2006-12-23 Kevin P. Fleming * Asterisk 1.4.0 released. 2006-12-22 22:33 +0000 [r48870-48906] Jason Parker * Makefile, main/stdtime/localtime.c: Minor fixes for Solaris. * channels/chan_skinny.c: Fix for issue 7774 - patch by alamantia 2006-12-21 20:26 +0000 [r48783] Joshua Colp * /, redhat/asterisk.spec: Merged revisions 48782 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48782 | file | 2006-12-21 15:25:01 -0500 (Thu, 21 Dec 2006) | 2 lines Add new silence sound files to the spec for Redhat. (issue #8652 reported by alvaro_palma_aste) ........ 2006-12-20 02:56 +0000 [r48592-48637] Joshua Colp * apps/app_voicemail.c: vms doesn't exist on non-IMAP storage builds. * apps/app_voicemail.c: Pass 'vms' pointer to record_and_review so it is then passed to the IMAP store file function. (issue #8614 reported by punknow) * doc/snmp.txt: find is not the same as bind when it comes to documentation. (issue #8626 reported by johann8384) 2006-12-19 21:28 +0000 [r48586] Kevin P. Fleming * channels/Makefile: suppress compiler warnings in this module until it can be improved 2006-12-19 21:12 +0000 [r48585] Joshua Colp * apps/app_dial.c, /: Merged revisions 48584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48584 | file | 2006-12-19 16:10:26 -0500 (Tue, 19 Dec 2006) | 2 lines Free localuser structure when we fail to dial (issue #8612 reported by rizzo) ........ 2006-12-19 21:03 +0000 [r48583] Luigi Rizzo * apps/app_sms.c: fix a bogus datalen in the frames generated by app_sms (causing noisy output if you listen to the output!) This affects trunk as well, whereas 1.2 is ok. 2006-12-19 14:57 +0000 [r48577] Kevin P. Fleming * res/res_config_odbc.c, funcs/func_odbc.c: use the proper variable type for these unixODBC API calls, eliminating warnings on 64-bit platforms that use the 'new' 64-bit types for ODBC API calls 2006-12-19 03:46 +0000 [r48571] Joshua Colp * Makefile: Use env -i to start a fresh environment when going to build menuselect. This is more portable then using unset. (issue #8543 reported by jtodd) 2006-12-18 17:23 +0000 [r48566] Luigi Rizzo * include/asterisk/channel.h: unbreak the macro used for incrementing the frame counters. I don't know when the bug was introduced, but with the typical usage c->fin = FRAMECOUNT_INC(c->fin) the frame counters stay to 0. affects trunk as well (fix coming). 2006-12-18 17:15 +0000 [r48564] Joshua Colp * channels/chan_iax2.c: Put thread into proper list if we abort handling due to an error, and also hold the lock while putting it back into the proper idle list so we don't prematurely get a signal. (issue #8604 reported by arkadia) 2006-12-18 11:59 +0000 [r48513-48554] Kevin P. Fleming * codecs/lpc10/Makefile, main/Makefile, codecs/gsm/Makefile, utils/astman.c, utils/smsq.c, codecs/ilbc/Makefile, utils/ael_main.c: remove some now-unnecessary explicit includes of autoconfig.h clean up per-file dependencies during 'make clean' * build_tools/prep_tarball: need an additional argument here to make the downloads actually occur * configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: use m4 quoting for AC_MSG_NOTICE calls, to keep these calls from thinking they have multiple arguments * codecs/ilbc, formats, utils/Makefile, agi/Makefile, Makefile, funcs, build_tools/mkdep (removed), codecs/lpc10, main/db1-ast, main, codecs/gsm, pbx, res, channels, codecs, utils, agi, main/Makefile, apps, Makefile.moddir_rules, Makefile.rules, cdr: simplify dependency tracking system, using the compiler's built-in method for generating them, and only doing dependency tracking if developer mode is enabled via the configure script * Makefile, include/asterisk.h, main/stdtime/localtime.c: since we really, really have to have autoconfig.h included before all other headers (especially system headers), the Makefile will now force it to happen (this will fix build problems with files like ast_expr2f.c, where we can't control the inclusion order in the file itself) * funcs/func_curl.c: instead of initializing the curl library every time the CURL() function is invoked, do it only once per thread (this allows multiple calls to CURL() in the dialplan for a channel to run much more quickly, and also to re-use connections to the server) (thanks to JerJer for frequently complaining about this performance problem) 2006-12-15 19:55 +0000 [r48502-48506] Joshua Colp * main/rtp.c: Turn payload_lock into bridge_lock and make it encompass all RTP structure contents that may relate to bridge information, including who we are bridged to. * channels/chan_iax2.c: Hold call structure lock in places where a qualify or peer action can destroy it. * channels/chan_iax2.c: Lock network retransmission queue in all places that it is used. 2006-12-15 10:55 +0000 [r48481-48487] Olle Johansson * /, channels/chan_sip.c: Issue #8592 - treat 504 as 503 (imported from 1.2) * channels/chan_sip.c: Update to latest IANA spec 2006-12-15 06:28 +0000 [r48461-48478] Joshua Colp * channels/chan_iax2.c: Use a wakeup variable so that we don't wait on IO indefinitely if packets need to be retransmitted. * main/rtp.c, include/asterisk/rtp.h: Payload values on the RTP structure can change AFTER a bridge has started. This comes from the packet handling of the SIP response when indication that it was answered has been sent. Therefore we need to protect this data with a lock when we read/write. (issue #8232 reported by tgrman) * main/rtp.c: Remove direct RTCP bridging. I've come to the conclusion that we should handle this through the core and not just forward it on. Should solve a few bugs. 2006-12-12 Kevin P. Fleming * Asterisk 1.4.0-beta4 released. 2006-12-12 04:13 +0000 [r48401] Joshua Colp * apps/app_voicemail.c: Use S_OR in my previous app_voicemail. This is the way it should have been done. 2006-12-11 23:02 +0000 [r48396-48399] Matt O'Gorman * sounds/Makefile: new sounds package with 100% more silence * /, apps/app_externalivr.c: Merged revisions 48394 via svnmerge from https://svn.digium.com/svn/asterisk/branches/1.2 ........ r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006) | 4 lines app_externalivr needs a real silence file, and additional changes to add silence files into core instead of extra patch provided by bug 8177 with minor additions. ........ 2006-12-11 21:31 +0000 [r48391] Joshua Colp * apps/app_voicemail.c: Return non-existant callerid handling to that which it was before. In 1.4 and trunk callerid can be allocated but not have any contents so we have to use ast_strlen_zero before passing it to the relevant functions. (issue #8567 reported by pabelanger) 2006-12-11 05:37 +0000 [r48382] Tilghman Lesher * funcs/func_strings.c: STRFTIME() does not actually require an argument (issue 8540) 2006-12-11 05:36 +0000 [r48377-48381] Joshua Colp * main/rtp.c: Merge in my latest RTP changes. Break out RTP and RTCP callback functions so they no longer share a common one. * apps/app_meetme.c: Use the correct API call to say a device state changed. (Yes, I'm a nub.) * apps/app_meetme.c: Don't access the conference structure after it has been freed. 2006-12-11 00:47 +0000 [r48375] Tilghman Lesher * apps/app_nbscat.c, /, apps/app_festival.c, apps/app_mp3.c, res/res_agi.c, apps/app_zapras.c, apps/app_externalivr.c, apps/app_ices.c, res/res_musiconhold.c: Merged revisions 48374 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) | 5 lines When doing a fork() and exec(), two problems existed (Issue 8086): 1) Ignored signals stayed ignored after the exec(). 2) Signals could possibly fire between the fork() and exec(), causing Asterisk signal handlers within the child to execute, which caused nasty race conditions. ........ 2006-12-10 03:04 +0000 [r48372] Steve Murphy * channels/chan_zap.c, /: Merged revisions 48371 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48371 | murf | 2006-12-09 19:14:13 -0700 (Sat, 09 Dec 2006) | 1 line This version applies the patch suggested by stevens in bug 7836 (make inbound channel RINGING state consistent with other channels). ........ 2006-12-09 15:59 +0000 [r48362-48363] Russell Bryant * channels/chan_iax2.c: Use locking when accessing the registrations list. This list is not actually used very often, so the likelihood of there being a problem is pretty small, but still possible. For example, if the CLI command to list the registrations was called at the same time that a reload was occurring and the registrations list was getting destroyed and rebuilt, a crash could occur. In passing, go ahead and convert this list to use the linked list macros. * /: Blocked revisions 48361 via svnmerge ........ r48361 | russell | 2006-12-09 10:45:37 -0500 (Sat, 09 Dec 2006) | 6 lines Use locking when accessing the registrations list. This list is not actually used very often, so the likelihood of there being a problem is pretty small, but still possible. For example, if the CLI command to list the registrations was called at the same time that a reload was occurring and the registrations list was getting destroyed and rebuilt, a crash could occur. ........ 2006-12-07 18:17 +0000 [r48357] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 48356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48356 | russell | 2006-12-07 13:14:13 -0500 (Thu, 07 Dec 2006) | 3 lines Ensure that the file position is not incremented beyond the total number of files available for playback. (issue #8539, ulogic) ........ 2006-12-07 15:33 +0000 [r48349] Steve Murphy * main/manager.c, UPGRADE.txt, CHANGES: Here lies the fixes that killed bug 8423 -- OriginateSuccess and OriginateError incomplete channel name. May it rest in peace. 2006-12-06 16:25 +0000 [r48326] Olle Johansson * /, channels/chan_sip.c: Issue #8258 - fix handling of 487 being retransmitted to Asterisk 2006-12-06 16:15 +0000 [r48323] Russell Bryant * configs/iax.conf.sample, /: Merged revisions 48322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48322 | russell | 2006-12-06 11:05:54 -0500 (Wed, 06 Dec 2006) | 3 lines Fix the name of the rtignoreregexpire option in the sample configuration file. (issue #8526, arkadia) ........ 2006-12-06 12:27 +0000 [r48316-48317] Olle Johansson * /, channels/chan_sip.c: Don't send Contact on MESSAGE 2006-12-05 20:42 +0000 [r48279] Jason Parker * configure.ac: Fix curl version number testing to be much more friendly to non-bash shells. Issue 8508, patch by me. This *SHOULD* be POSIX compliant now.. 2006-12-05 17:29 +0000 [r48264-48270] Olle Johansson * channels/chan_sip.c: Merging the invitestate-1.4 branch after successful testing. Will check if I can solve this with less changes in 1.2. * configs/sip.conf.sample: Add missing s from another repository. (thanks jcmoore!) * configs/sip.conf.sample: Updating sip.conf.sample with information about T38 not working when chan_local or chan_agent is involved in the call. I don't know how big a fix that would be to solve, but this is the current state of affairs. (Chan_sip currently checks if the other side of the bridge has a SIP tech. We could/should implement another check, possibly for udptl_write or some flag in the ast_channel structure). 2006-12-05 01:41 +0000 [r48252-48254] Tilghman Lesher * apps/app_voicemail.c: Oops, forgot to release the odbc handle * apps/app_voicemail.c, /: Merged revisions 48251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) | 6 lines If the recording in the database is too large, it will fail to retrieve with an mmap error. Not too sure why this doesn't happen when we put it in the database, also, but since that doesn't seem to be broken, I'm not going to fix it (at least until someone reports it). Solution is to ask for the file in smaller chunks. (Bug 8385) ........ 2006-12-04 21:48 +0000 [r48237-48248] Jason Parker * apps/app_voicemail.c: Fix an issue which didn't allow unavail/greet/busy/etc messages from being saved into ODBC (and probably IMAP). * /: Blocked revisions 48246 via svnmerge ........ r48246 | qwell | 2006-12-04 15:20:34 -0600 (Mon, 04 Dec 2006) | 7 lines Revert change from 8016 - this breaks other stuff... Needs further review. Tip: When you've reported a bug about something and somebody has put up a patch for it.. It's not a good idea to open a completely new bug and say that something is broken because of the patch in the other bug - PLEASE mention something in the bug where the patch was actually created. ........ * /: Blocked revisions 48236 via svnmerge ........ r48236 | qwell | 2006-12-04 13:06:26 -0600 (Mon, 04 Dec 2006) | 4 lines Fix an issue where a message isn't saved correctly when using ODBC storage and reviewing a message. Issue 8016 - patch by sokhapkin. ........ 2006-12-04 18:16 +0000 [r48234] Joshua Colp * /: Blocked revisions 48233 via svnmerge ........ r48233 | file | 2006-12-04 13:14:46 -0500 (Mon, 04 Dec 2006) | 2 lines If the generic bridge tells us not to retry, and we have a frame to spit out then break the bridge. Props to markit in #asterisk-bugs for bringing this up. ........ 2006-12-04 17:54 +0000 [r48228-48230] Jason Parker * configs/voicemail.conf.sample: Add documentation to voicemail.conf.sample for ODBC storage. Issue 8499 - patch by blitzrage. * doc/snmp.txt: Attempt to document some of the dependencies that are needed for net-snmp Issue 8499 - initial patch by blitzrage. 2006-12-03 06:34 +0000 [r48223] Russell Bryant * sounds/Makefile: When "fetch" is in use, instead of "wget", --continue is not a valid option. (issue #8451) 2006-12-02 21:45 +0000 [r48199-48219] Olle Johansson * channels/chan_sip.c: - Removing one of two pieces of code to handle 481 response on INVITE - Move handling of REFER response to handle_response_refer() * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h, configs/sip.conf.sample: - Disable RTP hold timers while T.38 fax transmission happens - Encapsulate RTP timers in the rtp structure so we have one for video and one for audio The video one is not used in 1.4, really. Will be used for RTP keepalives when we can send something that video phones support in the RTP stream. I now this is a big architectual change at this stage for 1.4, but decided it was needed to avoid future bug reports. - Document the RTP NAT keepalive option in sip.conf.sample Issue 7679 in the bug tracker. Please test. 2006-12-02 03:50 +0000 [r48195] Russell Bryant * include/asterisk/utils.h: Backport the comment containing the warning regarding the limitations on the usage of this function. It is thread safe, but not technically reentrant. 2006-12-01 23:37 +0000 [r48193] Kevin P. Fleming * apps/app_dial.c, /: Merged revisions 48192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48192 | kpfleming | 2006-12-01 17:30:59 -0600 (Fri, 01 Dec 2006) | 2 lines if Dial() is going to send music-on-hold to the calling party, it has to send PROGRESS first to ensure that the reverse audio path has been setup first (BE-106) ........ 2006-12-01 23:16 +0000 [r48190] Russell Bryant * Makefile, configure, configure.ac, makeopts.in, sounds/Makefile: FreeBSD 6.1 does not include wget by default. However, it has fetch which will work just fine for our purposes of downloading the sounds packages. So, check for both wget and fetch and the configure script and use what was found to download them. If neither one was found, and sound packages are selected that must be downloaded, the install process will print out an informative error message indicating the situation. Also, fix a couple places where "make" was hard coded into some output messages by replacing them with the $(MAKE) variable. (issue #8451, initial patch by pabelanger, with additional modifications by me) 2006-12-01 20:25 +0000 [r48184-48186] Jason Parker * configs/extensions.conf.sample, /: Merged revisions 48183 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48183 | qwell | 2006-12-01 14:19:10 -0600 (Fri, 01 Dec 2006) | 2 lines Fix a small typo - issue 8848, reported by pabelanger ........ 2006-12-01 19:38 +0000 [r48179] Tilghman Lesher * main/cli.c: Double-unlock error (reported by blitzrage on IRC) 2006-12-01 17:41 +0000 [r48177] Olle Johansson * channels/chan_sip.c, configs/sip.conf.sample: - Backport of the "limitonpeers" patch from trunk, to fix a lot of issues with queues and SIP device states - Remove support for T.38 early media, since it's impossible. (Two patches in one - extra friday evening offer due to being off line from svn today... :-) 2006-11-30 21:18 +0000 [r48168] Joshua Colp * main/rtp.c, include/asterisk/rtp.h, channels/chan_gtalk.c: Do not do a partial bridge for Google Talk since we need to handle STUN. (issue #8448 reported by phsultan) 2006-11-30 20:51 +0000 [r48166] Olle Johansson * /, channels/chan_sip.c: Issue 8319 - change noncecount before using it. 2006-11-30 20:28 +0000 [r48143-48162] Joshua Colp * /: Blocked revisions 48161 via svnmerge ........ r48161 | file | 2006-11-30 15:27:29 -0500 (Thu, 30 Nov 2006) | 2 lines Don't write AST_FRAME_NULL or AST_FRAME_IAX frames out to the channel driver. (issue #8390 reported by hselasky) ........ * /, channels/chan_iax2.c: Merged revisions 48157 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48157 | file | 2006-11-30 15:06:43 -0500 (Thu, 30 Nov 2006) | 2 lines Only print out debug message if bridged channel is not NULL. (issue #8412 reported by jubilex) ........ * /, res/res_features.c: Merged revisions 48154 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48154 | file | 2006-11-30 14:04:11 -0500 (Thu, 30 Nov 2006) | 2 lines Do not listen for DTMF on the bridge that comes into existence when ParkedCall is executed. This means native bridging can now occur for this. (issue #8406 reported by kebl0155) ........ * main/cdr.c, /: Merged revisions 48151 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48151 | file | 2006-11-30 13:42:45 -0500 (Thu, 30 Nov 2006) | 2 lines Print certain CDR messages out at the NOTICE level versus WARNING since they can occur when used with the CDR applications and are perfectly fine. (issue #8367 reported by dartvader) ........ * /: Blocked revisions 48146 via svnmerge ........ r48146 | file | 2006-11-30 13:17:54 -0500 (Thu, 30 Nov 2006) | 2 lines Remember the pointer to the allocated block of memory so that we can free it and not cause a memory leak. (issue #8449 reported by arkadia) ........ * /, configs/sip.conf.sample: Merged revisions 48142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48142 | file | 2006-11-30 12:55:23 -0500 (Thu, 30 Nov 2006) | 2 lines Document 'port' for SIP peers, came up because of the current mailing list thread. (issue #8450 reported by blitzrage) ........ 2006-11-30 14:29 +0000 [r48129-48135] Olle Johansson * doc/manager.txt: Explain status reports and make codefreeze more happy :-) * /, channels/chan_sip.c: Clean up bad dialogs properly. Caused by GS 487 adapter without CSEQ on separate line in the REGISTER request. Imported from 1.2. 2006-11-29 21:05 +0000 [r48115] Joshua Colp * apps/app_voicemail.c: Use MAILTMPLEN instead of sizeof in mm_login. (issue #8420 reported by slimey) 2006-11-29 19:56 +0000 [r48113] Olle Johansson * configs/sip.conf.sample: Explain the use device status system implemented in SIP for subscriptions, queues and manager a bit better. Like in 1.2, you will get more detailed information if you set a call limit for a device. When the call limit is reached, the status system will report a device as busy. For queues, setting a call limit per SIP device is propably a requirement. In most cases, it will work much better if you only use type=peer and not type=friend. We might decide to backport the new setting from trunk to apply all call limits to the peer part of a friend only. 2006-11-29 16:50 +0000 [r48107] Joshua Colp * main/rtp.c, /: Merged revisions 48106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48106 | file | 2006-11-29 11:47:10 -0500 (Wed, 29 Nov 2006) | 2 lines If the frame was duplicated before writing out then we need to free it. (issue #8429 reported by edguy3) ........ 2006-11-29 08:03 +0000 [r48105] Olle Johansson * configs/sip.conf.sample: Clarify RTP timers. Sorry, grandma. 2006-11-29 04:26 +0000 [r48101] Joshua Colp * apps/app_voicemail.c: Don't crash if the mailstream was not created. 2006-11-28 18:26 +0000 [r48095] Jason Parker * Makefile: Export several more variables in top level Makefile. Inspired by issue 8438. 2006-11-28 16:57 +0000 [r48054-48088] Joshua Colp * channels/chan_phone.c, /: Merged revisions 48087 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48087 | file | 2006-11-28 11:56:01 -0500 (Tue, 28 Nov 2006) | 2 lines According to the research I have done we never needed to include compiler.h in the first place so let's not! (issue #8430 reported by edguy3) ........ * apps/app_voicemail.c, /: Merged revisions 48053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48053 | file | 2006-11-27 13:03:57 -0500 (Mon, 27 Nov 2006) | 2 lines Use the proper function to get the new message count instead of always using the filesystem. (issue #8421 reported by slimey) ........ 2006-11-27 17:20 +0000 [r48049] Tilghman Lesher * /, res/res_musiconhold.c: Merged revisions 48045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48045 | tilghman | 2006-11-27 11:15:54 -0600 (Mon, 27 Nov 2006) | 2 lines Random MOH wasn't really random (bug 8381) ........ 2006-11-27 17:17 +0000 [r48046] Russell Bryant * main/manager.c: Remove a couple of unused variables (issue #8380, casper) 2006-11-27 15:32 +0000 [r48038] Joshua Colp * pbx/pbx_spool.c, /: Merged revisions 48037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r48037 | file | 2006-11-27 10:30:37 -0500 (Mon, 27 Nov 2006) | 2 lines Do not reference the freed outgoing structure in the debug message. (issue #8425 reported by arkadia) ........ 2006-11-27 06:41 +0000 [r48031] Olle Johansson * channels/chan_sip.c: Change logging message 2006-11-26 00:26 +0000 [r48015-48017] Steve Murphy * funcs/func_cdr.c: might as well also document the raw values of the flag vars * /, funcs/func_cdr.c: A little bit of func_cdr documentation upgrade-- no bug# involved, although 8221 may have inspired it. 2006-11-25 09:28 +0000 [r48002] Olle Johansson * /, channels/chan_sip.c: Not having a HINT is not an ERROR. In 1.4 and future releases, you can disable subscription support totally or per peer in sip.conf with allowsubscribe = yes | no 2006-11-24 17:17 +0000 [r47992] Steve Murphy * main/translate.c: bug 8189 posted this fix for main/translate.c for PLC 2006-11-24 15:46 +0000 [r47989] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn_config.c, channels/chan_misdn.c, /: Merged revisions 47968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47968 | crichter | 2006-11-23 17:10:23 +0100 (Do, 23 Nov 2006) | 1 line fixed a litle bug regarding HOLD/RETRIEVE. beatufied some logs, changed some loglevels. changed the default value of block_on_alarm ........ 2006-11-23 11:01 +0000 [r47959] Olle Johansson * /, channels/chan_sip.c: Don't allocate unused variable. 2006-11-22 21:47 +0000 [r47944] Joshua Colp * main/rtp.c: Video will never reach Packet2Packet bridging and can do more harm then good. 2006-11-21 17:32 +0000 [r47897] Joshua Colp * main/rtp.c: If we have the non standard G726-32 setting turned on we want to return G726-32 to the SDP, not our AAL2 string. (issue #8330 reported by voipgate) 2006-11-21 15:20 +0000 [r47892] Olle Johansson * channels/chan_sip.c: Apparently Exosip sends a 101 after a 100 provisional response. Let's not treat that as early media. (discovered at the AVTF meeting in Paris). 2006-11-20 20:01 +0000 [r47863-47864] Tilghman Lesher * apps/app_voicemail.c: Oops, merge missed release of odbc object * apps/app_voicemail.c, /: Merged revisions 47862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47862 | tilghman | 2006-11-20 13:59:07 -0600 (Mon, 20 Nov 2006) | 2 lines Failing to trap -1 error from mmap causes segfault (Issue 8385) ........ 2006-11-20 19:51 +0000 [r47850-47860] Joshua Colp * main/frame.c, /: Merged revisions 47859 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47859 | file | 2006-11-20 14:50:21 -0500 (Mon, 20 Nov 2006) | 2 lines Don't forget to byte swap if we are exiting the smoother feed early. (issue #8287 reported by arturs) ........ * /: Blocked revisions 47855 via svnmerge ........ r47855 | file | 2006-11-20 11:16:22 -0500 (Mon, 20 Nov 2006) | 2 lines Free history items at the end of use of the temporary SIP pvt structure. (issue #8383 reported by benh) ........ * main/rtp.c: Only remove/destroy the RTCP I/O item if it exists. * .cleancount, apps/app_dial.c, apps/app_directed_pickup.c, include/asterisk/channel.h: Use a separate variable in the channel structure to store the context that the channel was dialed from. (issue #8382 reported by jiddings) 2006-11-20 11:45 +0000 [r47843-47845] Olle Johansson * configs/sip.conf.sample: Explain properly how videosupport works. Committ from Asterisk Video Task Force meeting in Paris! * /, channels/chan_sip.c: Make sure we destroy scheduled items and not use them ever again after destruction (rizzo) 2006-11-18 17:59 +0000 [r47823] Luigi Rizzo * channels/chan_sip.c: fix bug 7450 - Parsing fails if From header contains angle brackets (the bug was only in a corner case where the < was right after the opening quote, and the fix is trivial). 2006-11-16 23:19 +0000 [r47781-47782] Jason Parker * apps/app_db.c, apps/app_dial.c: Fix a couple of typos. Initially pointed out by mrobinson. * /: Blocked revisions 47780 via svnmerge ........ r47780 | qwell | 2006-11-16 17:16:35 -0600 (Thu, 16 Nov 2006) | 2 lines Fix a couple of typos in applications.. Initially spotted by mrobinson. ........ 2006-11-16 23:00 +0000 [r47777] Kevin P. Fleming * /, doc/billing.txt: update documentation regarding IAX2 transfers and CDRs Merged revisions 47776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47776 | kpfleming | 2006-11-16 16:57:31 -0600 (Thu, 16 Nov 2006) | 2 lines update clearly wrong documentation regarding cdr_custom ........ 2006-11-16 21:11 +0000 [r47762-47764] Joshua Colp * channels/chan_sip.c: Compare technology using the pointers instead of a straight comparison based on name. (issue #8228 reported by dean bath) * /: Blocked revisions 47761 via svnmerge ........ r47761 | file | 2006-11-16 15:29:28 -0500 (Thu, 16 Nov 2006) | 2 lines Look for the header file specifically in all cases, not just the existence of the directory. (issue #8358 reported by mrness) ........ 2006-11-16 20:09 +0000 [r47758] Kevin P. Fleming * configure, configure.ac: check for pre-1.4 versions of Zaptel and abort the configure script if found with an appropriate error message 2006-11-16 19:24 +0000 [r47755] Olle Johansson * channels/chan_sip.c, configs/sip.conf.sample: Make the HOLD notification optional, in order to avoid a lot of extra database lookups for all those realtime users out there. 2006-11-16 18:29 +0000 [r47748-47751] Joshua Colp * channels/chan_local.c, /: Merged revisions 47750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47750 | file | 2006-11-16 13:26:50 -0500 (Thu, 16 Nov 2006) | 2 lines Because of the way chan_local is written we should be extra careful and make sure our callback functions have a tech_pvt. (issue #8275 reported by mflorell) ........ * apps/app_meetme.c: Don't unreference the SLA object if there is no SLA object in the devicestate callback. (issue #8354 reported by loloski) 2006-11-16 16:51 +0000 [r47733-47744] Olle Johansson * /, channels/chan_sip.c: Don't fixup if there's nothing to fixup * UPGRADE.txt: Warn users about change in canreinvite * channels/chan_sip.c, configs/sip.conf.sample: - CANCEL is never authenticated (according to the RFC) - Update docs on canreinvite. "nonat" is the recommended setting for most users with phones behind a NAT. 2006-11-15 22:31 +0000 [r47712] Joshua Colp * channels/chan_local.c, /: Merged revisions 47711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47711 | file | 2006-11-15 17:29:30 -0500 (Wed, 15 Nov 2006) | 2 lines Make sure that the pvt structure exists before trying to do fixup on Local channels. (issue #7937 reported by mada123, fix by alamantia with mods by me) ........ 2006-11-15 21:56 +0000 [r47709] Tilghman Lesher * apps/app_voicemail.c: Fix ODBC_STORAGE for when context is NULL 2006-11-15 21:33 +0000 [r47707] Joshua Colp * main/channel.c: We need to ensure timelimit stuff is included as well so warnings get played. (issue #8050 reported by KNK) 2006-11-15 20:50 +0000 [r47701] Kevin P. Fleming * main/file.c: don't try to call fclose() if fopen() failed 2006-11-15 20:31 +0000 [r47698] Olle Johansson * channels/chan_sip.c: - Improve SIP history - Never send reply to ACK (again...) 2006-11-15 20:31 +0000 [r47684-47697] Kevin P. Fleming * apps/app_voicemail.c, /: Merged revisions 47677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006) | 4 lines ensure that message duration is included in email notifications for forwarded messages (BE-96, fix by me after corydon used his clue-bat on me) ensure that duration in the message metadata is updated if prepending is done during forwarding (related to BE-96) remove prototype for API call that does not exist ........ * main/config.c, /: Merged revisions 47686,47688-47689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47686 | kpfleming | 2006-11-15 13:42:05 -0600 (Wed, 15 Nov 2006) | 2 lines clear the category's variable tail pointer as well when variables are detached from it ........ r47688 | kpfleming | 2006-11-15 13:47:43 -0600 (Wed, 15 Nov 2006) | 2 lines when appending a list of variable to a category, ensure the tail pointer points to the last variable in the list ........ r47689 | kpfleming | 2006-11-15 13:58:46 -0600 (Wed, 15 Nov 2006) | 2 lines when re-writing the config file, don't repeat the path if it hasn't changed ........ * main/config.c, /: Merged revisions 47682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47682 | kpfleming | 2006-11-15 12:39:47 -0600 (Wed, 15 Nov 2006) | 2 lines ouch... don't use printf, use ast_log/ast_verbose ........ 2006-11-15 17:46 +0000 [r47672] Luigi Rizzo * main/cli.c: fix longest match search in find_cli. Trunk already fixed. 1.2 not affected (well, i have no idea, the code is totally different there). 2006-11-15 15:25 +0000 [r47649-47656] Olle Johansson * /, channels/chan_sip.c: Send error message when we can't allocate SIP dialog, possibly due to limitation of file descriptors. (imported from 1.2) 2006-11-15 04:45 +0000 [r47645] Joshua Colp * main/rtp.c: If NAT detection is turned on or already detected then say NAT is active when setting the remote RTP peer when doing early bridging. (issue #8365 reported by marcelbarbulescu) 2006-11-15 00:19 +0000 [r47641] Kevin P. Fleming * main/term.c: more formatting cleanup, and avoid running off the end of the string 2006-11-15 00:14 +0000 [r47639] Joshua Colp * main/rtp.c: Turn notice about unknown RTCP packet type into a debug message instead. 2006-11-15 00:05 +0000 [r47635] Kevin P. Fleming * channels/misdn/isdn_lib.c: silence compiler warning on 64-bit platforms (this variable is an 'int' anyway, comparing it to 'signed long' is not useful) 2006-11-14 22:17 +0000 [r47625-47632] Joshua Colp * apps/app_voicemail.c, /: Merged revisions 47631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47631 | file | 2006-11-14 17:15:10 -0500 (Tue, 14 Nov 2006) | 2 lines Update copyright information in the ADSI logo blob. ........ * channels/chan_sip.c: Only keep the video RTP structure around if 1. Video support is enabled and 2. A video codec is enabled on the dialog * funcs/func_uri.c: Small documentation clarification for URIENCODE. (issue #8294 reported by salaud) 2006-11-14 18:54 +0000 [r47621] Tilghman Lesher * apps/app_voicemail.c: Conversion of res_odbc API to include ast_ prefix did not completely transition app_voicemail when ODBC_STORAGE is used (reported on IRC by caio1982, not in bugtracker) 2006-11-14 16:45 +0000 [r47617] Joshua Colp * apps/app_amd.c: Use LOG_DEBUG to print out the indication that app_amd is using default settings instead of using LOG_NOTICE. This stops needless logging of this information under normal circumstances. (issue #8361 reported by Seb7) 2006-11-14 16:22 +0000 [r47597-47613] Olle Johansson * channels/chan_sip.c: Update documentation to fit the implementation... * /, channels/chan_sip.c: Issue #8272 - Don't destroy dialog in retransmission system if it's an OPTION packet from peerpoke 2006-11-13 21:28 +0000 [r47584] Joshua Colp * /, cdr/cdr_pgsql.c: Merged revisions 47583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47583 | file | 2006-11-13 16:26:36 -0500 (Mon, 13 Nov 2006) | 2 lines Initialize global pointers for connection and result to NULL. (issue #8356 reported by james) ........ 2006-11-13 20:20 +0000 [r47581] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 47580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47580 | tilghman | 2006-11-13 14:18:30 -0600 (Mon, 13 Nov 2006) | 2 lines Having more than 255 old messages caused corruption in the new/old count ........ 2006-11-13 19:15 +0000 [r47576] Steve Murphy * main/config.c: This solves bug 8342, whereby a crash occurs under certain circumstances while reading a config file with comments-- a call to CB_ADD shouldn't happen if withcomments is zero 2006-11-13 19:11 +0000 [r47573] Tilghman Lesher * main/cli.c, channels/chan_sip.c: Re-enable old deprecated commands 2006-11-13 19:10 +0000 [r47572] Olle Johansson * /, channels/chan_sip.c: - Don't reply to INVITE already replied to when we get BYE - Declare errmsg as int. Oops. 2006-11-13 18:18 +0000 [r47564] Steve Murphy * pbx/ael/ael-test/ref.ael-test3: Eager people beat me to fixing the messed if, but we all forgot to update the regressions. Until now. 2006-11-13 17:13 +0000 [r47553] Steve Murphy * pbx/pbx_ael.c: AEL need not complain about parkedcalls not being found... just confuses users 2006-11-13 17:08 +0000 [r47542-47551] Joshua Colp * /, apps/app_sms.c: Merged revisions 47549 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47549 | file | 2006-11-13 12:05:32 -0500 (Mon, 13 Nov 2006) | 2 lines When sending an SMS with a user data header properly set the UDH flag in the first byte. (issue #8347 reported by hoffmeis) ........ * main/cli.c: Free full command string upon unregistering of CLI command. Backported from revision 47536 from rizzo. 2006-11-13 16:00 +0000 [r47540] Olle Johansson * channels/chan_sip.c: Only produce error message about sip history once 2006-11-13 05:48 +0000 [r47527] Russell Bryant * configure, acinclude.m4: AC_PROG_SED is included in autoconf 2.60, but apparently it is not included in 2.59. So, to maintain compatability with 2.59 since it is a small change, copy this macro into acinclude.m4 and rename it to AST_PROG_SED. (issue #8345) 2006-11-13 05:46 +0000 [r47523-47526] Tilghman Lesher * res/res_odbc.c, /: Merged revisions 47525 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47525 | tilghman | 2006-11-12 23:45:11 -0600 (Sun, 12 Nov 2006) | 2 lines If the execute fails a second time, make sure that we don't pass back a stale handle ........ * channels/chan_zap.c, /: Merged revisions 47522 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47522 | tilghman | 2006-11-12 18:34:44 -0600 (Sun, 12 Nov 2006) | 2 lines Don't play dialtone if the seizing the channel fails (Bug 7754) ........ 2006-11-12 16:12 +0000 [r47507-47513] Olle Johansson * channels/chan_sip.c: Issue 8314 - Restore auto-framing (Thanks DEA!!!) * channels/chan_sip.c: Part of issue 8078 - parse even if udptl is UDPTL in sdp... * channels/chan_sip.c: - Don't destroy SIP dialog because of a failed T.38 re-invite. Wait for a bye. Final response to a re-invite does not mean that the session dies, only that the re-invite fails. - Keep RTP active during processing of T.38 re-invite. If the re-invite fails, RTP needs to remain as before the re-invite. Issue 8338 - darren1713. Please test. * channels/chan_sip.c: -Remove blocking of ptime: parsing in sdp -Add some comments to t.38 code 2006-11-12 06:23 +0000 [r47492-47497] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 47496 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47496 | russell | 2006-11-12 01:09:03 -0500 (Sun, 12 Nov 2006) | 4 lines Only do the check to determine whether the channel calling this function is an IAX2 channel when getting the IP address using the special argument, CURRENTCHANNEL. (issue #8341, jcovert) ........ * Makefile: Add the target "menuconfig" as an alias for the "menuselect" target. This is just a favor to users so that if you accidentally type "make menuconfig" instead of "make menuselect", it still works. (inspired by a comment on IRC from wangster calling me an "especially devious asterisk developer" for having it be menuselect instead of menuconfig. :) ) * main/term.c: Tweak the formatting of this new function to better conform to coding guidelines. 2006-11-11 02:04 +0000 [r47490] Matt O'Gorman * main/term.c, /, main/logger.c, include/asterisk/term.h: woohoo safe output! 2006-11-10 22:23 +0000 [r47480] Matt Frederickson * channels/chan_zap.c: Make sure we don't use 32 bits when we only need one bit. 2006-11-10 21:42 +0000 [r47463-47476] Olle Johansson * channels/chan_sip.c: ...and make sure that the dialog is destroyed, even if we don't get any answer on the bye... This is the channel that remains dead after the SIP transfer * channels/chan_sip.c: Add debug output while trying to trace bug in bug report * channels/chan_sip.c: Make sure we destroy dialog... * /, channels/chan_sip.c: Small cleanup of handle_request_invite() - imported from 1.2 with changes 2006-11-10 19:47 +0000 [r47462] Matt Frederickson * channels/chan_zap.c: Fix for #7321. Be able to explicitly hide callerid name for switches that bork on it. 2006-11-10 18:56 +0000 [r47454] Olle Johansson * /, channels/chan_sip.c: Issue 8010 - Fix support for multipart SDP (alphaque) 2006-11-10 17:13 +0000 [r47444] Luigi Rizzo * build_tools/prep_moduledeps: grep -m is not available on BSD, so use head -1 instead 2006-11-10 16:53 +0000 [r47437] Joshua Colp * apps/app_chanspy.c: Only split up extension and context if a value exists. (issue #8332 reported by loloski) 2006-11-10 16:51 +0000 [r47436] Tilghman Lesher * channels/chan_mgcp.c, main/cli.c, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_iax2.c: Discussion of these CLI changes resulted in more consistency (Bug 8236) 2006-11-10 16:36 +0000 [r47432-47433] Kevin P. Fleming * apps/app_queue.c: if adding a queue member is LOG_NOTICE, then removing them should be LOG_NOTICE, not LOG_DEBUG * apps/app_queue.c: reflect addition/removal of dynamic queue members in queue_log, so that people using dialplan replacement for AgentCallbackLogin can still track login/logout (issue #7736, reported/patched by whoiswes but this commit was written by me and covers all three paths for AQM/RQM) 2006-11-10 13:04 +0000 [r47414-47418] Olle Johansson * channels/chan_sip.c: Rip out half implementation of 491 response support, since it wasn't implemented properly and caused memory leaks in the case of us getting 491's, which Asterisk actually sends... Since it is a bit too complicated to fix this, I'll rip it out of 1.4 and put it on the to-do-list for future releases. Now, we handle this as congestion, which it really is. Issue #8331 * channels/chan_sip.c: Fix bit definition for SIP_PAG2_CALL_ONHOLD. Thanks fenlander! 2006-11-10 03:44 +0000 [r47398-47405] Joshua Colp * channels/chan_h323.c: Fix building of chan_h323 by completeing some structure definitions. (issue #8327 reported by Mithraen) * apps/app_voicemail.c: Do conversion in a more easier to read and working way for \r, \n, and \t. (issue #8324 reported by johnlange) 2006-11-09 21:26 +0000 [r47391] Russell Bryant * apps/app_voicemail.c, channels/chan_zap.c, build_tools/prep_moduledeps: Work around an issue that caused menuselect to display a bogus description for app_voicemail and chan_zap. These modules use some preprocessor directives to determine what it will report to Asterisk as its description. However, the way we extract this information from the source files for menuselect is not smart enough to figure this out. (issue #8326, #8328) 2006-11-09 16:53 +0000 [r47380] Joshua Colp * channels/chan_phone.c, /: Merged revisions 47379 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47379 | file | 2006-11-09 11:48:05 -0500 (Thu, 09 Nov 2006) | 2 lines Don't include compiler.h on kernels 2.6.18 and higher as, well, it's apparently going to be removed. This should make all you FC6 fans happy as your Asterisk will now build without any mods. ........ 2006-11-09 16:28 +0000 [r47352-47377] Russell Bryant * main/cli.c: fix tab completion for "core debug channel" and "core no debug channel" * main/cli.c: Fix "core show channel". Also, fix tab completion for both "core show channel" and "core show channels". * main/cli.c: Fix "core debug channel ". I guess someone needs to go through and audit every CLI command that changed number of arguments ... * main/asterisk.c: revert the previous change, which actually modified the deprecated command, "show profile". Now, actually apply the change to "core show profile". * main/asterisk.c: Fix argument parsing for the "core show profile" CLI command (fixed by rizzo in his branch, team/rizzo/astobj2) * main/cli.c: Fix another CLI command, "core show uptime" ... (issue #8323, reported by johnlange, fixed by myself) * main/asterisk.c: fix "core show version" to reflect the new number of arguments for this CLI command (issue #8316, kshumard) 2006-11-08 23:14 +0000 [r47344-47348] Steve Murphy * main/channel.c: This update fixes 7531 * channels/chan_skinny.c: Committed in behalf of 8190. 2006-11-08 21:46 +0000 [r47333-47338] Kevin P. Fleming * main/frame.c: the battle over CLI command formats has broken stuff... * channels/chan_sip.c: add simple fix for SDP to report proper sample rate for G.722 media sessions 2006-11-08 17:03 +0000 [r47323-47331] Russell Bryant * utils/streamplayer.c: I occasionally get email from users that are trying to figure out what this does, or due to some misunderstanding as to what it is supposed to do, can't get it to work. So, I have added some text here to hopefully explain what this application does and does not do. * channels/chan_gtalk.c: Make this module build again * configure, configure.ac, acinclude.m4: Copy the macros from libtool.m4 to our own acinclude.m4 such that libtool is no longer required to be installed to be able to generated the configure script. 2006-11-08 07:43 +0000 [r47309-47310] Olle Johansson * /, channels/chan_sip.c: Destroy dialog properly at unload (rizzo) 2006-11-07 23:46 +0000 [r47303] Steve Murphy * channels/chan_oss.c, main/channel.c, channels/chan_phone.c, channels/chan_misdn.c, channels/chan_skinny.c, channels/chan_features.c, channels/chan_h323.c, channels/chan_alsa.c, channels/chan_nbs.c, channels/chan_mgcp.c, include/asterisk/stringfields.h, apps/app_voicemail.c, main/pbx.c, channels/chan_vpb.cc, channels/chan_local.c, channels/chan_zap.c, channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, main/utils.c, include/asterisk/channel.h, channels/chan_gtalk.c, channels/chan_iax2.c: These mods are to solve the problem in bug 7506. It's a lot of rework to solve a fairly small problem... such is life. 2006-11-07 20:14 +0000 [r47284-47287] Joshua Colp * channels/chan_local.c: Make MOH work as it did before in chan_local, without this then it can go funky when transfers and MOH are involved. (issue #7671 reported by jmls) 2006-11-07 18:56 +0000 [r47279] Kevin P. Fleming * configs/musiconhold.conf.sample: clean up sample config, and make native file playback the more obvious default choice 2006-11-07 18:38 +0000 [r47275] Matt O'Gorman * apps/app_voicemail.c: large overhaul to voicemail imap support. Allows support for more imap servers, also a better implementation of several parts of the original work. patch provided by 8033 with major upgrades. 2006-11-07 17:30 +0000 [r47268] Olle Johansson * channels/chan_sip.c: Issue 8303 (lrizzo) - break instead of continue. 2006-11-07 13:13 +0000 [r47250] Olle Johansson * /, channels/chan_sip.c: Fixing the attack shield so it doesn't produce attacks... Issue 8265 - never reply to an ACK 2006-11-07 01:25 +0000 [r47239] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 47238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r47238 | russell | 2006-11-06 20:22:58 -0500 (Mon, 06 Nov 2006) | 5 lines If random order is enabled for files mode music on hold, set a random initial position, instead of always starting at the first file, and doing the random operation only when switching to the next file. (bug reported by John Lange on the asterisk-dev mailing list) ........ 2006-11-04 18:32 +0000 [r47199] Olle Johansson * channels/chan_sip.c: Issue #8284: Fixes to Invite/replaces and transfer from "john" Thank you! 2006-11-04 18:10 +0000 [r47192-47196] Russell Bryant * main/cli.c: Fix another bug in "core set debug" ... * main/asterisk.c, main/cli.c: Really fix the "core set debug" and "core set verbose" CLI commands. * main/cli.c: fix the "atleast" option to the "core set verbose" and "core set debug" CLI commands 2006-11-03 23:17 +0000 [r47176] Steve Murphy * channels/chan_sip.c: This fix introduced via bug 8233 2006-11-03 17:53 +0000 [r47107-47108] Luigi Rizzo * bootstrap.sh: align bootstrap.sh with the version in trunk (needs to be blocked as it is already in trunk) * configure.ac: add proper environment vars to detect modules on freebsd. (already applied to trunk so it needs to be blocked there) 2006-11-02 23:49 +0000 [r47051-47053] Tilghman Lesher * main/rtp.c, main/udptl.c, channels/chan_skinny.c, res/res_agi.c, channels/chan_h323.c, apps/app_queue.c, res/res_jabber.c: More changes making the CLI more consistent with "category verb arguments" (continuation of issue 8236) * main/config.c, main/cli.c, main/channel.c, main/manager.c, channels/chan_skinny.c, channels/chan_features.c, res/res_agi.c, main/http.c, main/file.c, main/logger.c, main/image.c, res/res_indications.c, main/asterisk.c, res/res_odbc.c, channels/chan_mgcp.c, apps/app_voicemail.c, main/pbx.c, channels/chan_local.c, main/frame.c, channels/chan_sip.c, res/res_features.c, channels/chan_agent.c, res/res_crypto.c, res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c: Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments" 2006-11-02 19:56 +0000 [r46992-47015] Olle Johansson * channels/chan_sip.c: Move check for codec translation to sip_call() instead of in add_sdp. No one bothers with the result of add_sdp anyway... Yet... * channels/chan_sip.c: Disable code for T38 over TCP and RTP since there's no trace of actual functionality for it :-) 2006-11-02 17:49 +0000 [r46965] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 46964 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46964 | russell | 2006-11-02 12:47:56 -0500 (Thu, 02 Nov 2006) | 3 lines ignore files in a music on hold directory that begin with '.' (issue #8249, cboie) ........ 2006-11-02 17:17 +0000 [r46963] Nadi Sarrar * channels/misdn/isdn_lib.c: find_free_chan_in_stack usage fix 2006-11-02 16:45 +0000 [r46937] Kevin P. Fleming * channels/chan_sip.c: don't send INVITE when we have determined that we can't offer any audio formats due to lack of transcoding support (or incorrect configuration) 2006-11-02 16:06 +0000 [r46930] Joshua Colp * /, channels/chan_sip.c: Merged revisions 46920 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46920 | file | 2006-11-02 11:02:27 -0500 (Thu, 02 Nov 2006) | 2 lines Repeat after me oej: I will at least make sure my code compiles before I commit it. ........ 2006-11-02 15:24 +0000 [r46901] Olle Johansson * /, channels/chan_sip.c: Dont overwrite pkt->flags (from 1.2) 2006-11-02 14:02 +0000 [r46845-46883] Russell Bryant * /, main/callerid.c: Add the missing call to free described in issue #8268. Also, add a bunch of missing calls to free in callerid_feed_jp(). * main/say.c: fix saying one hundred and two hundred in hebrew (issue #7810, eldadran) * Makefile, configure, codecs/gsm/Makefile, configure.ac, build_tools/strip_nonapi, makeopts.in: Fixes for cross-compilation on mips (issue #8058, ywalther, with some modifications) * aclocal.m4, build_tools/menuselect-deps.in, configure, build_tools/embed_modules.xml, configure.ac: Add a check in the configure script to determine whether ld is GNU ld or not. This is needed because module embedding only works for gnu ld. GNU ld is now listed as a dependency for all of the module embedding options in menuselect. (issue #8143) 2006-11-01 20:35 +0000 [r46822] Matt O'Gorman * channels/chan_gtalk.c: bind address support from bug 8164 2006-11-01 19:49 +0000 [r46802] Steve Murphy * res/res_config_odbc.c: a fix for bug 8251; the var_val needs to accept longer strings or mass confusion and a lot of lost time is the result 2006-11-01 18:39 +0000 [r46780] Joshua Colp * main/Makefile: Force poll() emulation for Darwin to always be on. It's too broken to consider being used. This resolves the console issue OSX users have been seeing. I would have liked to autoconf this but I haven't been able to come up with a test case that works. Que sera. 2006-11-01 18:26 +0000 [r46778] Russell Bryant * res/res_monitor.c, /: Merged revisions 46776 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46776 | russell | 2006-11-01 13:24:17 -0500 (Wed, 01 Nov 2006) | 9 lines soxmix and Asterisk expect different file extensions for certain formats. This was already handled for the wav49 format. However, it was not handled for ulaw and alaw. I fixed this in such a way that using the alternate extensions for ulaw and alaw will only happen if we know we're calling soxmix, and not a custom script defined using the MONITOR_EXEC variable. The wav49 processing was left alone so that external scripts will see no behavior change. (issue #7550, reported by mnicholson, proposed patch by junky, committed fix is a bit different) ........ 2006-11-01 18:21 +0000 [r46775] Joshua Colp * channels/chan_iax2.c: It's another round of chan_iax2 fixes! Should hopefully fix the deadlock issues people have been reporting. IAXtel now has qualify turned on for 800 peers and it is handling it fine. 2006-11-01 17:48 +0000 [r46760] Steve Murphy * main/config.c: Cleanups suggested by Russell. 2006-11-01 16:39 +0000 [r46744] Russell Bryant * channels/chan_zap.c: Prevent an infinite loop when config processing gets to a jitterbuffer option 2006-10-31 22:02 +0000 [r46716] Jason Parker * main/translate.c: Fix "core show translation" output. Issue #8243, patch by Damin. 2006-10-31 21:47 +0000 [r46711-46714] Kevin P. Fleming * include/asterisk/translate.h, main/translate.c: add an API so that translators can activate/deactivate themselves when needed * include/asterisk/translate.h, main/translate.c: revert changes that were the wrong way to address this... proper fix coming * main/translate.c: let's set the seen flag early enough to actually make a difference... * include/asterisk/translate.h, main/translate.c: don't re-do setup operations for translators that can dynamically register themselves 2006-10-31 15:49 +0000 [r46663] Tilghman Lesher * /: Blocked revisions 46662 via svnmerge ........ r46662 | tilghman | 2006-10-31 09:46:04 -0600 (Tue, 31 Oct 2006) | 3 lines Move thread-unsafe initializer to the module loading code; add the corresponding function to the module unload to fix a memory leak. ........ 2006-10-31 10:56 +0000 [r46583-46631] Olle Johansson * main/enum.c, funcs/func_enum.c, include/asterisk/enum.h: Issue #8089 - Fix the ENUM support (picking one record by number). Thanks otmar! * /, channels/chan_sip.c, configs/sip.conf.sample: Support ;rport when we're supposed to support ;rport. Issue #7473. * /, channels/chan_sip.c: If peer fails ACL check, fail peer at REGISTER * channels/chan_sip.c: Fix T38 too. Thanks, tgrman ! 2006-10-31 06:30 +0000 [r46554-46563] Russell Bryant * contrib/init.d/rc.redhat.asterisk: Start Asterisk later in the boot process to ensure it starts after stuff like MySQL (issue #8253, Alric) * /, main/utils.c: Merged revisions 46560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46560 | russell | 2006-10-31 01:18:36 -0500 (Tue, 31 Oct 2006) | 3 lines When handling the case where the hostname is just an IPV4 numeric address, be sure to set the address type. (issue #8247, alexr) ........ * /, res/res_agi.c: Merged revisions 46557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46557 | russell | 2006-10-31 01:13:09 -0500 (Tue, 31 Oct 2006) | 3 lines fix some copy/paste bugs in the checking of arguments for the "control stream file" AGI command (issue #8255, mnicholson) ........ * main/translate.c: Add a small tweak to the code that checks to see whether destination formats are translatable based on the source format. If we have already determined that there is no translation path in one direction, don't bother checking the other direction. 2006-10-30 22:19 +0000 [r46511-46526] Kevin P. Fleming * main/translate.c: when unregistering a translator, don't rebuild the translation matrix unless needed when filtering formats out of an offer, ensure we check for translation ability in both directions * include/asterisk/linkedlists.h: ensure that items removed from a list are always unlinked from the list (next pointer set to NULL) 2006-10-30 21:09 +0000 [r46474-46506] Joshua Colp * configure, configure.ac: Don't explicitly link in crypt as it is not used on some platforms. * channels/chan_iax2.c: We need to lock the pvt structure during retransmission as another worker thread may be doing something as well. 2006-10-30 16:27 +0000 [r46382-46433] Olle Johansson * main/asterisk.c, apps/app_voicemail.c, include/asterisk/file.h, include/asterisk/doxyref.h, channels/chan_sip.c, main/ast_expr2f.c, include/asterisk/module.h, formats/format_ogg_vorbis.c, main/app.c, include/asterisk/channel.h, include/asterisk/lock.h, include/asterisk/frame.h: Issue #8246 - Doxygen fixes from kshumard. An extra big thankyou is given to everyone that contributes to doxygen! THANK YOU! * main/rtp.c, /: Bind RTCP to the same IP as RTP * /, channels/chan_sip.c: Issue #7869 - Stop retransmission of 302 redirects (imported from 1.2) * /, channels/chan_sip.c: Issue #7608 - Notifications sent with wrong content-type (imported from 1.2, modified) * channels/chan_sip.c, CHANGES: Backport of patch for #7828 that was reported for trunk, but obviously exists in 1.4 too. * channels/chan_sip.c: Restoring the old logic, since working around it and fixing it seemed too complicated. - The SIP_OUTGOING flag indicates the direction of the last transaction in the dialog. - The initreq stores the last request in the dialog, the request that opened the latest transaction. Please now retry all the 1.4 bug reports with mixed to/from headers, tags etc in ACK, BYE, CANCEL. Thanks! * channels/chan_sip.c: Accepting a message twice may be misinterpreted... * channels/chan_sip.c: - 183 is not reliable message... - Error should not have SDP 2006-10-28 16:37 +0000 [r46377] Joshua Colp * utils/Makefile: Don't build muted on OpenBSD, it is not supported. 2006-10-27 19:03 +0000 [r46370] Russell Bryant * channels/chan_zap.c: move the copy of the default settings to the global settings back out of process_zap, so that they aren't overwritten when process_zap is called multiple times 2006-10-27 18:29 +0000 [r46367] Olle Johansson * contrib/asterisk-ng-doxygen: Put some doxygen pressure on Christian :-) 2006-10-27 17:39 +0000 [r46358-46363] Russell Bryant * main/asterisk.c, res/res_agi.c, apps/app_externalivr.c, res/res_musiconhold.c: We should always be using _exit() after a fork() or vfork() instead of exit(). This is because exit() does some extra cleanup which in some implementations of vfork(), for example, can actually modify the state of the parent process, causing very weird bugs or crashes. (issue #7971, Nick Gavrikov) * /: Blocked revisions 46361 via svnmerge ........ r46361 | russell | 2006-10-27 12:36:07 -0500 (Fri, 27 Oct 2006) | 5 lines We should always be using _exit() after a fork() or vfork() instead of exit(). This is because exit() does some extra cleanup which in some implementations of vfork(), for example, can actually modify the state of the parent process, causing very weird bugs or crashes. (issue #7971, Nick Gavrikov) ........ * channels/chan_zap.c: Instead of iterating all of the options once to look for jitterbuffer options, and then again for everything else, move the processing of jitterbuffer options into the main loop so that there are no erroneous messages about ignoring unknown options. (issue #8226) 2006-10-27 10:03 +0000 [r46351-46353] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: Merged revisions 46350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46350 | crichter | 2006-10-27 11:24:01 +0200 (Fr, 27 Okt 2006) | 1 line fixed a bug which caused chan_misdn to try to allocate 2 times the same channel on high load, which then caused instability of mISDN. removed a useless function from isdn_lib.c ........ * channels/misdn_config.c: fixed not compile issue, which was just introduced * channels/misdn_config.c, channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: Merged revisions 46176 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46176 | crichter | 2006-10-25 10:41:59 +0200 (Mi, 25 Okt 2006) | 1 line added nttimeout option to configure wether we disconnect calls on NT timeouts or not during an overlapdial session ........ 2006-10-26 17:57 +0000 [r46335-46340] Jason Parker * /, contrib/scripts/astgenkey.8: Merged revisions 46337 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r46337 | qwell | 2006-10-26 12:47:52 -0500 (Thu, 26 Oct 2006) | 2 lines oops - somebody forgot to change this - long ago, probably. ........ * CHANGES: grammar check 2006-10-26 16:38 +0000 [r46331] Olle Johansson * CHANGES: Corrections to changes (Multiparking is not included) 2006-10-26 16:31 +0000 [r46329] Russell Bryant * main/translate.c: - If the source has no audio or no video portion, do not call powerof() to get the format index. - Don't run through the audio and video loops if there is no audio or video portion of the source If 0 is passed to powerof, it will return -1. This value of -1 was then being used as an array index in these loops, which caused a crash on some systems. Other than this issue, this code works as we expected it to. If a format is not in the source, and we have to translation path to it, it is not offered in the list of acceptable destination formats. (fixes issue #8231) 2006-10-26 12:15 +0000 [r46317] Kevin P. Fleming * CHANGES: update to reflect G.722 addition 2006-10-26 04:18 +0000 [r46298] Russell Bryant * doc/backtrace.txt: update backtrace documentation to reflect changes in 1.4 (issue #8230, kshumard) 2006-10-26 01:37 +0000 [r46287] Mark Spencer * main/config.c, main/manager.c: Fix config comment code preservation code (thanks murf!) 2006-10-25 20:14 +0000 [r46276] Olle Johansson * channels/chan_sip.c: Old todo note - Don't add Contact header on BYE and Cancel 2006-10-25 19:24 +0000 [r46253-46255] Russell Bryant * configure.ac: fix error output when checking for openh323 to refer to openh323 instead of pwlib (issue #8222, misaksen) 2006-10-25 19:16 +0000 [r46252] Olle Johansson * channels/chan_sip.c: Somewhat ugly code to try to fix issue #7608. Since the problem was not very well defined, the fix is a bit fuzzy too... Thanks to Luigi for accidentally spotting the possible problem! 2006-10-25 19:08 +0000 [r46249] Russell Bryant * apps/app_queue.c: update warning message to include "agi" option (issue #8225, jmls) 2006-10-25 18:13 +0000 [r46237-46248] Kevin P. Fleming * sounds/Makefile: use 1.4.3 extra sounds with corrected silence files * sounds/sounds.xml, sounds/Makefile: add support for prebuilt G.722 prompts and music on hold files 2006-10-25 15:56 +0000 [r46214-46216] Olle Johansson * channels/chan_sip.c: show settings doesn't produce a list of similar objects, it should stay a "show" 2006-10-25 14:32 +0000 [r46200] Kevin P. Fleming * main/cli.c, main/cdr.c, channels/chan_phone.c, pbx/pbx_spool.c, channels/chan_features.c, pbx/pbx_ael.c, channels/chan_h323.c, pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_sms.c, main/image.c, channels/chan_nbs.c, apps/app_rpt.c, main/db.c, cdr/cdr_custom.c, channels/chan_mgcp.c, apps/app_parkandannounce.c, apps/app_voicemail.c, channels/chan_sip.c, apps/app_softhangup.c, apps/app_record.c, res/res_adsi.c, main/utils.c, apps/app_ices.c, pbx/dundi-parser.c, channels/chan_iax2.c, apps/app_queue.c, apps/app_getcpeid.c: apparently developers are still not aware that they should be use ast_copy_string instead of strncpy... fix up many more users, and fix some bugs in the process 2006-10-25 04:58 +0000 [r46165] Tilghman Lesher * main/pbx.c: WaitExten truncates decimals of times to wait, instead of accepting them (Bug 8208) 2006-10-25 00:26 +0000 [r46152-46154] Kevin P. Fleming * main/rtp.c, main/frame.c, main/translate.c, formats/format_pcm.c, channels/chan_h323.c, channels/chan_iax2.c, include/asterisk/frame.h: add passthrough and file format support for G.722 16KHz audio (issue #5084, original patch by andrew, updated by mithraen) * channels/chan_sip.c, main/translate.c: code zone experiment: don't offer formats in the outbound INVITE that aren't either passthrough or translatable * main/translate.c: if multiple translators are registered for the same source/dest combination, ensure that the lowest-cost one is always inserted earlier in the list 2006-10-24 20:30 +0000 [r46142] Mark Spencer * res/res_agi.c: Fix FastAGI when there is no pid (bug #7628, #8147) 2006-10-24 19:29 +0000 [r46130] Joshua Colp * channels/chan_iax2.c: We need to initialize our scheduler pthread condition... yes. 2006-10-24 08:34 +0000 [r46114-46117] Luigi Rizzo * main/http.c: merge 45152 don't leak descriptors in http.c * channels/chan_sip.c: merge 45966 refer_to_domain potentially containing options * channels/chan_sip.c: merge 46026 improper checks on get_header() return values * channels/chan_sip.c: merge 46045 prevent NULL args to ast_strdupa() in chan_sip.c 2006-10-24 05:23 +0000 [r46093] Russell Bryant * Makefile: Restore the ability to remove the firmware directory without causing the installation to fail (issue #8111) 2006-10-24 03:53 +0000 [r46080-46083] Kevin P. Fleming * main/translate.c: ensure that the translation matrix is properly lock-protected every place it is used * include/asterisk/translate.h, main/translate.c: add an API call to allow channel drivers to determine which media formats are compatible (passthrough or transcode) with the format an existing channel is already using * doc/imapstorage.txt: simplify and correct voicemail IMAP storage build instructions 2006-10-24 03:01 +0000 [r46078] Tilghman Lesher * main/channel.c: Pass through a frame if we don't know what it is, rather than trying to pass a NULL, which will segfault a channel driver (Bug 8149) 2006-10-24 01:27 +0000 [r45999-46067] Russell Bryant * utils/muted.c, utils/ael_main.c: In muted.c, check the return value of strdup. In ael_main.c, check the return value of calloc. (issue #8157) In passing fix a few minor bugs in ael_main.c. The last argument to strncpy() was a hard-coded 100, where it should have been 99. I changed this to use sizeof() - 1. * apps/app_meetme.c: Fix the descriptions of some of the MeetMeAdmin options (issue #8098, mflorell) * res/res_jabber.c: don't crash when an incoming message has no "from" (issue #8205, jmls) 2006-10-23 00:27 +0000 [r45928] Joshua Colp * /, cdr/cdr_odbc.c: Merged revisions 45927 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45927 | file | 2006-10-22 20:25:28 -0400 (Sun, 22 Oct 2006) | 2 lines Don't leak memory mmmk? ........ 2006-10-22 21:44 +0000 [r45916] Christian Richter * channels/chan_misdn.c, /: Merged revisions 45808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45808 | crichter | 2006-10-21 14:35:13 +0200 (Sat, 21 Oct 2006) | 1 line fixed issue, that if chan_misdn is loaded and couldn't be initialized it would cause a segfault after 'reload'. Reported by Drew/Matt thx. ........ 2006-10-21 18:49 +0000 [r45818] Russell Bryant * res/res_monitor.c: Add a couple missing unregistrations of manager actions and remove duplicate unregistrations of applications. (issue #8194, jmls) 2006-10-21 18:48 +0000 [r45775-45817] Joshua Colp * main/loader.c: Don't use promotion on Darwin because it doesn't seem to work quite right in all cases, this should solve the unresolved symbol issue people have been seeing. * Makefile: Pass DESTDIR and ASTSBINDIR so that the utilities get installed in the proper location (reported on asterisk-dev mailing list) 2006-10-20 07:44 +0000 [r45741] Olle Johansson * channels/chan_sip.c: Let's understand SIP: - REFER can create dialog, Asterisk does not support it yet - NOTIFY can create dialog in Asterisk's implementation (voicemail) even though we don't support the server side of it. In this case, the standard is a side issue ;-) - Added extened functionality for unsupported methods (PING, PUBLISH) so we don't create PVT's for those either. Russellb needs to judge what to do with this in 1.2, but I think the current implementation n 1.2 is a bug since we're sending bad replies to NOTIFY and REFER outside of dialogs 2006-10-19 17:24 +0000 [r45678-45694] Joshua Colp * res/res_jabber.c: Let's remember to unregister JabberStatus too (issue #8184 reported by jmls) * /, apps/app_externalivr.c: Merged revisions 45691 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r45691 | file | 2006-10-19 13:16:37 -0400 (Thu, 19 Oct 2006) | 2 lines Respect language selection when seeing if the file exists (issue #8178 reported by mnicholson) ........ * channels/chan_sip.c: If the jitterbuffer is forced on then we can't partially bridge (reported by wangster on #asterisk-dev) 2006-10-19 00:59 +0000 [r45622] Russell Bryant * channels/chan_sip.c: Don't leak the actual thread-specific sip_pvt struct 2006-10-18 23:49 +0000 [r45621] Kevin P. Fleming * channels/chan_sip.c: don't leak memory when a chan_sip thread is destroyed that has a thread-local temp_pvt allocated 2006-10-18 21:03 +0000 [r45595] Joshua Colp * main/asterisk.c: Don't modify things if we are using vfork as this is very bad and may cause unexpected behavior (issue #7970 reported by Nick Gavrikov) 2006-10-18 11:54 +0000 [r45517] Olle Johansson * channels/chan_sip.c: remove duplicate declarations 2006-10-18 04:09 +0000 [r45464] Luigi Rizzo * main/http.c: merge from trunk: move ast_variables_destroy() to a better place in handle_uri() to avoid leaking memory on non existing files. 2006-10-18 03:02 +0000 [r45452] Joshua Colp * main/rtp.c: Don't segfault if you're using a channel driver that doesn't turn RTCP on 2006-10-18 02:41 +0000 [r45439-45441] Russell Bryant * main/channel.c: Don't attempt to access private data members of the pthread_mutex_t object, because this does not work on all linux systems. Instead, just access the reentrancy field in the ast_mutex_info struct when DEBUG_THREADS is enabled. If DEBUG_CHANNEL_LOCKS is enabled, the developer probably has DEBUG_THREADS on as well. (issue #8139, me) * configs/sip_notify.conf.sample: update entry to reboot a snom phone (issue #7850, pnlarsson) 2006-10-17 Kevin P. Fleming * Asterisk 1.4.0-beta3 released. 2006-10-17 22:31 +0000 [r45408-45410] Kevin P. Fleming * include/asterisk/stringfields.h, main/ast_expr2.c, main/channel.c, channels/chan_sip.c, channels/chan_iax2.c: optimize the 'quick response' code a bit more... no more malloc() or memset() for each response expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed 2006-10-17 20:38 +0000 [r45378-45381] Joshua Colp * channels/chan_sip.c: Don't create a "real" pvt structure for requests that shouldn't be able to create one. Instead use a temporary pvt and fill it with enough information so we can send a reply. 2006-10-17 17:39 +0000 [r45329] Olle Johansson * configs/sip.conf.sample: Adding information about Marks direct-RTP hack to the docs... 2006-10-17 17:22 +0000 [r45327] Kevin P. Fleming * LICENSE: provide licensing language for IAXy firmware file 2006-10-16 20:06 +0000 [r45246-45280] Joshua Colp * apps/app_dial.c, apps/app_directed_pickup.c: Backport of new directed pickup (BE-85). 2006-10-16 13:59 +0000 [r45196-45213] Olle Johansson * CREDITS: Adding Inotel to credits for SIP transfers. Thanks for your support! * channels/chan_sip.c: Don't destroy dialog for unexpected REFER response... 2006-10-14 04:38 +0000 [r45143] Steve Murphy * funcs/func_rand.c: update the doc string for both AEL and extensions.conf users. 2006-10-13 23:02 +0000 [r45125] Kevin P. Fleming * main/acl.c don't drop the entire permit/deny list when an attempt is made to add an invalid entry (BE-92) 2006-10-13 21:06 +0000 [r45104-45106] Joshua Colp * res/res_speech.c: Clear the quiet flag too since we are restarting a recognition again (reported on -dev by Stephan Edelman) * res/res_speech.c: Check return value from engine in case of failure (ie: out of licenses) (reported on -dev mailing list) 2006-10-13 20:52 +0000 [r45103] Steve Murphy * pbx/ael/ael-test/ref.ael-vtest17 (added), pbx/ael/ael-test/ael-vtest17/extensions.ael (added), pbx/ael/ael-test/ael-vtest17 (added), pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c: Bug 8128 fixed in this release via these changes 2006-10-13 19:19 +0000 [r45088] Christian Richter * channels/chan_misdn.c: avoiding warning, fixing potential bug 2006-10-13 18:42 +0000 [r45051-45079] Joshua Colp * codecs/lpc10/placev.c, codecs/lpc10/irc2pc.c, codecs/lpc10/decode.c, codecs/lpc10/dcbias.c, codecs/lpc10/pitsyn.c, codecs/lpc10/voicin.c, codecs/lpc10/difmag.c, codecs/lpc10/hp100.c, codecs/lpc10/synths.c, codecs/lpc10/preemp.c, codecs/lpc10/rcchk.c, codecs/lpc10/lpfilt.c, codecs/lpc10/mload.c, codecs/lpc10/lpcenc.c, codecs/lpc10/vparms.c, codecs/lpc10/dyptrk.c, codecs/lpc10/lpcini.c, codecs/lpc10/random.c, codecs/lpc10/ham84.c, codecs/lpc10/chanwr.c, codecs/lpc10/placea.c, codecs/lpc10/tbdm.c, codecs/lpc10/analys.c, codecs/lpc10/onset.c, codecs/lpc10/energy.c, codecs/lpc10/deemp.c, codecs/lpc10/lpcdec.c, codecs/lpc10/ivfilt.c, codecs/lpc10/median.c, codecs/lpc10/encode.c, codecs/lpc10/bsynz.c, codecs/lpc10/prepro.c, codecs/lpc10/invert.c: And file said... let the compiler warnings STOP! * apps/app_chanspy.c: Turn on volume adjustment if it needs to be on (issue #8136 reported by mnicholson) * apps/app_playback.c: Move say.conf existence check to do_say function since it is called from multiple places (issue #8144 reported by kshumard) 2006-10-13 16:19 +0000 [r45049] Kevin P. Fleming * channels/chan_iax2.c: when sending a call to a peer, use the proper socket if we have multiple bindings (reported on asterisk-dev) 2006-10-13 16:01 +0000 [r45031-45040] Joshua Colp * channels/chan_sip.c: Complete merging in RPID screen changes (issue #8101 reported by hristo, patch by oej in revision 44757) * main/dnsmgr.c: Pass the right value to usleep for sleeping, and always add the background refresh item back into the scheduler if enabled since it is deleted during reload. (issue #8142 reported by p_lindheimer) 2006-10-13 15:41 +0000 [r45027] Kevin P. Fleming * configure, include/asterisk/autoconfig.h.in, configure.ac, main/utils.c: use a configure script test for PMTU discovery control instead of just assuming it's available on Linux 2006-10-13 14:45 +0000 [r44994-45026] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c: fixed some echocandisable issues when bridged. this caused a kernel panic sometimes.. also some minor formatting fixes * channels/misdn/isdn_msg_parser.c: fixed issue that the hangupcause got a wrong isdn cause at RELEASE_COMPLETE 2006-10-12 22:07 +0000 [r44992] Luigi Rizzo * channels/chan_sip.c: merge formatting and minor code simplifications from trunk 2006-10-12 20:34 +0000 [r44982] Matt O'Gorman * channels/chan_gtalk.c: fix for bug 7764. 2006-10-12 19:14 +0000 [r44956-44971] Kevin P. Fleming * channels/chan_sip.c: we can only send one 'a=ptime' attribute per media session, not one for each format * main/netsock.c, include/asterisk/utils.h, channels/chan_sip.c, main/utils.c: ensure that IAX2 and SIP sockets allow UDP fragmentation when running on Linux (thanks to Brian Candler on the asterisk-dev list for the tip) 2006-10-12 16:56 +0000 [r44945] Russell Bryant * main/manager.c: fix a silly typo in a comment that I saw while reading the commit list 2006-10-12 16:08 +0000 [r44942] Joshua Colp * Makefile: Pass off AUDIO_LIBS so muted can link on OSX (issue #8135 reported by ssokol) 2006-10-12 12:55 +0000 [r44921] Nadi Sarrar * main/manager.c: append_event must be called while holding the session lock 2006-10-12 10:24 +0000 [r44911] Russell Bryant * res/res_jabber.c: change some debug output to use LOG_DEBUG instead of verbose output 2006-10-11 16:57 +0000 [r44888] Jason Parker * main/db1-ast/Makefile: These are already set by the parent Makefile.. There is no need to have this here (it doesn't actually work anyways). 2006-10-11 09:18 +0000 [r44854] Christian Richter * channels/misdn/isdn_lib.c: removed warning because of missing prototype declaration 2006-10-10 19:23 +0000 [r44830] Olle Johansson * channels/chan_sip.c: Do not set default/global values in the variable declaration, set it in reload_config() 2006-10-10 17:21 +0000 [r44819] Joshua Colp * channels/chan_sip.c: Move some stuff around so that a NOTIFY dialog won't hang around until the end of the world under certain circumstances 2006-10-10 16:44 +0000 [r44809] Paul Cadach * main/channel.c, funcs/func_channel.c, include/asterisk/channel.h: CHANNEL() function sometime mix parameter and value 2006-10-10 16:42 +0000 [r44808] Tilghman Lesher * funcs/func_logic.c: Lost of a bit of logic when this was simplified between 1.2 and 1.4 (Bug 8117) 2006-10-10 16:30 +0000 [r44806] Joshua Colp * channels/chan_sip.c: Bail out if we have no refer structure and we get a refer response 2006-10-10 16:21 +0000 [r44805] Luigi Rizzo * channels/chan_sip.c: more merge from trunk (comments and change a static function name) 2006-10-10 15:23 +0000 [r44788] Joshua Colp * channels/chan_sip.c: Only set DTMF information if an RTP structure exists 2006-10-10 13:50 +0000 [r44786] Christian Richter * channels/misdn/isdn_lib.c, channels/chan_misdn.c: (re)added support of dynamically enabling hdlc on bchannels 2006-10-10 08:25 +0000 [r44776-44777] Luigi Rizzo * channels/chan_sip.c: whitespace changes related to previous commit * channels/chan_sip.c: merge a few code simplifications that have gone into trunk during last week, to reduce differences between the two branches and make porting fixes easier. 2006-10-09 16:12 +0000 [r44764] Jason Parker * channels/chan_skinny.c: Fix a problem where phones that go "missing" never got unregistered. Issue #8067, reported by pj, patch by Anthony LaMantia (with minor whitespace modifications) 2006-10-09 15:46 +0000 [r44759-44760] Joshua Colp * channels/chan_iax2.c: iaxs[callno] may go away if we try to avoid the deadlock * channels/chan_iax2.c: Properly avoid a collision with iax2_hangup (issue #8115 reported by vazir) 2006-10-08 14:14 +0000 [r44746] Luigi Rizzo * channels/chan_sip.c: do not dereference p if we know it is NULL 2006-10-07 14:39 +0000 [r44684] Paul Cadach * channels/h323/ast_h323.cxx, channels/chan_h323.c, channels/h323/ast_h323.h, channels/h323/chan_h323.h: Propagate caller's transfer capability too 2006-10-07 11:37 +0000 [r44650-44665] Luigi Rizzo * channels/chan_sip.c: put common code in a function to avoid repetitions. * channels/chan_sip.c: remove hardwired usage of 5060, use DEFAULT_SIP_PORT instead * channels/chan_sip.c: option_debug checking before printing to debug channel. * channels/chan_sip.c: backport simplifications on sip_register, usage of ast_set2_flag(), and fixes to the handling of failed module loading. * channels/chan_sip.c: improve and document function get_in_brackets(), introducing a helper function find_closing_quote() of more general use. 2006-10-06 21:28 +0000 [r44629-44631] Kevin P. Fleming * include/asterisk/linkedlists.h: ensure that mutex locks inside list heads are initialized properly on platforms that require constructor initialization (issue #8029, patch from timrobbins) * CHANGES: remove Jingle as per mog 2006-10-06 21:08 +0000 [r44628] Joshua Colp * main/rtp.c: Remove the seqno check for RFC2833, the handler is smart enough to not need it. 2006-10-06 21:07 +0000 [r44627] Kevin P. Fleming * CHANGES: various cleanups 2006-10-06 18:46 +0000 [r44581-44605] Joshua Colp * main/rtp.c: When the sequence number rolls over then reset the recorded sequence number for DTMF (issue #8106 reported by bungalow) * main/file.c: Even more frames to treat as though the remote side disappeared (issue #8097 reported by eldadran) 2006-10-06 15:59 +0000 [r44567] Luigi Rizzo * main/manager.c, main/http.c: make sure sockets are blocking when they should be blocking. 2006-10-06 12:53 +0000 [r44559-44563] Christian Richter * channels/chan_misdn.c: fixed segfault which happens during hold/transfer action * channels/chan_misdn.c: if INFORMATION Message come with keypad instead of called party number, we just use the keypad as called party number. * channels/misdn/isdn_lib.c, channels/misdn_config.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: added the option 'reject_cause' to make it possible to set the RELEASE_COMPLETE - cause on the 3. incoming PMP channel, which is automatically rejected because chan_misdn does not support that kind of callwaiting. Therefore chan_misdn supports now 3 incoming channels on a PMP BRI Port. misdn_lib_get_free_bc now gets the info if the requested channel is incoming or outgoing to make the 3. channel possible * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c: fixed the hold/retrieve/transfer issues, removed a useless bc field, added setting of frame.delivery fields, some minor code cleanups 2006-10-05 19:57 +0000 [r44502] Joshua Colp * main/file.c: Treat busy control frames as hangup in the file streaming core (issue #8097 reported by eldadran) 2006-10-05 18:21 +0000 [r44488] Steve Murphy * pbx/pbx_ael.c: This mod fixes a problem pointed out by dgarstang. Many thanks to Doug! 2006-10-05 18:01 +0000 [r44486] Joshua Colp * channels/chan_sip.c: One more T.38 fix! Don't leave a reinvite hanging by a thread if the other side is already setup with T.38 2006-10-05 16:10 +0000 [r44476] Kevin P. Fleming * main/app.c: don't segfault when an argument without a close parenthesis is found stop parsing as soon as that situation occurs 2006-10-05 15:22 +0000 [r44465-44466] Steve Murphy * CHANGES: I put the accumulated changes from the commit logs and inspection, into CHANGES. Hope everyone approves! * configs/muted.conf.sample, utils/muted.c: Hang on a minute, the install process sticks muted.conf in /etc/asterisk, so that's where muted should look for it, right? 2006-10-05 02:40 +0000 [r44450] Joshua Colp * channels/chan_sip.c: Don't totally bail out if T.38 was negotiated 2006-10-05 01:42 +0000 [r44433-44436] Kevin P. Fleming * channels/chan_sip.c: fix Polycom presence notification again 2006-10-04 22:52 +0000 [r44407-44409] Luigi Rizzo * utils/Makefile: as far as i can tell astman only uses newt... * Makefile: put linker flags in ASTLDFLAGS where they belong 2006-10-04 21:17 +0000 [r44390-44393] Kevin P. Fleming * channels/chan_sip.c: remove workaround for old Polycom firmware SUBSCRIBE requests add workaround for new Polycom firmware SUBSCRIBE requests (bug is known to exist in 2.0.1 firmware) * include/asterisk.h, main/utils.c: make LOW_MEMORY builds actually work 2006-10-04 19:57 +0000 [r44380] Steve Murphy * pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael.tab.c, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, pbx/pbx_ael.c, pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ael-test16/extensions.ael (added), pbx/ael/ael-test/ael-test16 (added), pbx/ael/ael.y, pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test14, pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, pbx/ael/ael-test/ref.ael-test16 (added): These changes fix the problems reported in bug 8090 2006-10-04 19:47 +0000 [r44378] Kevin P. Fleming * channels/chan_oss.c, main/cdr.c, channels/chan_phone.c, main/manager.c, pbx/pbx_spool.c, res/res_smdi.c, channels/chan_skinny.c, channels/chan_h323.c, main/http.c, channels/chan_alsa.c, pbx/pbx_dundi.c, apps/app_mixmonitor.c, main/asterisk.c, channels/chan_mgcp.c, main/autoservice.c, include/asterisk/utils.h, main/dnsmgr.c, channels/chan_zap.c, channels/chan_sip.c, apps/app_meetme.c, res/res_snmp.c, main/devicestate.c, main/utils.c, res/res_musiconhold.c, channels/chan_iax2.c, apps/app_queue.c, res/res_jabber.c: update thread creation code a bit reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined 2006-10-04 17:04 +0000 [r44337-44365] Steve Murphy * configs/muted.conf.sample: I've been meaning to add some explanation about muted... here it is * configs/manager.conf.sample: CLI reverbification update to this config file * apps/app_macro.c: In response to bug 7776, a Warning has been added to the doc string for Macro(). 2006-10-04 00:25 +0000 [r44322] Kevin P. Fleming * main/asterisk.c, main/loader.c, main/term.c, Makefile, include/asterisk.h: ensure that local include files are always used avoid a duplicate function name (term_init()) 2006-10-03 22:35 +0000 [r44312] Matt O'Gorman * channels/chan_gtalk.c, res/res_jabber.c: fix issue with dialing client without resource. 2006-10-03 20:18 +0000 [r44298] Kevin P. Fleming * apps/app_queue.c: fix a logic error in my previous fix to the queue reload code 2006-10-03 18:42 +0000 [r44286] Paul Cadach * channels/h323/ast_h323.cxx: Change default presentation indicator to "user provided not screened" if octet 3a missed in CallingPartyNumber IE 2006-10-03 18:35 +0000 [r44284] Joshua Colp * channels/chan_sip.c: Use VideoSupport instead so it is considered a valid XML attribute name. (issue #8075 reported by renemendoza) 2006-10-03 18:30 +0000 [r44283] Paul Cadach * channels/h323/ast_h323.cxx: Fix preparation of type and presentation of calling number 2006-10-03 00:01 +0000 [r44240] Matt O'Gorman * doc/jingle.txt, channels/chan_jingle.c (removed), include/asterisk/jabber.h, configs/jingle.conf.sample (removed), res/res_jabber.c: updated res_jabber for even better component support, soon will be jep-0100 compliant. also removed chan_jingle and infromed info from jingle.txt, chan_gtalk still works and should be used in this version. 2006-10-02 20:11 +0000 [r44199-44215] Joshua Colp * channels/chan_sip.c: Change the fd on the I/O context in case it changed during the reload, which is indeed possible. (issue #7943 reported by eclubb) * contrib/init.d/rc.redhat.asterisk: We should be using $AST_SBIN instead of hardcoding the path for the error message (issue #7942 reported by eclubb) 2006-10-02 18:52 +0000 [r44186] Paul Cadach * configs/users.conf.sample, pbx/pbx_config.c: Missed part of userconf functionality for chan_h323 2006-10-02 17:25 +0000 [r44169] Joshua Colp * main/io.c: Shrink when current_ioc is unused. It is set to -1 when unused, not 0. (issue #7941 reported by eclubb) 2006-10-02 17:16 +0000 [r44166-44167] Paul Cadach * doc/realtime.txt: Typo fix * channels/chan_h323.c: Optimization of oh323_indicate(): less locks - less problems, plus single exit point 2006-10-02 02:38 +0000 [r44146] Mark Spencer * channels/chan_sip.c, channels/chan_iax2.c: Don't use Channel when you're not talking about a channel :) 2006-10-01 19:32 +0000 [r44135] Paul Cadach * channels/chan_h323.c: Do not simulate any audio tones if we got PROGRESS message 2006-10-01 18:30 +0000 [r44111-44125] Russell Bryant * Makefile: Fix a problem that cuased AST_DATA_DIR in defaults.h to be empty. The cause is that since ASTDATADIR is explicitly exported using "export ASTDATADIR" at the top of the Makefile, make no longer considers the variable "undefined", so the Makefile can't use ?= to set ASTDATADIR if not yet set. (issue #8063, reported by akohlsmith, fixed by me) * configs/queues.conf.sample: Fix the name of the "eventmemberstatus" option in the sample queues.conf (issue #8065, adamg) 2006-10-01 15:01 +0000 [r44109] Luigi Rizzo * channels/chan_sip.c: sync with trunk - move variable declarations to the beginning of a block. 2006-09-30 19:20 +0000 [r44090] Paul Cadach * main/rtp.c: Allow one-way RTP streams (device->Asterisk) 2006-09-30 16:28 +0000 [r44080] Luigi Rizzo * codecs/lpc10/Makefile, Makefile, main/Makefile: fix two recent build problems: - with AST_DEVMODE, building codecs/lpc10 fails because of lots of warnings, and the configure step in editline fails as well. Fix this by removing the -Werror in these steps. - on FreeBSD (but probably on other platforms as well), the final link of asterisk fails because AST_LIBS was not exported to the subdirs Makefiles. Add a proper fix in the top-level Makefile (a possible alternative way is to add "export AST_LIBS" near the beginning of the file). With this fix, i believe that some of the platform-specific conditionals in main/Makefile are redundant (because they should be already dealt with in the top level Makefile) but i don't have a platform to check. Merging to head will happen in a moment. 2006-09-30 16:12 +0000 [r44068-44078] Paul Cadach * channels/chan_sip.c: Fix issue #7928 correctly. Next is a comment of previous fix: Issue #7928 - Don't send both 404 and 503. Fix by phsultan with a small fix by me, myself or I. Thanks, Philippe! (This was caused by my changes to the transaction handling) * channels/chan_sip.c: Found some buggy SIP clients (phones Planet VIP-153T firmware 1.0, Linksys PAP2 firmware 3.1.9(LSc)) which sends ACK not on OK message only (when remote party answers) but on RINGING message too, so when we send 200 OK message, we get unidentified ACK message (because INVITE acknowledged on RINGING message already), so 200 OK retransmits within its retransmission interval then call gets dropped. If someone else knows how to provide workaround for such cases, please, fix it in correct way. Thanks to ssh from #asteriskru for provide access to his box to study and fix this case. 2006-09-29 22:51 +0000 [r44055-44057] Kevin P. Fleming * agi, utils: ignore temporary files made by the Makefiles during a build * codecs/lpc10/Makefile, main/db1-ast/Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, configure, build_tools/embed_modules.xml, codecs/gsm/Makefile, configure.ac, Makefile.moddir_rules, Makefile.rules, codecs/ilbc/Makefile, pbx/Makefile, res/Makefile, channels/Makefile: fix a few build system bugs, and convert Makefiles to be compatible with GNU make 3.80 2006-09-29 22:35 +0000 [r44053] Jason Parker * main/asterisk.c, main/cli.c: Fix a bug with the removal of 'atleast' argument to 'core verbose' and 'core debug'. Add that argument back in. 2006-09-29 21:09 +0000 [r44022-44043] Paul Cadach * channels/h323/ast_h323.cxx: Set TON/PRESENTATION information more carefully when no CallingNumber IE available * channels/h323/ast_h323.cxx: Fake display name by called number on incoming calls (until passing connected number/connected name is not implemented) * channels/h323/ast_h323.cxx: Ported code refers to H.450 - add includes * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Properly pass TON/PRESENTATION information - original H323Connection::SendSignalSetup() destroys Q.931 fields. 2006-09-29 18:49 +0000 [r44011-44012] Kevin P. Fleming * main/Makefile: yet another place where we were not using the correct CFLAGS by default * main/Makefile: missed one conversion to ASTCFLAGS 2006-09-29 18:30 +0000 [r44009] Paul Cadach * channels/h323/ast_h323.cxx, channels/chan_h323.c, channels/h323/ast_h323.h, channels/h323/chan_h323.h: Pass TON/PRESENTATION information too 2006-09-29 18:25 +0000 [r43952-44008] Kevin P. Fleming * main/db1-ast/Makefile, Makefile, codecs/Makefile, utils/Makefile, main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules, Makefile.rules, pbx/Makefile, channels/Makefile: don't abuse CFLAGS and LDFLAGS for build of Asterisk components, because they are also then used for non-Asterisk components (like menuselect); use our own variables instead * configure, configure.ac: support --without-curl in configure script * Makefile.rules: another cross-compile fix * Makefile: a couple more environment settings that can't leak into the menuselect build * main/cli.c: proper fix for ast_group_t change * include/asterisk/lock.h: eliminate compiler warning when DEBUG_CHANNEL_LOCKS is enabled and users of this header file don't also include channel.h 2006-09-28 20:11 +0000 [r43944] Jason Parker * apps/app_queue.c: Fix incorrect argument order for member names, on persisted members. Issue 8047, patch by jmls. 2006-09-28 18:05 +0000 [r43932-43933] Joshua Colp * apps/app_playback.c, res/res_monitor.c, include/asterisk/logger.h, channels/chan_misdn.c, res/res_smdi.c, channels/chan_skinny.c, apps/app_rpt.c, channels/chan_mgcp.c, main/udptl.c, main/frame.c, funcs/func_timeout.c, channels/chan_sip.c, apps/app_festival.c, channels/iax2-provision.c, apps/app_alarmreceiver.c, res/res_musiconhold.c, apps/app_followme.c, channels/chan_iax2.c: Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka) 2006-09-28 17:35 +0000 [r43919] Kevin P. Fleming * apps/app_queue.c: fix buggy (and overly complex) loop used during reload of app_queue for static member list updating 2006-09-28 17:34 +0000 [r43918] Paul Cadach * channels/h323/ast_h323.cxx: Extend call establishment timeout 2006-09-28 17:31 +0000 [r43913-43915] Joshua Colp * channels/chan_iax2.c: Make sure the pvt exists before accessing it again as it may have gone away (issue #7562 reported by Seb7 and issue #7939 reported by sorg) * main/cli.c: Warning be gone! 2006-09-28 16:41 +0000 [r43899] BJ Weschke * apps/app_queue.c: app_queue is comparing the device names incorrectly while checking their statuses. It's internal list of interfaces includes the dial string, while the argument passed to this function does not have the dial string (/n for a local channel). This causes it to ignore the device state changes because it thinks it belongs to none of its members. (#8040 reported and patch by tim_ringenbach) 2006-09-28 16:17 +0000 [r43893] Joshua Colp * apps/app_meetme.c: Stop the stream after waitstream returns so that our formats get restored. (issue #7370 reported by kryptolus) 2006-09-28 15:56 +0000 [r43877] Paul Cadach * channels/h323/ast_h323.cxx: Fix compiler warning 2006-09-28 15:29 +0000 [r43864-43873] BJ Weschke * apps/app_queue.c: Fix race conditioon crash with get_member_status (#7864 - tim_ringenbach reported and patched) * apps/app_queue.c: Autopause not working for queue members. (#8042 - jmls reported and patch) 2006-09-28 12:58 +0000 [r43861-43862] Paul Cadach * channels/h323/ast_h323.cxx, channels/h323/ast_h323.h: Force remote side to start media on outgoing PROGRESS message * include/asterisk/compiler.h: Put attribute tag at correct place 2006-09-28 11:03 +0000 [r43852] Christian Richter * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c: fixed a bug which led to chan_list zombies, when the call could not be properly established in misdn_call. also removed the ACK_HDLC stuff which is not really needed. 2006-09-28 10:51 +0000 [r43843-43846] Paul Cadach * channels/h323/ast_h323.cxx: Do not open transmit channel until TCS is received * main/file.c: Don't warn on HOLD/UNHOLD control frames * main/file.c: Don't treat unknown control frames as voice 2006-09-27 20:21 +0000 [r43816] Tilghman Lesher * apps/app_voicemail.c: Avoid inability to lock directory log message by creating the directory ahead of time. (Issue 7631) 2006-09-27 19:44 +0000 [r43801-43803] Jason Parker * apps/app_playback.c, main/pbx.c: Fix an issue with PLAYBACKSTATUS not being set under certain circumstances. Fix a minor issue, to make it use the filenames that were parsed, instead of the entire argument string. Fix Background() to return -1 like Playback(), if no args are specified. 2006-09-27 19:10 +0000 [r43783-43798] Joshua Colp * main/rtp.c: Compensate for out of order packets better if RFC2833 compensation is turned on. * channels/chan_iax2.c: Get rid of two functions from a time now past (we THINK these are from pre-recursive lock time) that may be contributing to two open issues on the bug tracker (7562/7939) and that has the potential to just make bad things happen if the timing is right. 2006-09-27 16:55 +0000 [r43779] Russell Bryant * main/channel.c,res/res_features.c: Fix a problem that occurred if a user entered a digit that matched a bridge feature that was configured using multiple digits, and the digit that was pressed timed out in the feature digit timeout period. For example, if blind transfer is configured as '##', and a user presses just '#'. In this situation, the call would lock up and no longer pass any frames. (issue #7977 reported by festr, and issue #7982 reported by michaels and valuable input provided by mneuhauser and kuj. Fixed by me, with testing help and peer review from Joshua Colp). There are a couple of issues involved in this fix: 1) When ast_generic_bridge determines that there has been a timeout, it returned AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls ast_generic_bridge over again with the same timestamp for the next event. This results in an endless loop of nothing until the call is terminated. This is resolved by simply changing ast_generic_bridge to return AST_BRIDGE_COMPLETE when it sees a timeout. 2) I also changed ast_channel_bridge such that if in the process of calculating the time until the next event, it knows a timeout has already occured, to immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the channels anyway. 3) In the process of testing the previous two changes, I ran into a problem in res_features where ast_channel_bridge would return because it determined that there was a timeout. However, ast_bridge_call in res_features would then determine by its own calculation that there was still 1 ms before the timeout really occurs. It would then proceed, and since the bridge broke out and did *not* return a frame, it interpreted this as the call was over and hung up the channels. The reason for this was because ast_bridge_call in res_features and ast_channel_bridge in channel.c were using different times for their calculations. channel.c uses the start_time on the bridge config, which is the time that the feature digit was recieved. However, res_features had another time, 'start', which was set right before calling ast_channel_bridge. 'start' will always be slightly after start_time in the bridge config, and sometimes enough to round up to one ms. This is fixed by making ast_bridge_call use the same time as ast_channel_bridge for the timeout calculation. ........ 2006-09-27 16:24 +0000 [r43775] Christian Richter * channels/chan_misdn.c, channels/Makefile: removed the chan_misdn versioning, since Asterisk has it's own 2006-09-27 16:23 +0000 [r43774] Joshua Colp * channels/chan_sip.c: Make rfc2833compensate a global option. 2006-09-27 04:35 +0000 [r43756] Russell Bryant * apps/app_voicemail.c: Backport revision 43754 from the trunk, which removes an unused buffer from mm_login to close bug 8038, as well as addresses some formatting and coding guidelines issues in passing. Originally, I did not commit this to 1.4 since it is not necessarily fixing a bug. However, since the IMAP storage code is brand new, I decided it would be better to make the change here as well, in case someone has to work on this code to address issues in the very near future. I don't want to make unnecessary merge problems going to the trunk. 2006-09-27 02:32 +0000 [r43739] Steve Murphy * configs/extensions.ael.sample: This change to extensions.ael was to fix bug 8031; the install scripts are causing it to be copied to /etc/asterisk/extensions.ael, and because it is a fairly direct conversion of the original extensions.conf, the macro and context names clash with the existing extensions.conf. So, I put an ael- in front of all macros and contexts, and checked every goto and macro call. Also, this file compiles under aelparse. 2006-09-26 20:56 +0000 [r43710] Russell Bryant * main/asterisk.c: Back in revision 4798, this message was changed from using ast_cli() to directly calling write(). During this change, checking if this was a remote console was removed. This caused this message about using "exit" or "quit" to exit an Asterisk console to come up in times where it did not make sense. This change restores the check to see if this is a remote console before printing the message. (fixes BE-65) 2006-09-26 20:47 +0000 [r43707] Joshua Colp * .cleancount, main/cli.c, channels/chan_sip.c, include/asterisk/channel.h: Use proper type to represent the group variable (issue #8025 reported by makoto) 2006-09-26 20:30 +0000 [r43700-43703] Russell Bryant * channels/chan_sip.c: Add missing newline character in the warning message about deprecated TOS values in configuration. * apps/app_voicemail.c: When parsing the sections of voicemail.conf that contain mailbox definitions, don't introduce a length limit on the definition by using a 256 byte temporary storage buffer. Instead, make the temporary buffer just as big as it needs to be to hold the entire mailbox definition. (fixes BE-68) 2006-09-26 20:19 +0000 [r43695-43697] Joshua Colp * channels/chan_local.c: Strip options off the argument passed for devicestate in chan_local. (issue #8034 reported by pcardozo) * apps/app_chanspy.c, main/channel.c, main/slinfactory.c: Slight overhaul of the whisper support. 1. We need to duplicate the frame from ast_translate 2. We need to ensure we always have signed linear coming in for signed linear combining. 3. We need to ensure we are always feeding signed linear out. 4. Properly store and restore write format when beeping on the channel we are whispering on. 5. Properly discontinue the stream on the channel for the beep. (issue #8019 reported by timkelly1980) 2006-09-26 18:34 +0000 [r43676] Kevin P. Fleming * sounds/Makefile: update to use 1.4.3 core sounds, with corrected beep/beeperr/tt-monkeys files 2006-09-26 18:08 +0000 [r43650-43674] Jason Parker * doc/rtp-packetization.txt, main/frame.c: Issue #8015, patch by Dan Austin. Maximum values were incorrect, which is why this is being put in 1.4 * channels/chan_skinny.c: Add proper codec support to chan_skinny. Works with at least ulaw, alaw, and g729a. This is technically a "new feature", but there are justifications for it. I found a bug with the recent rtp packetization changes, which caused the media setup to fail under certain circumstances, particularly when using allow=all, or having no allow= statements (globally or on the device). I could have either removed the rtp packetization features, or I could add proper codec support (which, without, I think most people would consider to be a bug anyways). 2006-09-25 22:07 +0000 [r43640-43642] Tilghman Lesher * apps/app_voicemail.c: Should have moved these lines up in the merge, instead of removing them * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue 7824): 1) delete=yes was ignored 2) maxmessages was ignored 2006-09-25 21:26 +0000 [r43626-43635] Paul Cadach * channels/h323/cisco-h225.cxx, channels/h323/cisco-h225.h, channels/h323/cisco-h225.asn: Fix ASN1 description of non-standard Cisco extensions * channels/h323/ast_h323.cxx, channels/chan_h323.c: Backport changes of trunk: 1) r43540: Avoid possible deadlock on channel destruction 2) r43590: Disable fastStart if requested by remote side 2006-09-25 15:23 +0000 [r43616] Jason Parker * sounds/Makefile: One more fix for sounds installation - this time for portability. Reported to asterisk-dev mailing list. 2006-09-25 14:52 +0000 [r43605] Steve Murphy * formats/format_ogg_vorbis.c: This tiny fix prevents asterisk from crashing if trying to play an OGG moh file. 2006-09-25 06:15 +0000 [r43582] Paul Cadach * channels/h323/caps_h323.cxx, channels/h323/compat_h323.h, channels/chan_h323.c: Merged revisions 43472,43495 from trunk 2006-09-24 14:58 +0000 [r43553-43564] Russell Bryant * channels/iax2-provision.c: Fix a CLI command registration issue where an erroneous message claiming that "iax2 show provisioning" was already registered. This was because this command was registering itself as both the command, as well as the command it is deprecating. (issue #8022, reported by bjweeks, fixed by myself) * channels/chan_iax2.c:Check to see if the channel that is activating the IAXPEER function is actually an IAX2 channel before proceeding to process it to avoid crashing. (issue #8017, reported by admott, fixed by myself) 2006-09-22 23:44 +0000 [r43524] Kevin P. Fleming * Makefile: don't output the 'build complete' message when the target being run is already going to do an installation 2006-09-22 22:12 +0000 [r43518] Jason Parker * channels/chan_skinny.c: Allow chan_skinny.so to be unloaded properly. Remove reload support, since it doesn't actually...work. 2006-09-22 21:36 +0000 [r43505-43508] Steve Murphy * pbx/pbx_ael.c: This commits a change to return MODULE_LOAD_FAILURE on error, and SUCCESS (instead of 0) when all goes well for bug 8004 * pbx/pbx_ael.c: If the extensions.ael file not found, or unreadable, we return AST_MODULE_LOAD_DECLINE, as per bug # 8004. 2006-09-22 17:25 +0000 [r43492] Jason Parker * main/cli.c: Make sure we explicitly set the CLI command to not be deprecated, if it isn't. 2006-09-22 16:42 +0000 [r43486-43489] Kevin P. Fleming * sounds/Makefile: use rebuilt extra sounds * main/channel.c: all the Linux systems I have don't use '__m_count' for this field, so I don't know where this came from... 2006-09-22 15:47 +0000 [r43477-43484] Russell Bryant * include/asterisk/threadstorage.h: backport the compatability fix to use attribute_malloc instaed of __attribute__ ((malloc)) * channels/chan_misdn.c: return AST_MODULE_LOAD_DECLIDE if mISDN could not be configured (issue #8006, Mithraen) * main/frame.c: Suppress a compiler warning about the use of a potentially uninitialized variable. It couldn't actually happen, though. 2006-09-22 03:01 +0000 [r43469] Jason Parker * channels/chan_skinny.c: First shot at unload_module in chan_skinny.. More to come. 2006-09-21 23:50 +0000 [r43466] Matt O'Gorman * include/asterisk/jabber.h, channels/chan_gtalk.c, res/res_jabber.c: updates for better compontent support 2006-09-21 23:24 +0000 [r43464] Tilghman Lesher * res/res_odbc.c, configs/res_odbc.conf.sample: Twould help if we actually documented how the new features in res_odbc actually work. (Oops) 2006-09-21 22:21 +0000 [r43454-43456] Joshua Colp * channels/chan_oss.c: Some more clean up in the load function for chan_oss (issue #8002 reported by Mithraen with minor mods by moi) * channels/chan_mgcp.c: Clean up chan_mgcp's module load function (issue #8001 reported by Mithraen with mods by moi) 2006-09-21 21:21 +0000 [r43450] Kevin P. Fleming * main/Makefile, build_tools/strip_nonapi (added): add another attempt to strip non-API symbols from the final binary... script will need to be extended to work on non-Linux systems 2006-09-21 20:22 +0000 [r43410-43445] Tilghman Lesher * apps/app_url.c: Fix documentation to reflect how Url() really works * cdr/cdr_tds.c, configure, configure.ac: TDS 0.64 updates 2006-09-21 Kevin P. Fleming * Asterisk 1.4.0-beta2 released. 2006-09-21 16:08 +0000 [r43404-43405] Kevin P. Fleming * main/Makefile: remove this change... it requires binutils 2.17 2006-09-20 23:19 +0000 [r43396] Jason Parker * build_tools/make_version: fix minor typo in the way version is handled 2006-09-20 Kevin P. Fleming * Asterisk 1.4.0-beta1 released.