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2009-08-18git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc2@212958 ↵v1.6.1.0-rc2kpfleming6-13/+13
f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Use autotagged externalslmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc2@180303 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Importing files for 1.6.1.0-rc2 releaselmadsen3-0/+54442
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc2@180302 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Creating tag for the release of asterisk-1.6.1.0-rc2lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.6.1.0-rc2@180301 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Merged revisions 180261 via svnmerge from russell1-62/+73
https://origsvn.digium.com/svn/asterisk/trunk ........ 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/1.6.1@180263 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Blocked revisions 180259 via svnmergetilghman0-0/+0
........ r180259 | tilghman | 2009-03-04 14:48:42 -0600 (Wed, 04 Mar 2009) | 2 lines Spacing changes only ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@180260 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/1.6.1@180197 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/1.6.1@180187 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/1.6.1@180122 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/1.6.1@180082 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/1.6.1@180080 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179973 via svnmerge from murf7-193/+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/1.6.1@180077 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/1.6.1@180009 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/1.6.1@179995 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179937 via svnmerge from mmichelson2-3/+95
https://origsvn.digium.com/svn/asterisk/trunk ........ 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/1.6.1@179939 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179903 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ 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/1.6.1@179905 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/1.6.1@179843 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/1.6.1@179747 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/1.6.1@179744 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/1.6.1@179674 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/1.6.1@179611 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/1.6.1@179539 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/1.6.1@179535 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/1.6.1@179471 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/1.6.1@179467 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 179462 via svnmerge from russell1-0/+5
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/1.6.1@179464 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/1.6.1@179407 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 179361 via svnmerge from tilghman1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r179361 | tilghman | 2009-03-02 11:18:48 -0600 (Mon, 02 Mar 2009) | 2 lines Backport 1.6.0 fix to trunk (failsafe if db is not loaded) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@179362 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/1.6.1@179325 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 179291 via svnmerge from file1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ 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/1.6.1@179293 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-01Merged revisions 179254 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ 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/1.6.1@179256 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-01Merged revisions 179219 via svnmerge from mmichelson1-2/+4
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_ERROR was returned by __sip_xmit. When retrans_pkt was called, this inevitably resulted in the reading and writing of freed memory. XMIT_ERROR 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/1.6.1@179221 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Merged revisions 179164 via svnmerge from russell3-0/+23
https://origsvn.digium.com/svn/asterisk/trunk ........ 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/1.6.1@179166 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/1.6.1@179163 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Merged revisions 179154 via svnmerge from russell1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ 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/1.6.1@179160 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/1.6.1@179124 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/1.6.1@179059 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/1.6.1@179023 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/1.6.1@178988 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Blocked revisions 178919 via svnmergetilghman0-0/+0
........ r178919 | tilghman | 2009-02-26 12:41:28 -0600 (Thu, 26 Feb 2009) | 8 lines Sound confirmation of call pickup success. (closes issue #13826) Reported by: azielke Patches: pickupsound2-trunk.patch uploaded by azielke (license 548) __20081124_bug_13826_updated.patch uploaded by lmadsen (license 10) Tested by: lmadsen ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178920 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/1.6.1@178875 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/1.6.1@178873 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Merged revisions 178828 via svnmerge from murf1-2/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines Merged revisions 178804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines This patch prevents the feature detection timeout from being cut in half. Because the ast_channel_bridge() call will return 0 and pass a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer field in hte config struct is getting decremented twice, which effectively cuts the digittimeout in half. I added conditions to the if statement to only let DTMF_END frames to flow thru, which solved the problem. Also, when the frame pointer is null, let control flow thru-- this usually happens on timeouts. I added a comment to the code to explain what's going on and why. Many thanks to sodom for reporting this problem. Personnally, it always seemed like something was wrong with the featuredigittimeout, but I never could quite decide what... and was too busy to investigate. This bug forced the issue, and now we know. Sodom had other issues in 14515, but I couldn't reproduce them. If he still has problems, and wants to get them solved, he is welcome to reopen 14515. (closes issue #14515) Reported by: sodom Patches: 14515.patch uploaded by murf (license 17) Tested by: murf, sodom ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178869 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Merged revisions 178801 via svnmerge from file1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r178801 | file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed. (closes issue #14541) Reported by: grant ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178803 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Merged revisions 178767 via svnmerge from dvossel1-16/+43
https://origsvn.digium.com/svn/asterisk/trunk ........ r178767 | dvossel | 2009-02-26 09:50:22 -0600 (Thu, 26 Feb 2009) | 8 lines IAX2 prune realtime fix Iax2 prune realtime had issues. If "iax2 prune realtime all" was called, it would appear like the command was successful, but in reality nothing happened. This is because the reload that was supposed to take place checks the config files, sees no changes, and does nothing. If there had been a change in the the config file, the realtime users would have been marked for deletion and everything would have been fine. Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain. (closes issue #14479) Reported by: mousepad99 Review: http://reviewboard.digium.com/r/176/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178769 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Blocked revisions 178764 via svnmergefile0-0/+0
........ r178764 | file | 2009-02-26 11:40:10 -0400 (Thu, 26 Feb 2009) | 5 lines Ensure there is a valid tone part before trying to play tones. (closes issue #14558) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178766 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-25Blocked revisions 178607 via svnmergetilghman0-0/+0
........ r178607 | tilghman | 2009-02-25 13:49:46 -0600 (Wed, 25 Feb 2009) | 2 lines Picky, picky buildbots ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178608 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-25Blocked revisions 178605 via svnmergetilghman0-0/+0
........ r178605 | tilghman | 2009-02-25 13:24:44 -0600 (Wed, 25 Feb 2009) | 9 lines Use notification when timezone files change and re-scan then. (closes issue #14300) Reported by: jamessan Patches: 20090127__bug14300.diff.txt uploaded by tilghman (license 14) 20090224__bug14300.diff uploaded by jamessan (license 246) Tested by: jamessan Review: http://reviewboard.digium.com/r/136/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178606 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-25Blocked revisions 178573 via svnmergetilghman0-0/+0
........ r178573 | tilghman | 2009-02-25 13:03:35 -0600 (Wed, 25 Feb 2009) | 2 lines Oops, wrong direction of command ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178574 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-25Merged revisions 178509 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) | 10 lines Merged revisions 178508 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines Update the copyright year for the main page of the doxygen documentation. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178511 f38db490-d61c-443f-a65b-d21fe96a405b
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