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2009-03-11Merged revisions 181371 via svnmerge from dvossel2-31/+71
https://origsvn.digium.com/svn/asterisk/trunk ................ r181371 | dvossel | 2009-03-11 12:34:57 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181340 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181340 | dvossel | 2009-03-11 12:25:31 -0500 (Wed, 11 Mar 2009) | 11 lines encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct. (closes issue #14607) Reported by: stevenla Tested by: dvossel Review: http://reviewboard.digium.com/r/192/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181372 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181345 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r181345 | file | 2009-03-11 14:26:40 -0300 (Wed, 11 Mar 2009) | 21 lines Merged revisions 181328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181328 | file | 2009-03-11 14:22:52 -0300 (Wed, 11 Mar 2009) | 14 lines Fix issue where an attended transfer could not be completed under a rare scenario. When completing an attended transfer chan_sip does a check to make sure the extension in the URI portion of the Refer-To header is a local valid extension. We don't actually need to check this since we know for sure the other channel is already up and talking to the extension. Some devices do not put the extension in the Refer-To header either, which can cause the extension check to fail. We now no longer do this check if it is an attended transfer. (closes issue #14628) Reported by: sverre Patches: 14628.diff uploaded by file (license 11) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181352 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181296 via svnmerge from file1-3/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r181296 | file | 2009-03-11 13:40:48 -0300 (Wed, 11 Mar 2009) | 16 lines Merged revisions 181295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181295 | file | 2009-03-11 13:36:50 -0300 (Wed, 11 Mar 2009) | 9 lines Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto. When dtmfmode was set to auto the inband DTMF detector was not setup on outgoing SIP calls. This caused inband DTMF detection to fail. The inband DTMF detector is now setup for both dtmfmode inband and auto. (closes issue #13713) Reported by: makoto ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181297 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Blocked revisions 181292 via svnmergerussell0-0/+0
........ r181292 | russell | 2009-03-11 11:19:38 -0500 (Wed, 11 Mar 2009) | 2 lines Replace contents of this doc with a pointer to its new home ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181293 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11add missing header filejpeeler1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181284 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Blocked revisions 181244 via svnmergemmichelson0-0/+0
........ r181244 | mmichelson | 2009-03-11 09:28:40 -0500 (Wed, 11 Mar 2009) | 11 lines Fix segfault when dialing a typo'd queue If trying to dial a non-existent queue, there would be a segfault when attempting to access q->weight, even though q was NULL. This problem was introduced during the queue-reset merge and thus only affects trunk. (closes issue #14643) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181245 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Blocked revisions 181210 via svnmergefile0-0/+0
........ r181210 | file | 2009-03-11 10:44:42 -0300 (Wed, 11 Mar 2009) | 3 lines Don't play the "you are about to be placed into the conference" and "the leader has left the conference" sounds if the quiet option is enabled. (reported by Vadim Lebedev on the asterisk-dev list) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181211 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Fix merge oops from 181137jpeeler1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181178 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181135 via svnmerge from jpeeler6-128/+49
https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines Fix malloc debug macros to work properly with h323. The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181137 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Blocked revisions 181099 via svnmergerussell0-0/+0
........ r181099 | russell | 2009-03-10 21:25:24 -0500 (Tue, 10 Mar 2009) | 2 lines tabs to spaces ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181100 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181032-181033 via svnmerge from mmichelson1-0/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r181032 | mmichelson | 2009-03-10 19:46:47 -0500 (Tue, 10 Mar 2009) | 19 lines Merged revisions 181029,181031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181029 | mmichelson | 2009-03-10 19:30:26 -0500 (Tue, 10 Mar 2009) | 9 lines Fix incorrect tag checking on transfers when pedantic=yes is enabled. (closes issue #14611) Reported by: klaus3000 Patches: patch_chan_sip_attended_transfer_1.4.23.txt uploaded by klaus3000 (license 65) Tested by: klaus3000 ........ r181031 | mmichelson | 2009-03-10 19:32:40 -0500 (Tue, 10 Mar 2009) | 3 lines Remove unused variables. ........ ................ r181033 | mmichelson | 2009-03-10 19:49:00 -0500 (Tue, 10 Mar 2009) | 3 lines Add missing comment that quotes RFC 3891 ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@181034 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Merged revisions 180944 via svnmerge from qwell4-24/+22
https://origsvn.digium.com/svn/asterisk/trunk ................ r180944 | qwell | 2009-03-10 17:03:41 -0500 (Tue, 10 Mar 2009) | 9 lines Merged revisions 180941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180941 | qwell | 2009-03-10 17:02:18 -0500 (Tue, 10 Mar 2009) | 1 line Make things happier when using autoconf 2.62+ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180946 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180942 via svnmergerussell0-0/+0
........ r180942 | russell | 2009-03-10 17:03:16 -0500 (Tue, 10 Mar 2009) | 2 lines Add some notes on getting in contact with the dev community ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180943 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180938 via svnmergerussell0-0/+0
........ r180938 | russell | 2009-03-10 16:55:49 -0500 (Tue, 10 Mar 2009) | 2 lines Remove difficulty and language specifiers ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180939 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180935 via svnmergerussell0-0/+0
........ r180935 | russell | 2009-03-10 16:45:54 -0500 (Tue, 10 Mar 2009) | 2 lines Expand upon documentation of manager event project ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180936 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180898 via svnmergemvanbaak0-0/+0
........ r180898 | mvanbaak | 2009-03-10 22:15:29 +0100 (Tue, 10 Mar 2009) | 10 lines list the move of the astvarrundir from /var/run to /var/run/asterisk (actually its $(localstatedir)/run/asterisk Makes setups with asterisk as non-root easier to manage because you can setup permissions on this dir instead of touching a file and setting permissions on that. Files that come to mind are asterisk.pid and asterisk.ctl socket. Prodded by and ok @russell ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180899 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180862 via svnmergerussell0-0/+0
........ r180862 | russell | 2009-03-10 14:36:21 -0500 (Tue, 10 Mar 2009) | 1 line add more projects ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180863 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Blocked revisions 180859 via svnmergerussell0-0/+0
........ r180859 | russell | 2009-03-10 14:23:41 -0500 (Tue, 10 Mar 2009) | 1 line add more project ideas ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180860 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Merged revisions 180800 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 | file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines Reset the thread local string buffer when handling the UserEvent action. (closes issue #14593) Reported by: JimDickenson ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180801 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10If a port is specified when dialing a peer then use it.file1-0/+5
(closes issue #14626) Reported by: acunningham git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180799 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-09Blocked revisions 180750 via svnmergerussell0-0/+0
........ r180750 | russell | 2009-03-09 17:00:42 -0500 (Mon, 09 Mar 2009) | 4 lines Add current mentors list, and first pass on a project list broken out of "PineMango" I will work on adding projects that have been sent to be via email tomorrow. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180751 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-09Blocked revisions 180719 via svnmergejpeeler0-0/+0
........ r180719 | jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines Add Doxygen documentation for API changes from 1.6.0 to 1.6.1 Copied from my review board description: This is a continuation of the API changes documentation started for describing changes between releases. Most of the API changes were pretty simple needing only to be brought to attention via the new "Asterisk API Changes" list. However, if you see anything that needs further explanation feel free to supplement what is there. The current method of documenting is to add (in the header file): \version <ver number> <description of changes> and then to add the function to the change list in doxyref.h on the AstAPIChanges page. I also made sure all the functions that were newly added were tagged with \since 1.6.1. I think this is a good habit to start both for the historical aspect as well as for the future ability to easily add a "New Asterisk API" page. Review: http://reviewboard.digium.com/r/190/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180720 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-09Ensure that the new outgoing dialog to a peer is able to set the socket ↵file1-7/+11
details, even if the default is present. (closes issue #14480) Reported by: jon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180718 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-09Blocked revisions 180684 via svnmergerussell0-0/+0
........ r180684 | russell | 2009-03-09 09:14:34 -0500 (Mon, 09 Mar 2009) | 2 lines Add skeleton for GSoC ideas list ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180685 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-07Blocked revisions 180641 via svnmergerussell0-0/+0
........ r180641 | russell | 2009-03-07 09:36:00 -0600 (Sat, 07 Mar 2009) | 7 lines Make some minor updates to the doxygen configuration - add bridges directory to be processed - add some res/ subdirs - alphabetize subdirs - use consistent indentation ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180642 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-06Merged revisions 180579 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r180579 | mmichelson | 2009-03-06 12:25:44 -0600 (Fri, 06 Mar 2009) | 9 lines Merged revisions 180567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180567 | mmichelson | 2009-03-06 12:23:09 -0600 (Fri, 06 Mar 2009) | 2 lines Make compilation succeed in dev-mode when IMAP storage is enabled. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180582 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-06Merged revisions 180534 via svnmerge from dvossel1-30/+50
https://origsvn.digium.com/svn/asterisk/trunk ................ r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009) | 15 lines Merged revisions 180532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) | 9 lines Fix handling of backreferences for ENUM lookups enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted. (closes issue #14576) Reported by: chris-mac Review: http://reviewboard.digium.com/r/187/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180535 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merged revisions 180465 via svnmerge from mmichelson1-0/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r180465 | mmichelson | 2009-03-05 17:26:58 -0600 (Thu, 05 Mar 2009) | 22 lines Merged revisions 180464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180464 | mmichelson | 2009-03-05 17:26:11 -0600 (Thu, 05 Mar 2009) | 16 lines [IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts. There was a fix put in a while back so that an X-Asterisk-VM-Context message header was added to stored IMAP voicemails. This would allow for us to differentiate if the same mailbox name was used in multiple contexts. The problem still left was that not all places where messages were retrieved actually attempted to use this header for information when retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain work as expected. (closes issue #13853) Reported by: vicks1 Patches: 13853_v2.patch uploaded by mmichelson (license 60) Tested by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180466 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merged revisions 180383 via svnmerge from mmichelson2-7/+17
https://origsvn.digium.com/svn/asterisk/trunk ................ r180383 | mmichelson | 2009-03-05 13:14:14 -0600 (Thu, 05 Mar 2009) | 31 lines Merged revisions 180380 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180380 | mmichelson | 2009-03-05 12:58:48 -0600 (Thu, 05 Mar 2009) | 25 lines Fix broken mailbox parsing when searchcontexts option is enabled. When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180404 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Blocked revisions 180382 via svnmergemvanbaak0-0/+0
........ r180382 | mvanbaak | 2009-03-05 20:05:20 +0100 (Thu, 05 Mar 2009) | 2 lines Make sure we terminate the first s| command so we can actually produce correct files. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180384 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merged revisions 180373 via svnmerge from kpfleming3-36/+113
https://origsvn.digium.com/svn/asterisk/trunk ................ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines Merged revisions 180372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180377 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Blocked revisions 180369 via svnmergefile0-0/+0
........ r180369 | file | 2009-03-05 14:18:27 -0400 (Thu, 05 Mar 2009) | 13 lines Merge phase 1 support for the new bridging architecture. This commit brings in the bridging core, bridging technologies, and the ConfBridge application. For usage information on the ConfBridge application please see the output of "core show application ConfBridge" from the CLI. For API documentation please see the doxygen page describing the architecture and the documentation for each API call. Review: http://reviewboard.digium.com/r/93/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180370 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Blocked revisions 180261 via svnmergerussell0-0/+0
........ r180261 | russell | 2009-03-04 15:01:05 -0600 (Wed, 04 Mar 2009) | 54 lines Resolve object matching issues related to the removal of the sip_user object. Previously, chan_sip had both sip_peer and sip_user objects in memory. A patch went in to remove sip_user to simplify the code, since everything could be done with just sip_peer. This patch resolves some regressions found that were introduced by those changes. This code comes from svn/asterisk/team/group/sip-object-matching/. Here is a list of the changes that have been made: 1) When doing a match by name with the find_peer() function, make it much easier to specify which objects should be matched by having a parameter that specifies exactly which object types should be considered. Also, update find_by_name() to handle this parameter. Finally, update all code to use the new option values. 2) When looking up an object for an outbound request by name, consider peers only. (create_addr()) 3) Only match peers on an incoming registration request. 4) When doing authentication (except for SUBSCRIBE), look up users by name, instead of all objects by name. 5) When doing authentication (except for SUBSCRIBE), after looking for a user by name, look for a peer by IP address, instead of all objects by IP address. 6) When handling the SIP qualify CLI command or manager action, look for a peer by name, instead of any object by name. 7) When handling the SIP unregister CLI command, look for a peer by name, instead of any object by name. 9) In sip_do_debug_peer(), search for a peer by name, instead of any object by name. 9) When handling the SIPPEER() dialplan function, search for a peer by name, instead of any object by name. 10) In the following session timer related functions, st_get_se(), st_get_refresher(), and st_get_mode(), when looking for an object for a given sip_pvt using pvt->peername, look for a peer by name, instead of any object by name. 11) Fix build_peer() to properly handle the case where separate type=peer and type=user entries were specified in sip.conf. (closes issue #14505) Reported by: lmadsen Review: http://reviewboard.digium.com/r/172/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180262 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Merged revisions 180195 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) | 11 lines Merged revisions 180194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4 lines Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion. (issue #AST-194) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180196 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Blocked revisions 180155 via svnmergemmichelson0-0/+0
........ r180155 | mmichelson | 2009-03-04 11:03:32 -0600 (Wed, 04 Mar 2009) | 14 lines Allow for "magic" pickups to work when we wish to ignore the context When the subscription context for a call pickup subscription differs from the context of the call pickup target, there's not an easy way to divine what context should be used for the pickup. The way to work around this is to use PICKUPMARK as the context for the pickup. This has been documented in the sip.conf.sample file (ABE-1708) closes issue #14567 submitted by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180158 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Merged revisions 180120 via svnmerge from file1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r180120 | file | 2009-03-04 10:39:28 -0400 (Wed, 04 Mar 2009) | 7 lines Remove duplicate 'k' and 'K' Dial options. (closes issue #14601) Reported by: alecdavis Patches: app_dial.optionk.diff.txt uploaded by alecdavis (license 585) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180121 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Blocked revisions 180079 via svnmergemurf0-0/+0
........ r180079 | murf | 2009-03-03 16:35:26 -0700 (Tue, 03 Mar 2009) | 1 line My bad! left check_expr2 in the ALL_UTILS list by mistake. Already done to 1.6.x ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180081 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 180032 via svnmerge from dvossel4-14/+33
https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180078 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179973 via svnmerge from murf7-187/+349
https://origsvn.digium.com/svn/asterisk/trunk ................ r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) | 33 lines Merged revisions 179807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some work to do to port these changes to trunk; the check_expr stuff hasn't been updated here for quite some time, it appears. I added some more tests to the check_expr2 suite. I had to play around with the makefile a bit, etc. I added STANDALONE2 #ifdefs to ast_expr2.y so as not to conflict structure with aelparse. ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) | 19 lines These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text. I modified and added rules in ast_expr2.fl to better handle the concatenations. I added some default routines to ast_expr2.y so the standalone would compile. It also looks like I haven't run this thru bison since 2.1, so it's good to get this updated. The Makefile has comments added now for check_expr2 and check_expr to explain what they are for, and how to run them. The testexpr2s stuff has been removed, in favor of check_expr2. expr2.testinput has been updated to include the two expressions that inspired these changes (from mcnobody on #asterisk this morning) The regression has been run and all looks well. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180058 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 180007 via svnmerge from mmichelson2-0/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r180007 | mmichelson | 2009-03-03 16:49:07 -0600 (Tue, 03 Mar 2009) | 22 lines Merged revisions 180006 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180006 | mmichelson | 2009-03-03 16:48:18 -0600 (Tue, 03 Mar 2009) | 17 lines Clarify some documentation of queues.conf.sample It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180008 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Blocked revisions 179972 via svnmergedvossel0-0/+0
........ r179972 | dvossel | 2009-03-03 16:01:24 -0600 (Tue, 03 Mar 2009) | 13 lines app_meetme not setting filename and fileformat correctly for realtime When app_meetme finds a realtime conference, it doesn't get the filename and fileformat correctly when 'r' is set. Now app_meetme first checks to see if fileformat and filename are declared in the db, if they're not it checks the .conf file, if its not declared there either it then uses defaults. (closes issue #14545) Reported by: dalbaech Patches: app_meetme-realtime5.patch uploaded by dvossel (license 671) Realtime_Conference_Record_workaround.txt uploaded by dalbaech (license 705) Tested by: dvossel, dalbaech Review: http://reviewboard.digium.com/r/180/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@180005 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Fix a memory leak when updating a realtime member field.mmichelson1-4/+10
This was discovered while looking at issue #14353 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179971 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Blocked revisions 179937 via svnmergemmichelson0-0/+0
........ r179937 | mmichelson | 2009-03-03 14:59:16 -0600 (Tue, 03 Mar 2009) | 20 lines Add documentation for timing modules used in Asterisk This document specifies the timing modules available in Asterisk beginning with Asterisk 1.6.1. The document goes into detail about the differences between each and gives a general overview of what timing is used for in Asterisk. There is also a section which can be used to help customize your setup or to troubleshoot timing issues you may have. I also added messages to the DAHDI timing test used in res_timing_dahdi.c that points to this new documentation if people experience problems. Big thanks to all who contributed comments on this. (closes issue #14490) Reported by: mmichelson Patches: timing.txt uploaded by mmichelson (license 60) Review: http://reviewboard.digium.com/r/164/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179938 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Blocked revisions 179903 via svnmergerussell0-0/+0
........ r179903 | bmd | 2009-03-03 14:02:20 -0600 (Tue, 03 Mar 2009) | 1 line fix a leaked channel lock (and future deadlock) when we try to pick up our own channel ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179904 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179841 via svnmerge from file1-2/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | 16 lines Merged revisions 179840 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing. It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to. We can not safely modify it afterwards because of this, so don't even try. (closes issue #14564) Reported by: meric ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179842 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Blocked revisions 179745 via svnmergemmichelson0-0/+0
........ r179745 | mmichelson | 2009-03-03 11:03:47 -0600 (Tue, 03 Mar 2009) | 8 lines Convert pbx_spool to use string fields instead of statically-sized buffers. In tests run after making this conversion, I noticed an approximate 85% reduction in memory usage for call file processing. Review: http://reviewboard.digium.com/r/168/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179746 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179742 via svnmerge from russell1-1/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) | 14 lines Merged revisions 179741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179743 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179672 via svnmerge from file1-6/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | 10 lines Merged revisions 179671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines Move where fdno is set to the default value to *after* the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179673 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179609 via svnmerge from russell1-0/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) | 17 lines Merged revisions 179608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179610 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179537 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) | 21 lines Merged revisions 179536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179538 f38db490-d61c-443f-a65b-d21fe96a405b
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