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2010-01-24Merged revisions 242521 via svnmerge from tilghman1-21/+30
https://origsvn.digium.com/svn/asterisk/trunk ................ r242521 | tilghman | 2010-01-24 00:40:31 -0600 (Sun, 24 Jan 2010) | 15 lines Merged revisions 242520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) | 8 lines Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script. Changed after discussion on the -dev list about possible unnecessary build failures, due to checkouts/untars causing these special source files to possibly be newer than their resulting C files. This should additionally ensure that nobody need learn about extra Makefile arguments to ensure the proper files get rebuilt when changes are made to these special source files. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@242524 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-08Merged revisions 238635 via svnmerge from dvossel1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r238635 | dvossel | 2010-01-08 13:39:30 -0600 (Fri, 08 Jan 2010) | 22 lines fixes AUDIOHOOK_INHERIT regression During the process of removing an audiohook from one channel and attaching it to another the audiohook's status is updated to DONE and then back to whatever it was previously. Typically updating the status after setting it to DONE is not a good idea because DONE can trigger unrecoverable audiohook destruction events... because of this a conditional check was added to audiohook_update_status to explicitly prevent the audiohook from ever changing after being set to DONE. It was this check that prevented audiohook inherit from work properly though. Now ast_audiohook_move_by_source is treated as a special exception, as the audiohook must be returned to its previous status after attaching it to the new channel. This is only a safe operation because the audiohook's lock is held the entire time, otherwise this could cause trouble. (closes issue #16522) Reported by: corruptor ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@238639 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07Merged revisions 209400 via svnmerge from tilghman1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r209400 | kpfleming | 2009-07-28 08:49:46 -0500 (Tue, 28 Jul 2009) | 3 lines Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files. (closes issue #16251) Reported by: asgaroth ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@238494 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-04Merged revisions 237406 via svnmerge from tilghman1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r237406 | tilghman | 2010-01-04 12:28:28 -0600 (Mon, 04 Jan 2010) | 23 lines Merged revisions 237405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines Add a flag to disable the Background behavior, for AGI users. This is in a section of code that relates to two other issues, namely issue #14011 and issue #14940), one of which was the behavior of Background when called with a context argument that matched the current context. This fix broke FreePBX, however, in a post-Dial situation. Needless to say, this is an extremely difficult collision of several different issues. While the use of an exception flag is ugly, fixing all of the issues linked is rather difficult (although if someone would like to propose a better solution, we're happy to entertain that suggestion). (closes issue #16434) Reported by: rickead2000 Patches: 20091217__issue16434.diff.txt uploaded by tilghman (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: rickead2000 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@237407 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-28Merged revisions 236613 via svnmerge from seanbright2-32/+32
https://origsvn.digium.com/svn/asterisk/trunk ................ r236613 | seanbright | 2009-12-28 10:22:54 -0500 (Mon, 28 Dec 2009) | 14 lines Merged revisions 236585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236585 | seanbright | 2009-12-28 10:12:08 -0500 (Mon, 28 Dec 2009) | 7 lines Try a test compile to see if PTHREAD_ONCE_INIT requires extra braces. There was conditional code (based on build platform) to optioinally wrap PTHREAD_ONCE_INIT in braces that was removed since it is fixed in newer versions of Solaris/OpenSolaris, but I am still running into it on Solaris 10 x86 so add a configure-time check for it. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@236633 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-18Merged revisions 235660 via svnmerge from jpeeler1-5/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r235660 | jpeeler | 2009-12-18 16:51:37 -0600 (Fri, 18 Dec 2009) | 55 lines Merged revisions 235635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235635 | jpeeler | 2009-12-18 16:29:51 -0600 (Fri, 18 Dec 2009) | 48 lines Correct CDR dispositions for BUSY/FAILED This patch is simple in that it reorders the disposition defines so that the fix for issue 12946 works properly (the default CDR disposition was changed to AST_CDR_NOANSWER). Also, the AST_CDR_FLAG_ORIGINATED flag was set in ast_call to ensure all CDR records are written. The side effects of CDR changes are scary, so I'm documenting the test cases performed to attempt to catch any regressions. The following tests were all performed using 1.4 rev 195881 vs head (235571) + patch: A calls B C calls B (busy) Hangup C Hangup A (Both SIP and features) A calls B A blind transfers to C Hangup C (Both SIP and features) A calls B A attended transfers to C Hangup C A calls B A attended transfers to C (SIP) C blind transfers to A (features) Hangup A All of the test scenario CDRs matched. The following tests were performed just with the patch to ensure proper operation (with unanswered=yes): exten =>s,1,Answer exten =>s,n,ResetCDR(w) exten =>s,n,ResetCDR(w) exten =>s,1,ResetCDR(w) exten =>s,n,ResetCDR(w) (closes issue #16180) Reported by: aatef Patches: bug16180.patch uploaded by jpeeler (license 325) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@235662 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-18Merged revisions 235573 via svnmerge from tilghman1-17/+29
https://origsvn.digium.com/svn/asterisk/trunk ................ r235573 | tilghman | 2009-12-18 15:19:43 -0600 (Fri, 18 Dec 2009) | 9 lines Merged revisions 235572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235572 | tilghman | 2009-12-18 15:18:16 -0600 (Fri, 18 Dec 2009) | 2 lines Point to the typical missing package, not the cryptic "termcap support". ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@235574 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-09Set a module load priority for format modules.russell1-0/+8
A recent change to app_voicemail made it such that the module now assumes that all format modules are available while processing voicemail configuration. However, when autoloading modules, it was possible that app_voicemail was loaded before the format modules. Since format modules don't depend on anything, set a module load priority on them to ensure that they get loaded first when autoloading. This version of the patch is specific to Asterisk 1.4 and 1.6.0. These versions did not already support module load priority in the module API. This adds a trivial version of this which is just a module flag to include it in a pass before loading "everything". Thanks to mmichelson for the review! (closes issue #16412) Reported by: jiddings Tested by: russell Review: https://reviewboard.asterisk.org/r/445/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@233841 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30Merged revisions 231692 via svnmerge from kpfleming1-2/+17
https://origsvn.digium.com/svn/asterisk/trunk ........ r231692 | kpfleming | 2009-11-30 15:47:42 -0600 (Mon, 30 Nov 2009) | 22 lines Another round of UDPTL stack fixes/improvements: 1) Allow users of UDPTL stack to associate a character-string tag with a UDPTL session, so that log/error/debug messages generated by the UDPTL stack can be 'connected' to the endpoint that caused them to be generated. 2) Improve comments (and process) of calculating the far end's maximum IFP size when redundancy mode is in use for error correction. 3) When an IFP larger than the calculated 'far max IFP' size is presented for writing, truncate it rather than putting in the buffer and allowing the buffer to overflow; this will cause the ends to retrain to a lower bit rate that produces IFPs of an appropriate size if possible, and if not possible, the FAX transfer will fail completely. In these cases, it is due to the one endpoint supplying a T38FaxMaxDatagram value that is improperly calculated and is too low to be of use; we have configuration options available to override this behavior. 4) Eliminate use of T38FaxMaxDatagram value in udptl.conf; it is no longer needed. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@231693 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-30Merged revisions 231688 via svnmerge from mnicholson1-0/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r231688 | mnicholson | 2009-11-30 15:31:55 -0600 (Mon, 30 Nov 2009) | 15 lines Merged revisions 231614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 | mnicholson | 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) | 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@231691 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-20Merged revisions 230583 via svnmerge from dvossel1-0/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r230583 | dvossel | 2009-11-20 11:26:20 -0600 (Fri, 20 Nov 2009) | 6 lines audiohook signal trigger on every status change (issue #14618) Review: https://reviewboard.asterisk.org/r/434/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@230587 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-09Merged revisions 228858 via svnmerge from mnicholson1-9/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r228858 | mnicholson | 2009-11-09 08:37:07 -0600 (Mon, 09 Nov 2009) | 15 lines Merged revisions 228827 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r228827 | mnicholson | 2009-11-09 08:16:03 -0600 (Mon, 09 Nov 2009) | 8 lines Perform limited bounds checking when destroying ast_mutex_t structures to make sure we don't try to use negative indices. (closes issue #15588) Reported by: zerohalo Patches: 20090820__issue15588.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@228861 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-28Merged revisions 226305 via svnmerge from tilghman1-5/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r226305 | tilghman | 2009-10-28 13:04:05 -0500 (Wed, 28 Oct 2009) | 9 lines Merged revisions 226304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r226304 | tilghman | 2009-10-28 13:02:25 -0500 (Wed, 28 Oct 2009) | 2 lines Fix documentation (pointed out by TheDavidFactor on #-dev) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@226306 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-22Merged revisions 225360 via svnmerge from tilghman1-4/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r225360 | tilghman | 2009-10-22 12:11:23 -0500 (Thu, 22 Oct 2009) | 11 lines Merged revisions 225105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@225361 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Merged revisions 224932 via svnmerge from russell3-35/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r224932 | russell | 2009-10-20 22:09:04 -0500 (Tue, 20 Oct 2009) | 12 lines Merged revisions 224931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@224933 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-08Merged revisions 222880 via svnmerge from russell2-19/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r222880 | russell | 2009-10-08 14:52:03 -0500 (Thu, 08 Oct 2009) | 51 lines Merged revisions 222878 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222878 | russell | 2009-10-08 14:45:47 -0500 (Thu, 08 Oct 2009) | 44 lines Make filestream frame handling safer by isolating frames before returning them. This patch is related to a number of issues on the bug tracker that show crashes related to freeing frames that came from a filestream. A number of fixes have been made over time while trying to figure out these problems, but there re still people seeing the crash. (Note that some of these bug reports include information about other problems. I am specifically addressing the filestream frame crash here.) I'm still not clear on what the exact problem is. However, what is _very_ clear is that we have seen quite a few problems over time related to unexpected behavior when we try to use embedded frames as an optimization. In some cases, this optimization doesn't really provide much due to improvements made in other areas. In this case, the patch modifies filestream handling such that the embedded frame will not be returned. ast_frisolate() is used to ensure that we end up with a completely mallocd frame. In reality, though, we will not actually have to malloc every time. For filestreams, the frame will almost always be allocated and freed in the same thread. That means that the thread local frame cache will be used. So, going this route doesn't hurt. With this patch in place, some people have reported success in not seeing the crash anymore. (SWP-150) (AST-208) (ABE-1834) (issue #15609) Reported by: aragon Patches: filestream_frisolate-1.4.diff2.txt uploaded by russell (license 2) Tested by: aragon, russell (closes issue #15817) Reported by: zerohalo Tested by: zerohalo (closes issue #15845) Reported by: marhbere Review: https://reviewboard.asterisk.org/r/386/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222881 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-08Merged revisions 222873 via svnmerge from dvossel1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r222873 | dvossel | 2009-10-08 14:35:30 -0500 (Thu, 08 Oct 2009) | 6 lines fixes an ast_netsock_list memory leak. ABE-1998 Review: https://reviewboard.asterisk.org/r/395/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222876 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-06Merged revisions 222176 via svnmerge from kpfleming1-10/+56
https://origsvn.digium.com/svn/asterisk/trunk ................ r222176 | kpfleming | 2009-10-05 20:24:24 -0500 (Mon, 05 Oct 2009) | 27 lines Recorded merge of revisions 222152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r222152 | kpfleming | 2009-10-05 20:16:36 -0500 (Mon, 05 Oct 2009) | 20 lines Fix ao2_iterator API to hold references to containers being iterated. See Mantis issue for details of what prompted this change. Additional notes: This patch changes the ao2_iterator API in two ways: F_AO2I_DONTLOCK has become an enum instead of a macro, with a name that fits our naming policy; also, it is now necessary to call ao2_iterator_destroy() on any iterator that has been created. Currently this only releases the reference to the container being iterated, but in the future this could also release other resources used by the iterator, if the iterator implementation changes to use additional resources. (closes issue #15987) Reported by: kpfleming Review: https://reviewboard.asterisk.org/r/383/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@222185 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-30Merged revisions 221266 via svnmerge from twilson1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines Merged revisions 221086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines Change the SSRC by default when our media stream changes Be default, change SSRC when doing an audio stream changes Asterisk doesn't honor marker bit when reinvited to already-bridged RTP streams,resulting in far-end stack discarding packets with "old" timestamps that areactually part of a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a reinvite, unless the 'constantssrc' is set to true in sip.conf. The original issue reported to Digium support detailed the following situation: ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in fromITSP, Asterisk dials the app server which sends a re-invite back toAsterisk--not to negotiate to send media directly to the ITSP, but to indicatethat it's changing the stream it's sending to Asterisk. The app servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker bit on the new stream. Asterisk passes through the teimstamp of the new stream, butdoes not reset the SSRC, sequence numbers, or set the marker bit. When the timestamp on the new stream is older than the timestamp on the originalstream, the ITSP (which doesn't know there has been any change) discards the newframes because it thinks they are too old. This patch addresses this by changing the SSRC on a stream update unless constantssrc=true is set in sip.conf. Review: https://reviewboard.asterisk.org/r/374/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@221301 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-26Merged revisions 220586 via svnmerge from tilghman1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r220586 | tilghman | 2009-09-26 10:10:28 -0500 (Sat, 26 Sep 2009) | 2 lines Allow AES to compile, when OpenSSL is not present. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@220587 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-17Merged revisions 219139 via svnmerge from mnicholson2-0/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r219139 | mnicholson | 2009-09-17 10:18:01 -0500 (Thu, 17 Sep 2009) | 17 lines Merged revisions 219136 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r219136 | mnicholson | 2009-09-17 09:58:39 -0500 (Thu, 17 Sep 2009) | 10 lines Prevent a potential race condition and crash when hanging up a channel by removing the channel from the channel list before begining channel tear down. This fix may potentially cause problems with CDR backends that access the channel a CDR is associated with via the channel list. This fix makes the channel unavabile at the time when the CDR backend is invoked. This has been documented in include/asterisk/cdr.h. (closes issue #15316) Reported by: vmarrone Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/362/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@219198 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-08Merged revisions 217074 via svnmerge from kpfleming1-29/+17
https://origsvn.digium.com/svn/asterisk/trunk ........ r217074 | kpfleming | 2009-09-08 11:37:28 -0500 (Tue, 08 Sep 2009) | 9 lines Ensure that the default autoconf CFLAGS are not used. A recent change to the configure script that allows the user to specify CFLAGS and/or LDFLAGS to the script had the unfortunate side effect of letting autoconf's default CFLAGS (-g -O2) feed in to the rest of the build system, thereby overriding the DONT_OPTIMIZE setting in menuselect. That problem is now corrected. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@217075 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-03Merged revisions 215955 via svnmerge from dvossel2-0/+12
https://origsvn.digium.com/svn/asterisk/trunk ........ r215955 | dvossel | 2009-09-03 11:31:54 -0500 (Thu, 03 Sep 2009) | 6 lines Merge code associated with AST-2009-006 (closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@216003 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-30Merged revisions 214819 via svnmerge from tilghman1-17/+32
https://origsvn.digium.com/svn/asterisk/trunk ........ r214819 | tilghman | 2009-08-30 01:43:04 -0500 (Sun, 30 Aug 2009) | 4 lines If lua is detected with the lua5.1 prefix (or not), adjust the include path accordingly. Based upon feedback to a release announcement on the -users list. See http://lists.digium.com/pipermail/asterisk-users/2009-August/236954.html ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@214820 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-28Merged revisions 214696 via svnmerge from kpfleming1-29/+17
https://origsvn.digium.com/svn/asterisk/trunk ........ r214696 | kpfleming | 2009-08-28 15:01:21 -0500 (Fri, 28 Aug 2009) | 9 lines Ensure that CFLAGS and/or LDFLAGS provided to configure script are preserved. Cross-compilation environments want to provide 'defaults' for compiler and linker options, and frequently do this by specifying CFLAGS and LDFLAGS in the environment or as command-line arguments to the configure script. This patch modifies the configure script and Makefile to preserve these settings and ensure they are used in the build process. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@214697 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-27Merged revisions 214518 via svnmerge from tilghman1-1/+55
https://origsvn.digium.com/svn/asterisk/trunk ................ r214518 | tilghman | 2009-08-27 16:46:46 -0500 (Thu, 27 Aug 2009) | 14 lines Merged revisions 214517 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214517 | tilghman | 2009-08-27 16:45:34 -0500 (Thu, 27 Aug 2009) | 7 lines Use autoconf to detect libcurl, as this enables cross-compilation checks, something we didn't allow before. (closes issue #15714) Reported by: pprindeville Patches: 20090813__issue15714.diff.txt uploaded by tilghman (license 14) Tested by: pprindeville ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@214519 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-27Merged revisions 214466 via svnmerge from tilghman1-277/+183
https://origsvn.digium.com/svn/asterisk/trunk ................ r214466 | tilghman | 2009-08-27 12:28:01 -0500 (Thu, 27 Aug 2009) | 9 lines Merged revisions 214436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214436 | tilghman | 2009-08-27 11:53:58 -0500 (Thu, 27 Aug 2009) | 2 lines One more build system change, to make the descriptions look better, if we have better information. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@214474 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-27Merged revisions 214360 via svnmerge from tilghman1-182/+182
https://origsvn.digium.com/svn/asterisk/trunk ................ r214360 | tilghman | 2009-08-27 11:12:03 -0500 (Thu, 27 Aug 2009) | 10 lines Merged revisions 214357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r214357 | tilghman | 2009-08-27 11:03:50 -0500 (Thu, 27 Aug 2009) | 3 lines Make autoheader descriptions render correctly in our autoconfig.h file. (Figured out while working with issue #14906) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@214361 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-25Merged revisions 214152 via svnmerge from tilghman1-17/+29
https://origsvn.digium.com/svn/asterisk/trunk ........ r214152 | tilghman | 2009-08-25 17:39:51 -0500 (Tue, 25 Aug 2009) | 4 lines Not all versions of gnu-linux use glibc, which contains iconv. Some (especially embedded systems) don't have iconv at all. (closes issue #15169) Reported by: pprindeville ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@214153 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-21Merged revisions 213560 via svnmerge from tilghman1-0/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r213560 | tilghman | 2009-08-21 11:53:52 -0500 (Fri, 21 Aug 2009) | 14 lines Merged revisions 213559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r213559 | tilghman | 2009-08-21 11:52:53 -0500 (Fri, 21 Aug 2009) | 7 lines Permit DEBUG_FD_LEAKS to be used with C++ source files. (closes issue #15698) Reported by: slavon Patches: 20090817__issue15698.diff.txt uploaded by tilghman (license 14) Tested by: slavon, tilghman ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@213561 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-06Merged revisions 210908 via svnmerge from tilghman1-3/+9
https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 | tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines Allow Gosub to recognize quote delimiters without consuming them. (closes issue #15557) Reported by: rain Patches: 20090723__issue15557.diff.txt uploaded by tilghman (license 14) Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@210909 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-30Merged revisions 209554 via svnmerge from dbrooks1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines Fixes numerous spelling errors. Patch submitted by alecdavis. (closes issue #15595) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@209555 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-27Merged revisions 209098 via svnmerge from dbrooks1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 | dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines Fixing typos. Replaces "recieved" with "received" and "initilize" with "initialize" (closes issue #15571) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@209221 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-24Merged revisions 208548 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 | kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8 lines Resolve a T.38 negotiation issue left over from the udptl-updates merge. The udptl-updates branch that was merged yesterday failed to properly send back T.38 SDP responses with the correct error correction mode, if the incoming SDP from the other end caused us to change error correction modes. This patch corrects that situation. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@208549 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-23Merged revisions 208464 via svnmerge from kpfleming2-31/+27
https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@208468 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-09Merged revisions 205696 via svnmerge from kpfleming1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover. Recent changes in T.38 negotiation in Asterisk caused these applications to not respond when the other endpoint initiated a switchover to T.38; this resulted in the T.38 switchover failing, and the FAX attempt to be made using an audio connection, instead of T.38 (which would usually cause the FAX to fail completely). This patch corrects this problem, and the applications will now correctly respond to the T.38 switchover request. In addition, the response will include the appopriate T.38 session parameters based on what the other end offered and what our end is capable of. (closes issue #14849) Reported by: afosorio ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205697 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-09Merged revisions 205600 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r205600 | dvossel | 2009-07-09 11:19:09 -0500 (Thu, 09 Jul 2009) | 9 lines Merged revisions 205599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205599 | dvossel | 2009-07-09 11:18:09 -0500 (Thu, 09 Jul 2009) | 2 lines Changing ast_samp2tv to not use floating point. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205608 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-09Merged revisions 205479 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r205479 | dvossel | 2009-07-08 18:19:09 -0500 (Wed, 08 Jul 2009) | 16 lines Merged revisions 205471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205471 | dvossel | 2009-07-08 18:15:54 -0500 (Wed, 08 Jul 2009) | 10 lines Fixes 8khz assumptions Many calculations assume 8khz is the codec rate. This is not always the case. This patch only addresses chan_iax.c and res_rtp_asterisk.c, but I am sure there are other areas that make this assumption as well. Review: https://reviewboard.asterisk.org/r/306/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205597 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08Merged revisions 205412 via svnmerge from dvossel2-9/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r205412 | dvossel | 2009-07-08 17:15:06 -0500 (Wed, 08 Jul 2009) | 12 lines Merged revisions 205409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines moving ast_devstate_to_extenstate to pbx.c from devicestate.c ast_devstate_to_extenstate belongs in pbx.c. This change fixes a compile time error with chan_vpb as well. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205415 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08Merged revisions 205216 via svnmerge from dvossel1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r205216 | dvossel | 2009-07-08 11:54:24 -0500 (Wed, 08 Jul 2009) | 17 lines Merged revisions 205215 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r205215 | dvossel | 2009-07-08 11:53:40 -0500 (Wed, 08 Jul 2009) | 10 lines ast_samp2tv needs floating point for 16khz audio In ast_samp2tv(), (1000000 / _rate) = 62.5 when _rate is 16000. The .5 is currently stripped off because we don't calculate using floating points. This causes madness with 16khz audio. (issue ABE-1899) Review: https://reviewboard.asterisk.org/r/305/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205220 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08Merged revisions 205120 via svnmerge from russell1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r205120 | russell | 2009-07-08 10:17:19 -0500 (Wed, 08 Jul 2009) | 16 lines Move OpenSSL initialization to a single place, make library usage thread-safe. While doing some reading about OpenSSL, I noticed a couple of things that needed to be improved with our usage of OpenSSL. 1) We had initialization of the library done in multiple modules. This has now been moved to a core function that gets executed during Asterisk startup. We already link OpenSSL into the core for TCP/TLS functionality, so this was the most logical place to do it. 2) OpenSSL is not thread-safe by default. However, making it thread safe is very easy. We just have to provide a couple of callbacks. One callback returns a thread ID. The other handles locking. For more information, start with the "Is OpenSSL thread-safe?" question on the FAQ page of openssl.org. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205139 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-08SIP Dialog ref countingdvossel1-0/+18
This patch adds reference counting for sip dialogs into 1.6.0. When proc_session_timer() is called from the scheduler thread it has no guarantee the session timer's dialog won't be freed from underneath it. Now the session timer holds a reference to the dialog, preventing it from being destroyed during the middle of proc_session_timer(). (closes issue #13623) Reported by: Nik Soggia Review: https://reviewboard.asterisk.org/r/302/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@205117 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-02Merged revisions 204710 via svnmerge from dvossel1-0/+59
https://origsvn.digium.com/svn/asterisk/trunk ................ r204710 | dvossel | 2009-07-02 11:03:44 -0500 (Thu, 02 Jul 2009) | 21 lines Merged revisions 204681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204681 | dvossel | 2009-07-02 10:05:57 -0500 (Thu, 02 Jul 2009) | 14 lines Improved mapping of extension states from combined device states. This fixes a few issues with incorrect extension states and adds a cli command, core show device2extenstate, to display all possible state mappings. (closes issue #15413) Reported by: legart Patches: exten_helper.diff uploaded by dvossel (license 671) Tested by: dvossel, legart, amilcar Review: https://reviewboard.asterisk.org/r/301/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@204754 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merged revisions 203699 via svnmerge from file1-0/+26
https://origsvn.digium.com/svn/asterisk/trunk ........ r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines Improve T.38 negotiation by exchanging session parameters between application and channel. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@203701 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-17Merged revisions 201262 via svnmerge from kpfleming1-5/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r201262 | kpfleming | 2009-06-17 07:04:17 -0500 (Wed, 17 Jun 2009) | 15 lines Merged revisions 201261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty. When the list to be appended is empty, and the list to be appended to is *not*, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@201263 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-16Merged revisions 201056,201090 via svnmerge from kpfleming3-14/+50
https://origsvn.digium.com/svn/asterisk/trunk ................ r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ ................ r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines Another minor fix to compiler attribute checking. Defaulting to 'static' for the function scope was bad... so remove it. ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@201093 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-16Merged revisions 200764 via svnmerge from kpfleming1-29/+17
https://origsvn.digium.com/svn/asterisk/trunk ........ r200764 | kpfleming | 2009-06-15 20:28:08 -0500 (Mon, 15 Jun 2009) | 11 lines Ensure that configure-script testing for compiler attributes actually works. The configure script tests for compiler attributes didn't actually enable enough warnings or provide a proper test harness to determine whether the compiler supports the attribute in question or not; this caused gcc 4.1 to report that it supports 'weakref', but it doesn't actually support it in the way that is needed for our optional API mechanism. The new configure script test will properly distinguish between full support and partial support for this attribute, among others. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@200767 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-10Merged revisions 199857 via svnmerge from seanbright1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines Merged revisions 199856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines __WORDSIZE is not available on all platforms, so use sizeof(void *) instead. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199858 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-08Merged revisions 199630 via svnmerge from seanbright1-4/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines Merged revisions 199626,199628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines Increase the size of our thread stack on 64 bit processors. We were setting the stack size for each thread to 240KB regardless of architecture, which meant that in some scenarios we actually had less available stack space on 64 bit processors (pointers use 8 bytes instead of 4). So now we calculate the stack size we reserve based on the platform's __WORDSIZE, which gives us: 32 bit -> 240KB 64 bit -> 496KB 128 bit -> 1008KB (that's right, we're ready for 128 bit processors) Patch typed by me but written by several members of #asterisk-dev, including Kevin, Tilghman, and Qwell. (closes issue #14932) Reported by: jpiszcz Patches: 06052009_issue14932.patch uploaded by seanbright (license 71) Tested by: seanbright ........ r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines Fix a typo in the stack size calculation just introduced. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199632 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-04Merged revisions 199051 via svnmerge from seanbright1-0/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@199052 f38db490-d61c-443f-a65b-d21fe96a405b
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