable definitions not actually being defined properly. (closes issue #14804) Reported by: jvandal ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186323 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03Merged revisions 186286 via svnmerge from mmichelson1-16/+8
https://origsvn.digium.com/svn/asterisk/trunk ........ r186286 | mmichelson | 2009-04-03 09:32:05 -0500 (Fri, 03 Apr 2009) | 20 lines Fix the ability to retrieve voicemail messages from IMAP. A recent change made interactive vm_states no longer get added to the list of vm_states and instead get stored in thread-local storage. In trunk and all the 1.6.X branches, the problem is that when we search for messages in a voicemail box, we would attempt to update the appropriate vm_state struct by directly searching in the list of vm_states instead of using the get_vm_state_by_imap_user function. This meant we could not find the interactive vm_state that we wanted. (closes issue #14685) Reported by: BlargMaN Patches: 14685.patch uploaded by mmichelson (license 60) Tested by: BlargMaN, qualleyiv, mmichelson ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186288 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03Merged revisions 186230 via svnmerge from russell1-2/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r186230 | russell | 2009-04-02 21:03:48 -0500 (Thu, 02 Apr 2009) | 29 lines Merged revisions 186229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186229 | russell | 2009-04-02 20:57:44 -0500 (Thu, 02 Apr 2009) | 21 lines Fix a memory leak in cdr_radius. I came across this while doing some testing of my ast_channel_ao2 branch. After running a test overnight that generated over 5 million calls, Asterisk had taken up about 1 GB of my system memory. So, I re-ran the test with MALLOC_DEBUG turned on. However, it showed no leaks in Asterisk during the test, even though Asterisk was still consuming it somehow. Instead, I turned to valgrind, which when run with --leak-check=full, told me exactly where the leak came from, which was from allocations inside the radiusclient-ng library. This explains why MALLOC_DEBUG did not report it. After a bit of analysis, I found that we were leaking a little bit of memory every time a CDR record was passed to cdr_radius. I don't actually have a radius server set up to receive CDR records. However, I always have my development systems compile and install all modules. In addition to making sure there are not build errors across modules, always loading modules helps find bugs like this, too, so it is strongly recommend for all developers. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186232 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186175 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r186175 | mmichelson | 2009-04-02 16:56:21 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 186174 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186174 | mmichelson | 2009-04-02 16:55:34 -0500 (Thu, 02 Apr 2009) | 5 lines Fix instructions in one-step parking comment to make more sense. Changed a capital K to a lowercase k. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186177 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186101 via svnmerge from kpfleming1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r186101 | kpfleming | 2009-04-02 12:26:07 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r186081 | kpfleming | 2009-04-02 12:21:29 -0500 (Thu, 02 Apr 2009) | 3 lines ensure that the buffer passed to DAHDI_SET_BUFINFO is fully initialized ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186108 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 186060 via svnmerge from tilghman2-10/+106
https://origsvn.digium.com/svn/asterisk/trunk ................ r186060 | tilghman | 2009-04-02 12:10:28 -0500 (Thu, 02 Apr 2009) | 16 lines Merged revisions 186059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r186059 | tilghman | 2009-04-02 12:09:13 -0500 (Thu, 02 Apr 2009) | 9 lines Merged revisions 186056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@186062 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merged revisions 185953 via svnmerge from kpfleming1-11/+20
https://origsvn.digium.com/svn/asterisk/trunk ................ r185953 | kpfleming | 2009-04-02 08:51:44 -0500 (Thu, 02 Apr 2009) | 11 lines Merged revisions 185952 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185952 | kpfleming | 2009-04-02 08:43:43 -0500 (Thu, 02 Apr 2009) | 5 lines the DAHDI_GETCONF, DAHDI_SETCONF and DAHDI_GET_PARAMS ioctls were recently corrected to show that they do, in fact, read data from userspace as part of their work. due to this fix, valgrind now reports a number of cases where chan_dahdi passed an uninitialized (or partially) buffer to these ioctls, which could lead to unexpected behavior. this patch corrects chan_dahdi to ensure that buffers passed to these ioctls are always fully initialized. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185956 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-01Merged revisions 185846 via svnmerge from dvossel1-13/+30
https://origsvn.digium.com/svn/asterisk/trunk ................ r185846 | dvossel | 2009-04-01 14:03:32 -0500 (Wed, 01 Apr 2009) | 16 lines Merged revisions 185845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185845 | dvossel | 2009-04-01 14:02:00 -0500 (Wed, 01 Apr 2009) | 10 lines Fixes issue with dropped calles due to re-Invite glare and re-Invites never executing after a 491 Acknowledgement for 491 responses were never being processed because it didn't match our pending invite's seqno. Since the ACK was never processed, the 491 frame would continue to be retransmitted until eventually the call was dropped due to max retries. Now during a pending invite, if we receive another invite, we send an 491 and hold on to that glare invite's seqno in the "glareinvite" variable for that sip_pvt struct. When ACK's are received, we first check to see if it is in response to our pending invite, if not we check to see if it is in response to a glare invite. In this case, it is in response to the glare invite and must be dealt with or the call is dropped. I've changed the wait time for resending the re-Invite after receving a 491 response to comply with RFC 3261. Before this patch the scheduled re-Invite would only change a flag indicating that the re-Invite should be sent out, now it actually sends it out as well. (closes issue #12013) Reported by: alx Review: http://reviewboard.digium.com/r/213/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185848 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-01Blocked revisions 185777 via svnmergemmichelson0-0/+0
........ r185777 | mmichelson | 2009-04-01 08:59:34 -0500 (Wed, 01 Apr 2009) | 5 lines Address Russell's comments regarding rev 185704. Use ast_debug and ast_softhangup_nolock. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185779 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-01Merged revisions 185772 via svnmerge from russell1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r185772 | russell | 2009-04-01 08:48:26 -0500 (Wed, 01 Apr 2009) | 14 lines Merged revisions 185771 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185771 | russell | 2009-04-01 08:47:30 -0500 (Wed, 01 Apr 2009) | 6 lines Fix a case where DTMF could bypass audiohooks. This change fixes a situation where an audiohook that wants DTMF would not actually get it. This is in the code path where we end DTMF digit length emulation while handling a NULL frame. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185774 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-01Blocked revisions 185704 via svnmergemmichelson0-0/+0
........ r185704 | mmichelson | 2009-03-31 19:39:01 -0500 (Tue, 31 Mar 2009) | 8 lines Allow the AMI Hangup command to accept a Cause header. (closes issue #14695) Reported by: mneuhauser Patches: cause-for-hangup-manager-action.patch uploaded by mneuhauser (license 425) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185706 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Merged revisions 185664 via svnmerge from kpfleming0-0/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r185664 | kpfleming | 2009-03-31 17:35:07 -0500 (Tue, 31 Mar 2009) | 1 line ignore copied (generated) file ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185666 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Merged revisions 185600 via svnmerge from mmichelson1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r185600 | mmichelson | 2009-03-31 17:02:48 -0500 (Tue, 31 Mar 2009) | 12 lines Merged revisions 185599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185599 | mmichelson | 2009-03-31 17:00:01 -0500 (Tue, 31 Mar 2009) | 6 lines Fix crash that would occur if an empty member was specified in queues.conf. (closes issue #14796) Reported by: pida ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185602 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Blocked revisions 185581 via svnmergekpfleming0-0/+0
........ r185581 | kpfleming | 2009-03-31 16:29:50 -0500 (Tue, 31 Mar 2009) | 19 lines Optimizations to the stringfields API This patch provides a number of optimizations to the stringfields API, focused around saving (not wasting) memory whenever possible. Thanks to Mark Michelson for inspiring this work and coming up with the first two optimizations that are represented here: Changes: - Cleanup of some code, fix incorrect doxygen comments - When a field is emptied or replaced with a new allocation, decrease the amount of 'active' space in the pool it was held in; if that pool reaches zero active space, and is not the current pool, then free it as it is no longer in use - When allocating a pool, try to allocate a size that will fit in a 'standard' malloc() allocation without wasting space - When allocating space for a field, store the amount of space in the two bytes immediately preceding the field; this eliminates the need to call strlen() on the field when overwriting it, and more importantly it 'remembers' the amount of space the field has available, even if a shorter string has been stored in it since it was allocated - Don't automatically double the size of each successive pool allocated; it's wasteful http://reviewboard.digium.com/r/165/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185593 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Blocked revisions 185532 via svnmergemmichelson0-0/+0
................ r185532 | mmichelson | 2009-03-31 15:56:46 -0500 (Tue, 31 Mar 2009) | 8 lines Blocked revisions 185531 via svnmerge ........ r185531 | mmichelson | 2009-03-31 15:55:47 -0500 (Tue, 31 Mar 2009) | 3 lines Use AST_SCHED_DEL_SPINLOCK instead of manually using the logic. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185534 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Merged revisions 185469 via svnmerge from mmichelson1-6/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r185469 | mmichelson | 2009-03-31 14:46:18 -0500 (Tue, 31 Mar 2009) | 14 lines Merged revisions 185468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185468 | mmichelson | 2009-03-31 14:45:30 -0500 (Tue, 31 Mar 2009) | 8 lines Fix Russian voicemail intro to say the word "messages" properly. (closes issue #14736) Reported by: chappell Patches: voicemail_no_messages.diff uploaded by chappell (license 8) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185471 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Merged revisions 185363 via svnmerge from dbrooks1-7/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r185363 | dbrooks | 2009-03-31 11:46:57 -0500 (Tue, 31 Mar 2009) | 44 lines Merged revisions 185362 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185362 | dbrooks | 2009-03-31 11:37:12 -0500 (Tue, 31 Mar 2009) | 35 lines Fix incorrect parsing in chan_gtalk when xmpp contains extra whitespaces To drill into the xmpp to find the capabilities between channels, chan_gtalk calls iks_child() and iks_next(). iks_child() and iks_next() are functions in the iksemel xml parsing library that traverse xml nodes. The bug here is that both iks_child() and iks_next() will return the next iks_struct node *regardless* of type. chan_gtalk expects the next node to be of type IKS_TAG, which in most cases, it is, but in this case (a call being made from the Empathy IM client), there exists iks_struct nodes which are not IKS_TAG data (they are extraneous whitespaces), and chan_gtalk doesn't handle that case, so capabilities don't match, and a call cannot be made. iks_first_tag() and iks_next_tag(), on the other hand, will not return the very next iks_struct, but will check to see if the next iks_struct is of type IKS_TAG. If it isn't, it will be skipped, and the next struct of type IKS_TAG it finds will be returned. This assures that chan_gtalk will find the iks_struct it is looking for. This fix simply changes all calls to iks_child() and iks_next() to become calls to iks_first_tag() and iks_next_tag(), which resolves the capability matching. The following is a payload listing from Empathy, which, due to the extraneous whitespace, will not be parsed correctly by iksemel: <iq from='dbrooksjab@235-22-24-10/Telepathy' to='astjab@235-22-24-10/asterisk' type='set' id='542757715704'> <session xmlns='http://www.google.com/session' initiator='dbrooksjab@235-22-24-10/Telepathy' type='initiate' id='1837267342'> <description xmlns='http://www.google.com/session/phone'> <payload-type clockrate='16000' name='speex' id='96'/> <payload-type clockrate='8000' name='PCMA' id='8'/> <payload-type clockrate='8000' name='PCMU' id='0'/> <payload-type clockrate='90000' name='MPA' id='97'/> <payload-type clockrate='16000' name='SIREN' id='98'/> <payload-type clockrate='8000' name='telephone-event' id='99'/> </description> </session> </iq> Review: http://reviewboard.digium.com/r/181/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185427 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Blocked revisions 185299 via svnmergemmichelson0-0/+0
................ r185299 | mmichelson | 2009-03-31 10:34:29 -0500 (Tue, 31 Mar 2009) | 15 lines Blocked revisions 185298 via svnmerge ........ r185298 | mmichelson | 2009-03-31 10:34:05 -0500 (Tue, 31 Mar 2009) | 10 lines Fix some state_interface stuff that was in trunk but not in the backport to 1.4. Issue #14359 was fixed between the time that I posted the review of the backport of the state interface change for 1.4. This merges the changes from that issue back into 1.4. (closes issue #14359) Reported by: francesco_r ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185300 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Merged revisions 185261 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r185261 | russell | 2009-03-31 09:53:45 -0500 (Tue, 31 Mar 2009) | 5 lines Don't free() an astobj2 object. (closes issue #14672) Reported by: makoto ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185263 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-31Merged revisions 185197 via svnmerge from file1-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r185197 | file | 2009-03-31 11:07:36 -0300 (Tue, 31 Mar 2009) | 15 lines Merged revisions 185196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185196 | file | 2009-03-31 11:06:39 -0300 (Tue, 31 Mar 2009) | 8 lines Fix crash when moving audiohooks between channels. Handle the scenario where we are called to move audiohooks between channels and the source channel does not actually have any on it. (closes issue #14734) Reported by: corruptor ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185199 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 185123 via svnmerge from rmudgett2-8/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r185123 | rmudgett | 2009-03-30 15:42:14 -0500 (Mon, 30 Mar 2009) | 9 lines Merged revisions 185121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185121 | rmudgett | 2009-03-30 15:40:11 -0500 (Mon, 30 Mar 2009) | 1 line Update the channel allocation method documentation. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185127 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 185122 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r185122 | rmudgett | 2009-03-30 15:41:24 -0500 (Mon, 30 Mar 2009) | 26 lines Merged revisions 185120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185120 | rmudgett | 2009-03-30 15:38:11 -0500 (Mon, 30 Mar 2009) | 19 lines Make chan_misdn BRI TE side normally defer channel selection to the NT side. Channel allocation collisions are not handled by chan_misdn very well. This patch simply avoids the problem for BRI only. For PRI, allocation collisions are still possible but less likely since there are simply more channels available and each end could use a different allocation strategy. misdn.conf options available: te_choose_channel - Use to force the TE side to allocate channels. method - Specify the channel allocation strategy. (closes issue #13488) Reported by: Christian_Pinedo Patches: isdn_lib.patch.txt uploaded by crich Tested by: crich, siepkes, festr ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185126 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 185072 via svnmerge from mmichelson1-70/+81
https://origsvn.digium.com/svn/asterisk/trunk ................ r185072 | mmichelson | 2009-03-30 11:26:48 -0500 (Mon, 30 Mar 2009) | 45 lines Merged revisions 185031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r185031 | mmichelson | 2009-03-30 11:17:35 -0500 (Mon, 30 Mar 2009) | 39 lines Fix queue weight behavior so that calls in low-weight queues are not inappropriately blocked. (This is copied and pasted from the review request I made for this patch) Asterisk has some odd behavior when queue weights are used. The current logic used when potentially calling a queue member is: If the member we are going to call is part of another queue and _that other queue has any callers in it_ and has a higher weight than the queue we are calling from, then don't try to contact that member. The issue here is what I have marked with underscores. If the higher-weighted queue has any callers in it at all, then the queue member will be unreachable from the lower-weighted queue. This has the potential to be really really bad if using a queue strategy, such as leastrecent or fewestcalls, with the potential to call the same member repeatedly. The fix proposed by garychen on issue 13220 is very simple and, as far as I can see, works well for this situation. With this set of changes, the logic used becomes: If the member we are going to call is part of another queue, the other queue has a higher weight than the queue we are calling from, and the higher weight queue has at least as many callers as available members, then do not try to contact the queue member. If the higher weighted queue has fewer callers than available members, then there is no reason to deny the call to this member since the other queue can afford to spare a member. Since the fix involved writing a generic function for determining the number of available members in the queue, I also modified the is_our_turn function to make use of the new num_available_members function to determine if it is our turn to try calling a member. There is one small behavior change. Before writing this patch, if you had autofill disabled, then if you were the head caller in a queue, you would automatically be told that it was your turn to try calling a member. This did not take into account whether there were actually any queue members available to take the call. Now we actually make sure there is at least one member available to take the call if autofill is disabled. (closes issue #13220) Reported by: garychen Review: http://reviewboard.digium.com/r/202/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@185088 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Blocked revisions 184986 via svnmergemmichelson0-0/+0
................ r184986 | mmichelson | 2009-03-30 10:25:04 -0500 (Mon, 30 Mar 2009) | 27 lines Blocked revisions 184980 via svnmerge ........ r184980 | mmichelson | 2009-03-30 10:23:59 -0500 (Mon, 30 Mar 2009) | 22 lines Backport state interface changes to app_queue from trunk. After several issues raised on the Asterisk bugtracker against the 1.4 branch were determined to be fixable with the state interface change available in the 1.6.X series, it finally came time to just suck it up and backport the change. For a detailed explanation of what this change entails, the original trunk commit for this feature may be found here: http://svn.digium.com/view/asterisk?view=revision&revision=97203 In addition, the details for the use of this change to fix the problems stated in issue #12970 may be found in the review request I made for this change. It is linked below. (closes issue #12970) Reported by: edugs15 Review: http://reviewboard.digium.com/r/116 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@184992 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 184948 via svnmerge from file1-270/+238
https://origsvn.digium.com/svn/asterisk/trunk ................ r184948 | file | 2009-03-30 11:37:47 -0300 (Mon, 30 Mar 2009) | 21 lines Merged revisions 184947 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184947 | file | 2009-03-30 11:35:47 -0300 (Mon, 30 Mar 2009) | 14 lines Improve our handling of T38 in the initial INVITE from a device. We now answer with matching media streams to what is requested. If an INVITE is received with both a T38 and RTP media stream this means we answer with both. For any outgoing calls created as a result of this inbound one no T38 is requested in the initial INVITE. Instead if we start receiving udptl packets we trigger a reinvite on the outbound side. (closes issue #12437) Reported by: marsosa Tested by: pinga-fogo, okrief, file, afu Review: http://reviewboard.digium.com/r/208/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@184950 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-30Merged revisions 184910 via svnmerge from russell1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r184910 | russell | 2009-03-30 08:55:44 -0500 (Mon, 30 Mar 2009) | 4 lines Fix build error when chan_h323 is not being built. (reported by cai1982 in #asterisk-dev) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@184912 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-29Merged revisions 184843 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r184843 | russell | 2009-03-29 00:52:20 -0500 (Sun, 29 Mar 2009) | 13 lines Merged revisions 184842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184842 | russell | 2009-03-29 00:51:55 -0500 (Sun, 29 Mar 2009) | 5 lines Ensure targs variable is fully initialized. (closes issue #14758) Reported by: tim_ringenbach ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@184845 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-29Merged revisions 184838 via svnmerge from russell1-15/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r184838 | russell | 2009-03-29 00:32:04 -0500 (Sun, 29 Mar 2009) | 8 lines Simplify chan_h323 build to not require a second run of "make". (closes issue #14715) Reported by: jthurman Patches: h323-makefile-1.6.2.0-beta1.patch uploaded by jthurman (license 614) Tested by: tzafrir, russell ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@184840 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-27Merged revisions 184762 via svnmerge from kpfleming5-136/+121
https://origsvn.digium.com/svn/asterisk/trunk ........ r184762 | kpfleming | 2009-03-27 14:10:32 -0500 (Fri, 27 Mar 2009) | 12 lines Improve timing interface to remember which provider provided a timer The ability to load/unload timing interfaces is nice, but it means that when a timer is allocated, it may come from provider A, but later provider B becomes the 'preferred' provider. If this happens, all timer API calls on the timer that was provided by provider A will actually be handed to provider B, which will say WTF and return an error. This patch changes the timer API to include a pointer to the provider of the timer handle so that future operations on the timer will be forwarded to the proper provider. (closes issue #14697) Reported by: moy Review: http://reviewboard.digium.com/r/211/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@184765 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-27Merged revisions 184726 via svnmerge from russell2-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r184726 | russell | 2009-03-27 13:04:43 -0500 (Fri, 27 Mar 2009) | 2 lines Use ast_random() instead of rand() to ensure we use the best RNG available. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@184728 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-27Merged revisions 184673 via svnmerge from file1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r184673 | file | 2009-03-27 12:46:46 -0300 (Fri, 27 Mar 2009) | 7 lines Fix speech structure leak in the AGI speech recognition integration. The AGI dialplan applications did not destroy the speech structure automatically if it was not destroyed by the running AGI script. They will now do this. (issue LUMENVOX-15) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@184675 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-27Merged revisions 184630 via svnmerge from russell6-10/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r184630 | russell | 2009-03-27 09:00:18 -0500 (Fri, 27 Mar 2009) | 2 lines Change g_eid to ast_eid_default. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@184631 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-27Merged revisions 184566 via svnmerge from file1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r184566 | file | 2009-03-27 10:15:26 -0300 (Fri, 27 Mar 2009) | 16 lines Merged revisions 184565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r184565 | file | 2009-03-27 10:06:45 -0300 (Fri, 27 Mar 2009) | 9 lines Fix an issue where nat=yes would not always take effect for the RTP session on outgoing calls. If calls were placed using an IP address or hostname the global nat setting was copied over but was not set on the RTP session itself. This caused the RTP stack to not perform symmetric RTP actions. (closes issue #14546) Reported by: acunningham ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@184587 f38db490-d61c-443f-a65b-d21fe96a405b
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