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2009-08-18git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0.7-rc1@212958 ↵v1.6.0.7-rc1kpfleming6-12/+12
f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-06Use autotagged externalslmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0.7-rc1@180556 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-06Importing files for 1.6.0.7-rc1 releaselmadsen3-0/+50054
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0.7-rc1@180553 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-06Creating tag for the release of asterisk-1.6.0.7-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.0.7-rc1@180540 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
2009-03-02Merged revisions 179533 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r179533 | russell | 2009-03-02 17:36:38 -0600 (Mon, 02 Mar 2009) | 48 lines Merged revisions 179532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179532 | russell | 2009-03-02 17:34:13 -0600 (Mon, 02 Mar 2009) | 40 lines Move ast_waitfor() down to avoid the results of the API call becoming stale. This call to ast_waitfor() was being done way too soon in this section of code. Specifically, there was code in between the call to waitfor and the code that uses the result that puts the channel in autoservice. By putting the channel in autoservice, the previous results of ast_waitfor() become meaningless, as the autoservice thread will do it's own ast_waitfor() and ast_read() on the channel. So, when we came back out of autoservice and eventually hit the block of code that calls ast_read() on the channel, there may not actually be any input on the channel available. Even though the previous call to ast_waitfor() in app_meetme said there was input, the autoservice thread has since serviced the channel for some period of time. This bug manifested itself while dvossel was doing some testing of MeetMe in Asterisk trunk. He was using the timerfd timing module. When the code hit ast_read() erroneously, it determined that it must have been called because of input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was the cause of the last legitimate call to ast_read() done by autoservice. In this test, an IAX2 channel was calling into the MeetMe conference. It was _much_ more likely to be seen with an IAX2 channel because of the way audio is handled. Every audio frame that comes in results in a call to ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify the channel thread that a frame is waiting to be handled. So, the chances of ast_waitfor() indicating that a channel needs servicing due to a timer event on an IAX2 event is very high. Finally, it is interesting to note that if a different timing interface was being used, this bug would probably not be noticed. When ast_read() is called and erroneously thinks that there is a timer event to handle, it calls the ast_timer_ack() function. The pthread and dahdi timing modules handle the ack() function being called when there is no event by simply ignoring it. In the case of the timerfd module, it results in a read() on the timer fd that will block forever, as there is no data to read. This caused Asterisk to lock up very quickly. Thanks to dvossel and mmichelson for the fun debugging session. :-) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179534 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 151464 via svnmerge from mmichelson1-12/+21
https://origsvn.digium.com/svn/asterisk/trunk ........ r151464 | mmichelson | 2008-10-21 18:54:41 -0500 (Tue, 21 Oct 2008) | 11 lines Make the sip_standard_port function more granular by allowing separate type and port arguments. This is necessary because when building our From and Contact headers, we need to be absolutely sure that we are placing our source port there and not the peer's source port. (closes issue #12761) Reported by: asbestoshead Patches: patch-chan-sip-contact-port.txt uploaded by asbestoshead (license 455) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179473 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 179469 via svnmerge from tilghman1-1/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) | 17 lines Merged revisions 179468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179470 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Blocked revisions 179465 via svnmergerussell0-0/+0
........ r179465 | russell | 2009-03-02 17:06:16 -0600 (Mon, 02 Mar 2009) | 4 lines Fix a reference leak in timerfd_set_rate(). (found during a debugging session with dvossel and mmichelson.) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179466 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 179462 via svnmerge from russell1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) | 16 lines Merged revisions 179461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179463 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 179396 via svnmerge from qwell4-5/+129
https://origsvn.digium.com/svn/asterisk/trunk ................ r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | 9 lines Merged revisions 179395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat. (closes issue #14264) Reported by: dimas ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179402 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02KeepAlive application no longer exists, so fix gosub implementation to not ↵tilghman1-18/+4
use it. (closes issue #14571) Reported by: zktech Patches: 20090302__bug14571.diff.txt uploaded by tilghman (license 14) Tested by: tilghman git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179363 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02If cdr registration somehow succeeds without a config file, don't crash.tilghman1-0/+5
(closes issue #14563) Reported by: alerios git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179360 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Blocked revisions 179323 via svnmergefile0-0/+0
........ r179323 | file | 2009-03-02 10:28:09 -0400 (Mon, 02 Mar 2009) | 5 lines Do not try to remove a registration scheduled item if the scheduler context has already been destroyed. (closes issue #14580) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179324 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Blocked revisions 179291 via svnmergefile0-0/+0
........ r179291 | file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines Fix issue where changing the volume of both directions of audio did not work. (closes issue #14574) Reported by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK (license 545) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179292 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-01Blocked revisions 179254 via svnmergemmichelson0-0/+0
........ r179254 | mmichelson | 2009-03-01 17:25:23 -0600 (Sun, 01 Mar 2009) | 5 lines Swap reversed timevals. This was pointed out by ScribbleJ in #asterisk-dev. Thanks very much, ScribbleJ! ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179255 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-01Add error checking when updating the "paused" field of a realtime queue member.mmichelson1-3/+14
This code already existed in trunk and 1.6.1, but was not in 1.6.0 prior to this commit. (closes issue #14338) Reported by: fiddur Patches: 14338.patch uploaded by mmichelson (license 60) Tested by: fiddur git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179222 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-01Merged revisions 179219 via svnmerge from mmichelson1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r179219 | mmichelson | 2009-03-01 15:45:08 -0600 (Sun, 01 Mar 2009) | 18 lines Properly free memory and remove scheduler entries when a transmission failure occurs. Previously, only the "data" field of the sip_pkt created during __sip_reliable_xmit was freed when XMIT_FAILURE was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_FAILURE is a condition meaning that we don't want to attempt resending the packet at all. The proper action to take is to remove the scheduler entry we just created, free the packet's data as well as the packet itself, and unlink it from the list of packets on the sip_pvt structure. (closes issue #14455) Reported by: Nick_Lewis Patches: 14455.patch uploaded by mmichelson (license 60) Tested by: Nick_Lewis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179220 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Blocked revisions 179164 via svnmergerussell0-0/+0
........ r179164 | russell | 2009-02-27 15:47:18 -0600 (Fri, 27 Feb 2009) | 2 lines Mark res_ais as experimental, as the binary event format is subject to change. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179165 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Merged revisions 179161 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r179161 | tilghman | 2009-02-27 15:32:13 -0600 (Fri, 27 Feb 2009) | 3 lines If config file is blank, don't load module. (Closes issue #14563) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179162 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Blocked revisions 179154 via svnmergerussell0-0/+0
........ r179154 | russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines Add a note about the ordering of entries in sip.conf in 1.6.1. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179159 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Blocked revisions 179122 via svnmergemvanbaak0-0/+0
........ r179122 | mvanbaak | 2009-02-27 21:34:00 +0100 (Fri, 27 Feb 2009) | 16 lines Add reload support to chan_skinny. Special thanks goes to DEA who had to redo this patch twice because we first put unload/load support in and later redid the way we configure devices and lines. (closes issue #10297) Reported by: DEA Patches: skinny-reload-trunkv2.diff uploaded by wedhorn (license 30) skinny-reload-trunk-v4.txt uploaded by DEA (license 3) With mods by me based on feedback from wedhorn and Russell and seanbright Tested by: DEA, mvanbaak, pj Review: http://reviewboard.digium.com/r/130/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179123 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Merged revisions 179057 via svnmerge from qwell1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r179057 | qwell | 2009-02-27 13:04:57 -0600 (Fri, 27 Feb 2009) | 8 lines Update documentation for DIALEDTIME and ANSWEREDTIME variables. (closes issue #14566) Reported by: klaus3000 Patches: ANSWEREDTIME-1.4-patch.txt uploaded by klaus3000 (license 65) ANSWEREDTIME-trunk-patch.txt uploaded by klaus3000 (license 65) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179058 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Blocked revisions 179021 via svnmergerussell0-0/+0
........ r179021 | russell | 2009-02-27 09:51:56 -0600 (Fri, 27 Feb 2009) | 7 lines Fix downloading SIREN7 and SIREN14 sound packages. In passing, also fix downloading SLIN16 extra sound packages. (closes issue #14565) Reported by: jtodd ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@179022 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Merged revisions 178986 via svnmerge from murf2-3/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines Merged revisions 178956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@178987 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Merged revisions 178871 via svnmerge from dvossel1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r178871 | dvossel | 2009-02-26 11:46:12 -0600 (Thu, 26 Feb 2009) | 6 lines IAX2 prune realtime, minor tweak to last fix A return statement was missing which caused unexpected cli output. issue #14479 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@178874 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Blocked revisions 178870 via svnmergemurf0-0/+0
........ r178870 | murf | 2009-02-26 10:45:22 -0700 (Thu, 26 Feb 2009) | 1 line These small fixes prevent compiler warnings with ubuntu 8.10's gcc-4.3.2, which tend to break my dev-mode build. Not a problem in 1.6.x. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@178872 f38db490-d61c-443f-a65b-d21fe96a405b
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