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Diffstat (limited to 'ChangeLog')
| -rw-r--r-- | ChangeLog | 4270 |
1 files changed, 0 insertions, 4270 deletions
diff --git a/ChangeLog b/ChangeLog deleted file mode 100644 index 720041cb2..000000000 --- a/ChangeLog +++ /dev/null @@ -1,4270 +0,0 @@ -2006-12-12 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.14 released - -2006-12-12 05:11 +0000 [r48403] Kevin P. Fleming <kpfleming@digium.com> - - * sounds/silence (added), sounds/silence/1.gsm (added), - sounds/silence/10.gsm (added), sounds/silence/2.gsm (added), - sounds/silence/3.gsm (added), sounds/silence/4.gsm (added), - sounds/silence/5.gsm (added), sounds/silence/6.gsm (added), - sounds/silence/7.gsm (added), sounds/silence/8.gsm (added), - sounds/silence/9.gsm (added): add silence files - -2006-12-11 23:00 +0000 [r48394-48398] Matt O'Gorman <mogorman@digium.com> - - * Makefile, apps/app_externalivr.c, sounds.txt: 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 00:33 +0000 [r48374] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * 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: 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 02:14 +0000 [r48371] Steve Murphy <murf@digium.com> - - * channels/chan_zap.c: This version applies the patch suggested by - stevens in bug 7836 (make inbound channel RINGING state - consistent with other channels). - -2006-12-09 15:45 +0000 [r48361] Russell Bryant <russell@digium.com> - - * 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. - -2006-12-07 18:14 +0000 [r48356] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: Ensure that the file position is not - incremented beyond the total number of files available for - playback. (issue #8539, ulogic) - -2006-12-06 16:05 +0000 [r48322] Russell Bryant <russell@digium.com> - - * configs/iax.conf.sample: Fix the name of the rtignoreregexpire - option in the sample configuration file. (issue #8526, arkadia) - -2006-12-06 15:48 +0000 [r48321] Christian Richter <christian.richter@beronet.com> - - * doc/README.misdn, channels/chan_misdn.c, - channels/misdn/isdn_msg_parser.c: added the export and import of - the MISDN_ADDRESS_COMPLETE Variable to inidcate wether the - extension is already completely dialed or if there might come - additional digits by information elements. also added some docs - for that. - -2006-12-06 15:42 +0000 [r48320] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8528 - make sure we don't delete the - dialog too quickly after receiving a 487. Move 487 handling into - handle_response_invite where it really belongs and don't add an - ALREADYGONE flag to the dialog. - -2006-12-06 14:35 +0000 [r48319] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: changed a few debugs to higher debug - levels - -2006-12-06 12:14 +0000 [r48272-48315] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't add Contact header on BYE, CANCEL, - MESSAGE requests (Bye, Cancel backported from 1.4, MESSAGE bug - reported to me by Gunnar at Omnitor) - - * channels/chan_sip.c: Only set the ALREADYGONE flag once in - handle_response() - -2006-12-05 01:26 +0000 [r48251] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: 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 21:20 +0000 [r48236-48246] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c: 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. - - * apps/app_voicemail.c: 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 18:14 +0000 [r48233] Joshua Colp <jcolp@digium.com> - - * channel.c: 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. - -2006-12-01 23:30 +0000 [r48192] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_dial.c: 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 20:19 +0000 [r48183] Jason Parker <jparker@digium.com> - - * configs/extensions.conf.sample: Fix a small typo - issue 8848, - reported by pabelanger - -2006-11-30 20:47 +0000 [r48165] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 8319 - noriyuki - nonce-count updated - *after* use - -2006-11-30 20:27 +0000 [r48142-48161] Joshua Colp <jcolp@digium.com> - - * channel.c: 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: Only print out debug message if bridged - channel is not NULL. (issue #8412 reported by jubilex) - - * res/res_features.c: 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) - - * cdr.c: 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) - - * res/res_features.c: 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: Document 'port' for SIP peers, came up - because of the current mailing list thread. (issue #8450 reported - by blitzrage) - -2006-11-30 09:05 +0000 [r48127] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Do proper test and don't leave dialogs - hanging... - -2006-11-29 16:47 +0000 [r48053-48106] Joshua Colp <jcolp@digium.com> - - * rtp.c: If the frame was duplicated before writing out then we - need to free it. (issue #8429 reported by edguy3) - - * channels/chan_phone.c: 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: Use the proper function to get the new - message count instead of always using the filesystem. (issue - #8421 reported by slimey) - -2006-11-27 17:15 +0000 [r48045] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_musiconhold.c: Random MOH wasn't really random (bug 8381) - -2006-11-27 15:30 +0000 [r48037] Joshua Colp <jcolp@digium.com> - - * pbx/pbx_spool.c: Do not reference the freed outgoing structure in - the debug message. (issue #8425 reported by arkadia) - -2006-11-24 14:33 +0000 [r47987] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Change some logging levels. Not having hints - is not an ERROR, but still should be reported. - -2006-11-23 16:10 +0000 [r47968] Christian Richter <christian.richter@beronet.com> - - * channels/misdn_config.c, channels/chan_misdn.c, - channels/misdn/isdn_lib.c: fixed a litle bug regarding - HOLD/RETRIEVE. beatufied some logs, changed some loglevels. - changed the default value of block_on_alarm - -2006-11-23 10:54 +0000 [r47958] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Remove unused variable (rizzo) - -2006-11-22 02:19 +0000 [r47910] Steve Murphy <murf@digium.com> - - * channel.c: This is the fix for bug 8386, wherein the time-limit - args to dial didn't work correctly - -2006-11-20 19:59 +0000 [r47862] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Failing to trap -1 error from mmap causes - segfault (Issue 8385) - -2006-11-20 19:50 +0000 [r47855-47859] Joshua Colp <jcolp@digium.com> - - * frame.c: Don't forget to byte swap if we are exiting the smoother - feed early. (issue #8287 reported by arturs) - - * channels/chan_sip.c: Free history items at the end of use of the - temporary SIP pvt structure. (issue #8383 reported by benh) - -2006-11-20 10:17 +0000 [r47842] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Just to be safe, disable all the scheduled - items after deleting a scheduler entry (rizzo) - -2006-11-17 19:02 +0000 [r47802] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: backport proper channel_find_locked behavior from 1.4 - branch (noted by Steve Davies on asterisk-dev list) - -2006-11-16 23:16 +0000 [r47780] Jason Parker <jparker@digium.com> - - * apps/app_dial.c, apps/app_cut.c, apps/app_directory.c, - apps/app_db.c: Fix a couple of typos in applications.. Initially - spotted by mrobinson. - -2006-11-16 22:57 +0000 [r47776] Kevin P. Fleming <kpfleming@digium.com> - - * doc/README.cdr: update clearly wrong documentation regarding - cdr_custom - -2006-11-16 20:29 +0000 [r47750-47761] Joshua Colp <jcolp@digium.com> - - * cdr/Makefile: Look for the header file specifically in all cases, - not just the existence of the directory. (issue #8358 reported by - mrness) - - * channels/chan_local.c: 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) - -2006-11-16 16:44 +0000 [r47743] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't fixup if we haven't got PVT. - Suggestion from Martin Vit on -dev mailing list inspired by - file's commit to chan_local. "This shouldn't happen" ;-) - -2006-11-15 22:29 +0000 [r47711] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c: 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 21:18 +0000 [r47705] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: CANCEL requests are never authenticated - (according to RFC 3261) - -2006-11-15 20:30 +0000 [r47666-47696] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c: correct argument name typo that caused - global variable to be used instead of the one for the specified - voicemail user - - * config.c: when re-writing the config file, don't repeat the path - if it hasn't changed - - * config.c: when appending a list of variable to a category, ensure - the tail pointer points to the last variable in the list - - * config.c: clear the category's variable tail pointer as well when - variables are detached from it - - * config.c: ouch... don't use printf, use ast_log/ast_verbose - - * apps/app_voicemail.c, include/asterisk/config.h: 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 - -2006-11-15 15:17 +0000 [r47648-47655] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Send error message if we fail to allocate - sip socket, possibly caused by too few file handles (wasn't - possible before, but with the new way of sending temp messages, - it is). Found this bug under heavy load testing with SIPP. - - * channels/chan_sip.c: Sending 200 OK and not getting ACK is - considered critical for the call. - -2006-11-14 22:15 +0000 [r47631] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Update copyright information in the ADSI - logo blob. - -2006-11-14 11:06 +0000 [r47596] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Avoid collissions between the peerpoke - system and the retransmits. Issue #8272. In some cases, changed - timers caused the retransmit system to destroy the dialog before - peerpoke_expire got a chance. - -2006-11-13 21:26 +0000 [r47583] Joshua Colp <jcolp@digium.com> - - * cdr/cdr_pgsql.c: Initialize global pointers for connection and - result to NULL. (issue #8356 reported by james) - -2006-11-13 20:18 +0000 [r47580] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Having more than 255 old messages caused - corruption in the new/old count - -2006-11-13 19:04 +0000 [r47571] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't send 487 if we've already sent 200 OK - on invite at time of receiving a BYE in the same transaction. - (SIPP testing) - -2006-11-13 17:05 +0000 [r47549] Joshua Colp <jcolp@digium.com> - - * apps/app_sms.c: When sending an SMS with a user data header - properly set the UDH flag in the first byte. (issue #8347 - reported by hoffmeis) - -2006-11-13 05:45 +0000 [r47522-47525] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_odbc.c: If the execute fails a second time, make sure - that we don't pass back a stale handle - - * channels/chan_zap.c: Don't play dialtone if the seizing the - channel fails (Bug 7754) - -2006-11-12 06:09 +0000 [r47496] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: 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) - -2006-11-10 20:46 +0000 [r47452-47470] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Clear dialog on loop (backport from 1.4 by - mistake) - - * channels/chan_sip.c: - Don't check for ignore in blocks that - isn't reached if ignore is on... - return properly after sending - reply in handle_request_invite - - * channels/chan_sip.c: Fix multipart/mixed SDP support (issue 8010, - alphaque) - -2006-11-09 16:48 +0000 [r47379] Joshua Colp <jcolp@digium.com> - - * channels/chan_phone.c: 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 13:09 +0000 [r47359] Christian Richter <christian.richter@beronet.com> - - * channels/misdn_config.c, channels/chan_misdn.c, - channels/misdn/chan_misdn_config.h: Fixed segfault when no - misdn.conf exists, reported by Igor Neves, thanks. - -2006-11-08 07:40 +0000 [r47307-47308] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Remove dialog properly at unload of module - (rizzo) - -2006-11-07 18:22 +0000 [r47274] Steve Murphy <murf@digium.com> - - * include/asterisk/channel.h, channel.c: This mod for bug_7506, to - make the manager code output the proper event - -2006-11-07 13:02 +0000 [r47248] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't ever reply to an ACK. (Issue 8265) - -2006-11-07 01:22 +0000 [r47238] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: 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) - -2006-11-02 17:47 +0000 [r46964] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: ignore files in a music on hold directory - that begin with '.' (issue #8249, cboie) - -2006-11-02 15:15 +0000 [r46899] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't overwrite flags in the packet - -2006-11-02 13:55 +0000 [r46876] Russell Bryant <russell@digium.com> - - * callerid.c: Add a missing call to free before returning in an - error condition (issue #8268, mrness) - -2006-11-01 21:20 +0000 [r46838] Matt O'Gorman <mogorman@digium.com> - - * logger.c: fix for bug #8083 crash caused by double free on m->msg - -2006-11-01 19:52 +0000 [r46803] Steve Murphy <murf@digium.com> - - * 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 18:24 +0000 [r46776] Russell Bryant <russell@digium.com> - - * res/res_monitor.c: 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) - -2006-10-31 15:46 +0000 [r46662] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_curl.c: 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 09:49 +0000 [r46585-46610] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: Another try to fix - ;rport NAT traversal support (issue #7473) - - * channels/chan_sip.c: If peer fails ACL check, fail the REGISTER - attempt - - * channels/chan_sip.c: On the other hand, we already copy the NAT - flags... Reverting. - - * channels/chan_sip.c: Issue 7473 - support ;rport on REGISTER - requests too. - -2006-10-31 06:18 +0000 [r46557-46560] Russell Bryant <russell@digium.com> - - * utils.c: 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: fix some copy/paste bugs in the checking of - arguments for the "control stream file" AGI command (issue #8255, - mnicholson) - -2006-10-30 16:00 +0000 [r46402-46430] Olle Johansson <oej@edvina.net> - - * rtp.c: Bind rtcp to proper IP address - - * channels/chan_sip.c: Issue #7869 - Stop sending 302 redirect when - not getting an answer... - - * channels/chan_sip.c: issue #7608: Notifications with wrong - content-type. Reported by jsiddall. - -2006-10-27 17:36 +0000 [r46361] Russell Bryant <russell@digium.com> - - * res/res_agi.c, asterisk.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) - -2006-10-27 09:24 +0000 [r46350] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: - 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 - -2006-10-26 20:06 +0000 [r46344] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7240, by mistake only committed to - trunk (now 1.4), reported by edgreenberg in Issue #7966. Thanks - Ed! - -2006-10-26 17:47 +0000 [r46332-46337] Jason Parker <jparker@digium.com> - - * contrib/scripts/astgenkey.8: oops - somebody forgot to change - this - long ago, probably. - - * channels/chan_skinny.c: Remove a useless ast_mutex_unlock. Issue - #8186, patch by anthonyl (fix suggested by benh). - -2006-10-25 19:28 +0000 [r46213-46258] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Working to resolve #7608 - adding debug - output - - * channels/chan_sip.c: Fix the attack shield for 1.2 too. REFER and - NOTIFY can create dialogs in the world of Asterisk. - -2006-10-25 08:41 +0000 [r46176] Christian Richter <christian.richter@beronet.com> - - * channels/misdn_config.c, channels/chan_misdn.c, - channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample: - added nttimeout option to configure wether we disconnect calls on - NT timeouts or not during an overlapdial session - -2006-10-23 00:25 +0000 [r45927] Joshua Colp <jcolp@digium.com> - - * cdr/cdr_odbc.c: Don't leak memory mmmk? - -2006-10-21 12:35 +0000 [r45808] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: 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. - -2006-10-19 17:16 +0000 [r45691] Joshua Colp <jcolp@digium.com> - - * apps/app_externalivr.c: Respect language selection when seeing if - the file exists (issue #8178 reported by mnicholson) - -2006-10-17 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.13 released - -2006-10-17 20:37 +0000 [r45380] Joshua Colp <jcolp@digium.com> - - * 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 17:50 +0000 [r45332] Jason Parker <jparker@digium.com> - - * channels/chan_skinny.c: Fix an integer signedness problem. - -2006-10-17 17:22 +0000 [r45326] Kevin P. Fleming <kpfleming@digium.com> - - * LICENSE: provide licensing language for IAXy firmware file - -2006-10-17 15:50 +0000 [r45306] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c, configs/sip.conf.sample: After some - research, we realized that the default behaviour since a long - time was doing the right thing, even though the change optimized - a bit and removed a lot of potential risks. Conclusion: No need - for a configuration option at all. - -2006-10-16 19:59 +0000 [r45260-45265] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample: Use responses - rather then replies even though they mean the same thing. - - * channels/chan_sip.c, configs/sip.conf.sample: Add - 'ignoreoodreplies' option which will not create a pvt structure - on a SIP response but instead basically drop it. - -2006-10-14 00:16 +0000 [r45134] Steve Murphy <murf@digium.com> - - * pbx/pbx_ael.c: Made a small update to solve bug 8128; The - switch-case fallthru goto to a pattern extension needed to - resolved the wildcards to an appropriate digit for extension - matching to work - -2006-10-13 22:57 +0000 [r45119] Kevin P. Fleming <kpfleming@digium.com> - - * 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 19:27 +0000 [r45090] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: avoiding warning, fixing potential bug - (backported from 1.2) - -2006-10-13 17:01 +0000 [r45060] Joshua Colp <jcolp@digium.com> - - * apps/app_chanspy.c: Turn on volume adjustment if it needs to be - on (issue #8136 reported by mnicholson) - -2006-10-13 16:18 +0000 [r45048] Kevin P. Fleming <kpfleming@digium.com> - - * 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 15:49 +0000 [r45030] Joshua Colp <jcolp@digium.com> - - * 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 13:11 +0000 [r44993-45020] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.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 18:31 +0000 [r44955] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/utils.h, channels/chan_sip.c, utils.c, - netsock.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) - -2006-10-10 13:34 +0000 [r44785] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c, channels/misdn/isdn_lib.c: (re)added - support of dynamical enabling hdlc on bchannels - -2006-10-09 14:36 +0000 [r44757] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #8101 - wrong parameter for screening - in remote-party-id - -2006-10-06 16:52 +0000 [r44501-44580] Joshua Colp <jcolp@digium.com> - - * file.c: Even more frames to treat as though the remote side - disappeared (issue #8097 reported by eldadran) - - * file.c: Treat busy control frames as hangup in the file streaming - core (issue #8097 reported by eldadran) - -2006-10-05 10:02 +0000 [r44460] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed segfault which happens during - hold/transfer action - -2006-10-05 01:27 +0000 [r44392-44432] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: fix Polycom presence notification again - - * 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) - -2006-10-04 16:02 +0000 [r44343] Steve Murphy <murf@digium.com> - - * apps/app_macro.c: For bug 7776, I have inserted a warning about - Macro nesting vs. stack limitations - -2006-10-04 15:26 +0000 [r44334-44335] Christian Richter <christian.richter@beronet.com> - - * 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_config.c, channels/misdn/isdn_lib.h, - channels/chan_misdn.c, channels/misdn/chan_misdn_config.h, - configs/misdn.conf.sample, channels/misdn/isdn_lib.c: 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 - -2006-10-03 20:14 +0000 [r44296] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: fix a logic error in my previous fix to the - queue reload code - -2006-10-02 20:07 +0000 [r44168-44213] Joshua Colp <jcolp@digium.com> - - * 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) - - * 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 13:28 +0000 [r44149] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/isdn_lib.c: fixed the hold/retrieve/transfer - issues, removed a useless bc field, added setting of - frame.delivery fields, some minor code cleanups - -2006-10-01 15:19 +0000 [r44110] Russell Bryant <russell@digium.com> - - * configs/queues.conf.sample: Fix the name of the - "eventmemberstatus" option in the sample queues.conf (issue - #8065, adamg) - -2006-09-29 13:44 +0000 [r43977] Kevin P. Fleming <kpfleming@digium.com> - - * cli.c: proper fix for ast_group_t change - -2006-09-28 18:00 +0000 [r43924] Joshua Colp <jcolp@digium.com> - - * frame.c, include/asterisk/logger.h, channels/chan_misdn.c, - channels/chan_sip.c, channels/chan_skinny.c, - funcs/func_timeout.c, apps/app_festival.c, res/res_features.c, - apps/app_hasnewvoicemail.c, apps/app_alarmreceiver.c, - channels/iax2-provision.c, res/res_musiconhold.c, - res/res_monitor.c: Put in missing \ns on the end of ast_logs - (issue #7936 reported by wojtekka) - -2006-09-28 17:31 +0000 [r43916] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: fix buggy (and overly complex) loop used during - reload of app_queue for static member list updating - -2006-09-28 16:37 +0000 [r43897] BJ Weschke <bweschke@btwtech.com> - - * 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 16:32 +0000 [r43895] Kevin P. Fleming <kpfleming@digium.com> - - * cli.c: eliminate compiler warning introduced by recent changes - -2006-09-28 16:13 +0000 [r43891] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Stop the stream after waitstream returns so - that our formats get restored. (issue #7370 reported by - kryptolus) - -2006-09-28 15:18 +0000 [r43871] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: Fix race condion crash with get_member_status - (#7864 - tim_ringenbach reported and patched) - -2006-09-27 20:20 +0000 [r43815] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Avoid inability to lock directory log - message by creating the directory ahead of time. (Issue 7631) - -2006-09-27 19:35 +0000 [r43800] Jason Parker <jparker@digium.com> - - * apps/app_playback.c, pbx.c: Playback() wasn't setting - PLAYBACKSTATUS under several circumstances. Playback() returns -1 - on missing args - so should Background() - -2006-09-27 16:54 +0000 [r43778] Russell Bryant <russell@digium.com> - - * res/res_features.c, channel.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 12:51 +0000 [r43764] Christian Richter <christian.richter@beronet.com> - - * channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/isdn_lib.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. - -2006-09-26 20:49 +0000 [r43708] Russell Bryant <russell@digium.com> - - * 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-4) - -2006-09-26 20:38 +0000 [r43705-43706] Joshua Colp <jcolp@digium.com> - - * .cleancount: I changed the channel structure... let's be sure - this gets updated! - - * channels/chan_sip.c, include/asterisk/channel.h: Use proper type - to represent the group variable (issue #8025 reported by makoto) - -2006-09-26 20:23 +0000 [r43699] Russell Bryant <russell@digium.com> - - * 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) - -2006-09-25 21:14 +0000 [r43634] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Two bugs when forwarding voicemail (Issue - 7824): 1) delete=yes was ignored 2) maxmessages was ignored - -2006-09-24 13:50 +0000 [r43552] Russell Bryant <russell@digium.com> - - * 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 21:53 +0000 [r43509] Joshua Colp <jcolp@digium.com> - - * apps/app_chanspy.c, channel.c: Yay another 'round of spy fixes! - This fixes a small logic flaw with the cleanup function and a - memory allocation issue. (issue #7960 reported by jojo & issue - #7999 reported by aster1) Special thanks to csum77 for letting me - into a box where this issue was happening. - -2006-09-21 17:01 +0000 [r43409-43420] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_rpt.c: Whitespace change... really just an excuse to - test repotools - - * cdr/cdr_tds.c, cdr/Makefile: TDS 0.64 updates - -2006-09-20 05:08 +0000 [r43314] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_misdn.c, channels/chan_sip.c, - channels/chan_skinny.c: make some more functions static - -2006-09-19 16:21 +0000 [r43269] Matt O'Gorman <mogorman@digium.com> - - * pbx/pbx_gtkconsole.c, apps/app_dial.c, channels/chan_sip.c, - apps/app_macro.c, asterisk.c, config.c, apps/app_queue.c, pbx.c: - fixes some verbose vs debug issues. patch from bug 2617 - -2006-09-19 12:28 +0000 [r43248] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: cid is passed to a destructive function; - thus a copy is needed (issue 7961) - -2006-09-18 20:08 +0000 [r43220] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7682 - don't add contacts to 4xx - responses. (Ugly fix, not proud at all) - -2006-09-18 15:30 +0000 [r43163] Joshua Colp <jcolp@digium.com> - - * apps/app_math.c: Add deprecation notice about app_math (issue - #7957 reported by k-egg) - -2006-09-18 15:05 +0000 [r43160] Steve Murphy <murf@digium.com> - - * configs/zapata.conf.sample: Clarified what "callwaiting" does in - zapata.conf. - -2006-09-18 15:05 +0000 [r43159] Joshua Colp <jcolp@digium.com> - - * configs/indications.conf.sample: Add number unobtainable tone for - New Zealand (issue #7969 reported by nic_bellamy) - -2006-09-17 13:54 +0000 [r43072] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_directory.c: Directory used the wrong context for - delivery of 0- and *- keypresses (according to Directory's own - documentation) - Issue 7965 - -2006-09-16 07:57 +0000 [r43003-43019] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_iax2.c: When a realtime peer expires, reset the - ipaddress in the realtime database back to 0 (Issue 6656) - - * apps/app_meetme.c: When the marked user enters the conference, we - should no longer timeout - -2006-09-14 22:16 +0000 [r42946] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_zap.c: Error message references wrong argument - (Issue 7951) - -2006-09-13 19:51 +0000 [r42892] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Backport bugfix patch from 7918 to 1.2 - - msg_cfg destroyed before used - -2006-09-11 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.12.1 released - -2006-09-11 21:47 +0000 [r42697-42783] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_meetme.c, apps/app_page.c: When paging, only wait 5 - seconds for the marked user to enter the conference. After that, - assume the paging already completed by the time the channel - entered the conference and drop back out. (Issue 7275) - - * configs/extensions.conf.sample, configs/alsa.conf.sample, - configs/zapata.conf.sample, configs/iax.conf.sample, - configs/osp.conf.sample, configs/dundi.conf.sample, - configs/enum.conf.sample, configs/vpb.conf.sample, - configs/cdr.conf.sample, configs/voicemail.conf.sample, - configs/phone.conf.sample, configs/misdn.conf.sample, - configs/sip.conf.sample, configs/skinny.conf.sample, - configs/features.conf.sample: Spelling/grammar fixes (Issue 7929) - - * configs/voicemail.conf.sample: Two grammar issues (bug 7927) - -2006-09-09 20:24 +0000 [r42600] Joshua Colp <jcolp@digium.com> - - * channel.c: Only truly consider the channel in the same format if - the format matches the raw format OR if a translation path - already exists to translate between them. (issue #7887 reported - by softins & issue #7803 reported by alvaro_palma_aste). Thanks - goes to stubert for giving me access to a box and showing me a - scenario where this occured. - -2006-09-09 12:14 +0000 [r42535] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Reset proper flag - Don't delete SIP - dialog prematurely Strangely enough imported from svn trunk... - It's confusing here in Greenland. (Committing from 36.000 feet - above Greenland, on the way to asterisk@von - http://www.pulver.com/asterisk ) - -2006-09-08 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.12 released - -2006-09-08 18:50 +0000 [r42452] Joshua Colp <jcolp@digium.com> - - * channel.c: Swap spies during masquerading - -2006-09-08 16:06 +0000 [r42421] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_authenticate.c: Jump logic was backwards: goto returns 0 - if it succeeds, and we should jump if authentication fails. (Bug - #7907) - -2006-09-08 04:37 +0000 [r42402] Joshua Colp <jcolp@digium.com> - - * channels/chan_local.c: Use ast_best_codec to set the read/write - format - -2006-09-07 23:12 +0000 [r42355] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_record.c: Format vulnerability fix - allowing the user - to specify a format is not a good idea (Bug 7811) - -2006-09-07 16:30 +0000 [r42260] Joshua Colp <jcolp@digium.com> - - * cdr.c: Let's use the same thing we use in other places to - calculate our time for ast_cond_timedwait (issue #7697 reported - by bn999) - -2006-09-07 02:14 +0000 [r42150-42200] Steve Murphy <murf@digium.com> - - * logger.c: This should fix the problem reported in 7564: logger - config file errors getting lost because logging isn't configured - yet. The problem was that the code that exists to handle this - case was not getting reached, because other tests were causing an - early return from ast_log(). - - * Makefile: added hours,minutes,seconds .gsm files to the install - portion of the makefile, as per bug 7545 - -2006-09-06 20:02 +0000 [r42148] Joshua Colp <jcolp@digium.com> - - * res/res_agi.c: Don't close the second file descriptor if it's the - same as the first one, as it will have already been closed - elsewhere and could cause massive panic. (issue #7699 reported by - bn999) - -2006-09-06 18:16 +0000 [r42133] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_agent.c: Look ma! No more deadlocks! <sic> As - posted from #7458 and others similar to it in Mantis: p->app_lock - was a mutex really designed for use with agents not in callback - mode. That being the case, I've tried to code it so that when - callback mode is used, the app_lock mutex will not be - locked/unlocked at all. Please let me know how you make out - and - if you continue to deadlock now, please reproduce the deadlock - logging information and post to Mantis. - -2006-09-06 17:10 +0000 [r42110] Christian Richter <christian.richter@beronet.com> - - * channels/chan_misdn.c: fixed pipe consuming bug when using - chanIsAvail (#7878), also moved a debug log to the very begining - of misdn_hangup. - -2006-09-06 15:55 +0000 [r42054-42086] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Make realtime regseconds work as people - expected (0 on registration expiration or release, and actual on - normal state) (issue #7684 reported by kshumard) - - * include/asterisk/chanspy.h, apps/app_chanspy.c, - apps/app_mixmonitor.c, channel.c: Merge in last round of spy - fixes. This should hopefully eliminate all the issues people have - been seeing by distinctly separating what each component - (core/spy) is responsible for. Core is responsible for adding a - spy to a channel, feeding frames to the spy, removing the spy - from a channel, and telling the spy to stop. Spy is responsible - for reading frames in, and cleaning up after itself. - -2006-09-05 16:27 +0000 [r42014] Jason Parker <jparker@digium.com> - - * configs/zapata.conf.sample: Small typo in zapata.conf.sample - Reported by ppyy in 7881 - -2006-09-04 15:46 +0000 [r41989] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't kill the pvt before we have sent ACK - on CANCEL - -2006-09-03 17:38 +0000 [r41827-41882] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: Make sure the forwarded channel inherits - variables appropriately when we receive a call forward in the - queue. (#7867 - raarts reported and patched) - - * apps/app_queue.c: Don't keep trying the same member in certain - strategies when members of the queue are unavailable (#7278 - - diLLec reported and patched) - - * apps/app_chanspy.c: Let's NOT spy on Zap/psuedo channels, - mmmmmmmmk? - - * apps/app_queue.c: Setting a retry of 0 is generally not a good - idea and shouldn't be allowed. (#7574 - reported by regin) - -2006-09-01 22:49 +0000 [r41768] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Only wipe the redirected audio & video - IP/port if it's specified, and trigger a reinvite. - -2006-09-01 17:35 +0000 [r41716] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c, include/asterisk/rtp.h, rtp.c: put in proper - fix for issue #7294 instead of the broken partial fix that was - committed, and thereby also fix issue #7438 - -2006-09-01 16:33 +0000 [r41690-41691] Joshua Colp <jcolp@digium.com> - - * channel.c: Finish up the last commit (was worse then originally - reported) - - * channel.c: Don't treat an unexpected control subclass as voice - (issue #7858 reported by PCadach) - -2006-08-30 19:01 +0000 [r41423] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7572 - Hangup when receiving a buggy - 487 response to an INVITE - -2006-08-30 18:59 +0000 [r41411] Russell Bryant <russell@digium.com> - - * channels/chan_mgcp.c, channels/chan_phone.c, - channels/chan_local.c, channels/chan_misdn.c, - channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_features.c, channels/chan_h323.c, - channels/chan_iax2.c: Restore original functionality of 1.2 in - places where ANI was not set, but was changed to be set. The - original change was done to ensure that the behavior of the - "callerid" option in each channel driver was consistent, but it - caused an unexpected behavior change of CDR records for users, so - this change is being reverted in 1.2. (issue #7695) - -2006-08-30 17:58 +0000 [r41390] Joshua Colp <jcolp@digium.com> - - * include/asterisk/lock.h: Properly handle an ETIMEDOUT result from - pthread_cond_timedwait (issue #7318 reported by arkadia) - -2006-08-30 14:31 +0000 [r41334] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 7822 - don't use SRV lookups if it's - disabled. - -2006-08-29 13:33 +0000 [r41269] Russell Bryant <russell@digium.com> - - * pbx/pbx_config.c: clean up last commit ... most notably, there is - no reason to do heap allocations here, and it also included a - potential memory leak - -2006-08-29 05:49 +0000 [r41239-41262] Steve Murphy <murf@digium.com> - - * pbx/pbx_config.c: Fixes for bug 7813, via patch submitted by - stevens. - - * doc/README.variables: Removed from the docs the mention of the ! - and =~ operators, as these were knocked out of ast_expr2 because - they were new features. Let's hope I can keep them from getting - knocked out of the trunk, too! - - * apps/app_macro.c: According to a note added to 7731 by - mneuhauser, this will repair a break caused by the last fix - (7731). - -2006-08-25 15:21 +0000 [r41066-41069] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Don't send proceeding twice (#7800) - -2006-08-25 15:07 +0000 [r41065] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Text only - clarify the reason for entry - into authentication mode when the skipuser option is ignored - -2006-08-24 19:41 +0000 [r40994] Russell Bryant <russell@digium.com> - - * include/asterisk/linkedlists.h, channel.c, pbx.c: Fix a few - issues related to the handling of channel variables - in - pbx_builtin_serialize_variables(), the variable list traversal - would stop on a variables with empty name/values, which is not - appropriate - When removing the GROUP variables, use - AST_LIST_REMOVE_CURRENT instead of AST_LIST_REMOVE - During - masquerading, when copying the variables list from one channel to - the other, using AST_LIST_INSERT_TAIL is not valid for appending - a whole list. It leaves the tail pointer of the list invalid. - Introduce a new macro, AST_LIST_APPEND_LIST that appends a list - properly. (issue #7802, softins) - -2006-08-24 17:13 +0000 [r40971-40979] Joshua Colp <jcolp@digium.com> - - * configs/zapata.conf.sample: Minor documentation fix to add the - 'dynamic' dialplan option from angler - -2006-08-23 16:05 +0000 [r40901] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_agi.c: Revert last change - breaks retrieval of builtin - variables - -2006-08-22 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.11 released - -2006-08-22 02:59 +0000 [r40821] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_random.c: Bug 7779 - Using initstate(3) means that we - cannot unload this module once loaded. - -2006-08-21 22:34 +0000 [r40798] Matt O'Gorman <mogorman@digium.com> - - * asterisk.c: Move the load_modules call so that if a module needs - realtime support it will work, none do currently but a good move - none the less. - -2006-08-20 22:09 +0000 [r40692] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * CREDITS: Reformat to match the contribution style of other - contributors - -2006-08-20 04:49 +0000 [r40601] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Turn media level c= parsing on by default - (issue #7725 reported by psm) - -2006-08-19 01:03 +0000 [r40446] Jason Parker <jparker@digium.com> - - * apps/app_voicemail.c, apps/app_directory.c: Fix a bug with - app_voicemail when trying to use app_directory to leave messages - to another user (options 3, 5, 2). If the context/extension - didn't exist in the dialplan (and why should it have to?), it - would fail, saying that it's an "invalid extension". Fix was - different in svn trunk. (issue BE-71) - -2006-08-18 19:10 +0000 [r40310-40392] Kevin P. Fleming <kpfleming@digium.com> - - * configs/zapata.conf.sample: make a feeble attempt to avoid the - 'how do I enable my hardware echo canceler' questions - - * channels/misdn_config.c (added), channels/chan_misdn_config.c - (removed): rename file per crichter's request - -2006-08-17 21:57 +0000 [r40306] Christian Richter <christian.richter@beronet.com> - - * doc/README.misdn, channels/misdn/mISDN.patch (removed), - channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/fac.c (added), channels/misdn/Makefile, - channels/misdn/chan_misdn_config.h, channels/misdn/ie.c, - channels/misdn/fac.h (added), channels/misdn/portinfo.c - (removed), channels/misdn/isdn_lib_intern.h, - channels/chan_misdn_config.c, channels/misdn/isdn_msg_parser.c, - configs/misdn.conf.sample, channels/Makefile, - channels/misdn/isdn_lib.c: This rather small ;-) commit merges - the changes from my team branch 0.3.0 into t he 1.2 branch. These - changes include the new mISDN mqueue interface which makes it - possible to compile chan_misdn against the current cvs version of - mISDN/mISDNuser. These changes also contain various additions and - numerous bugfixes to chan_misdn . Each change is documented in - the commit logs in the team/crichter/0.3.0 branch. - -2006-08-17 16:36 +0000 [r40227] Russell Bryant <russell@digium.com> - - * channel.c: revert bogus change to attempt to fix bug 7506 which - actually causes half of the channels not to get "Newchannel" - events at all (issue #7745) - -2006-08-17 16:22 +0000 [r40223-40225] Joshua Colp <jcolp@digium.com> - - * funcs/func_cdr.c: Use the last CDR entry instead of the first CDR - entry for variable retrieving variables using the CDR dialplan - function. (issue #7689 reported by voipgate) - - * apps/app_macro.c: Make app_macro compile again - -2006-08-17 16:07 +0000 [r40220] Steve Murphy <murf@digium.com> - - * apps/app_macro.c: In app_macro, changed the previously changed - upper recursion depth limit to a variable, default of the - original val of 7. MACRO_RECURSION is a channel variable that - will override the limit, but until I can understand and fix why - this limit is neccessary, I am not advertising this variable in - the docs. This fix mirrors the changes made in r40200 in trunk. - -2006-08-16 18:57 +0000 [r40057] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_mgcp.c: don't allow AUEP responses to overflow the - stack during a string copy (reported by Mu Security) - -2006-08-15 22:49 +0000 [r39935] Russell Bryant <russell@digium.com> - - * res/res_agi.c: use pbx_builtin_getvar_helper() so that GET - VARIABLE can retrieve global variables (issue #7609) - -2006-08-15 22:13 +0000 [r39931] Steve Murphy <murf@digium.com> - - * apps/app_macro.c: This revision fixes bug 7731, the inability for - macros to be called more than one level deep in the 'h' - extension. It also pushes up the limit of recursion depth from 7 - to 20. - -2006-08-08 18:39 +0000 [r39379] Kevin P. Fleming <kpfleming@digium.com> - - * CREDITS: add explicit listing of anthm's contributions (issue - #7683) - -2006-08-08 17:04 +0000 [r39350] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: Increase the buffer size for the callid - (issue #7675, reported by pssatcs) - -2006-08-07 01:28 +0000 [r39081] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: Fix a crash reported to me by hads on IRC. - This crash would occur with the use of the - "distinctiveringaftercid" option. Also, on this user's system, - the crash would only occur when built without optimizations. This - is because the bug is that the code would write past the end of - an array that was allocated on the stack, and the structure of - the stack is different with or without optimizations enabled. - -2006-08-07 00:15 +0000 [r39056] Joshua Colp <jcolp@digium.com> - - * channel.c: Reset our stream and vstream pointers back to NULL so - that any generator that uses them (file based MOH) will not try - to close them again. (issue #7668 reported by jmls) - -2006-08-05 09:01 +0000 [r38903-38982] Russell Bryant <russell@digium.com> - - * channel.c: Always generate a Newstate event in ast_setstate() - instead of making it a Newchannel event if the state was - AST_STATE_DOWN. The Newchannel event will always be generated in - ast_request(), so this just causes a duplicated Newchannel event - in some cases. (issue #7506, repoted by capouch, fixed by me) - - * apps/app_queue.c: remove duplicate queue log entry when the - caller exits on a timeout (issue #7616, ppyy) - - * channels/chan_sip.c: don't advertise that this function can set a - SIP header when it can only do reads - - * apps/app_dial.c: make sure the priv-callerintros directory exists - before trying to create a file there (issue #7659, patch by hads, - with some modifications by me) - - * channels/chan_mgcp.c, channels/chan_vpb.c, channels/chan_phone.c, - channels/chan_misdn.c, channels/chan_zap.c, channels/chan_sip.c, - channels/chan_skinny.c, channels/chan_h323.c, - channels/chan_modem.c, channels/chan_iax2.c: Fix an issue that - would cause a NewCallerID manager event to be generated before - the channel's NewChannel event. This was due to a somewhat recent - change that included using ast_set_callerid() where it wasn't - before. This function should not be used in the channel driver - "new" functions. (issue #7654, fixed by me) Also, fix a couple - minor bugs in usecount handling. chan_iax2 could have increased - the usecount but then returned an error. The place where chan_sip - increased the usecount did not call ast_update_usecount() - - * channel.c: suppress a compiler warning about the usage of a - potentially uninitialized variable - -2006-08-03 19:54 +0000 [r38825] Joshua Colp <jcolp@digium.com> - - * res/res_musiconhold.c: Treat the file as invalid if we have no - valid formats for it (issue #7643 reported by KNK) - -2006-08-03 05:22 +0000 [r38761] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 7648 - Checking wrong count for - plurality on new messages for Dutch language - -2006-08-02 19:29 +0000 [r38686-38731] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: fix brain-damage I introduced when trying to - fix the CANCEL/BYE sending mechanism for pending INVITES accept - unknown 1xx responses as 183 responses (as RFC3261 mandates we - should do) - - * res/res_features.c, channel.c: ensure that the 'feature digit - timeout' value is taken into account when deciding how long the - bridge should run (this fixes a problem report where a digit - press that did not invoke a feature is never passed across the - bridge) - -2006-08-01 19:20 +0000 [r38654] Joshua Colp <jcolp@digium.com> - - * res/res_musiconhold.c: Close the stream when file based MOH stop. - This won't get rid of their position in the file but it will - cause the translation path to be setup again. (issue #7634 - reported by asimpson) - -2006-07-31 21:14 +0000 [r38611] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: don't reissue hangup requests for SIP - channels that have expired their RTP timeouts (one time is - enough) don't rescan the SIP private structure list too fast, it - can cause channels to not be able to hang up (issue #7495, and - probably others) use ast_softhangup_nolock() since we already - hold the channel's lock - -2006-07-31 17:09 +0000 [r38585] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: Add missing code to bring transferee channel - out of MOH/autoservice under certain circumstance (issue #7611 - reported by guillecabeza with minor mods by myself) - -2006-07-31 04:06 +0000 [r38546-38547] Russell Bryant <russell@digium.com> - - * frame.c: one more small tweak for thread-safety of TRACE_FRAMES - - * frame.c: Make the frame counting done with TRACE_FRAMES defined - thread-safe - -2006-07-29 23:18 +0000 [r38501] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: How many attempts does it take to make a SIP - URI parser that works well? I'm up to 5 personally. On to the - good stuff - parse the domain first, user second, and get rid of - port & options/params last. (issue #7616 reported by andrew) - -2006-07-28 18:49 +0000 [r38420] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Make a copy of the request URI in - check_user_full instead of modifying the one on the structure, - and also strip params properly from the user portion of the SIP - URI so as to preserve the domain (issue #7552 reported by dan42) - -2006-07-27 22:23 +0000 [r38347-38370] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_chanspy.c: use the enum that defines the option - arguments, so that the likelihood of mismatched option indexes is - reduced (which in this case was a bug, the volume argument was - not checked properly) - - * channel.c: do a better job avoiding translation path - teardown/setup when not needed - -2006-07-27 04:25 +0000 [r38328] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Fix crash when using the "regexten" option - with MALLOC_DEBUG enabled. This was not reported in the bug - tracker but the same bug has been demonstrated in other places in - the code. - -2006-07-27 02:43 +0000 [r38310] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: don't do useless translation destroy/build when the - channel is already in the correct format - -2006-07-27 01:58 +0000 [r38288] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: fix a crash when MALLOC_DEBUG is enabled and - the regexten is enabled. The crash would occur when the extension - got removed. (fixes issue #7484) - -2006-07-26 15:26 +0000 [r38234] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Put default callerid into contact when the - one specified is either NULL or has a zero string length. (issue - #7590 reported by key2) - -2006-07-25 19:43 +0000 [r38200] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: This resolves a deadlock that a tech support - customer was getting frequently when his users would answer call - waiting. If another thread is currently holding the zt_pvt lock - for the first channel, unlock both channels and let asterisk - retry the native bridge, just like what is done for the second - channel directly below these changes. - -2006-07-24 17:05 +0000 [r38167] Steve Murphy <murf@digium.com> - - * codecs/gsm/Makefile: This fixes a compile problem for s390 as - reported in bug 7253. Tested on both an s390 and non-s390 - machine. - -2006-07-19 17:10 +0000 [r37949] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: ensure that global 'maxauthreq' is reset to - zero during 'reload' - -2006-07-18 00:41 +0000 [r37828-37856] Russell Bryant <russell@digium.com> - - * frame.c: don't crash if the frame has no data, but has a src - - * frame.c: if asked to duplicate a frame that has no data, don't - set the frame's data pointer past the end of the allocatted - buffer for the new frame - -2006-07-17 22:36 +0000 [r37765-37808] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * formats/format_h263.c: Backport buffer increase to 1.2 - - * formats/format_h263.c: Overflow bad - -2006-07-15 23:29 +0000 [r37691] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * enum.c: Bug 7513 - ensure that each time we do a query, the - results are returned in the same logical order, so that when we - iterate over the list, we get all results, not some results - repeated, due to insufficient sorting. - -2006-07-14 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.10 released - -2006-07-14 13:31 +0000 [r37612] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_sms.c: Bug 7526 - previous commit broke app_sms - -2006-07-13 21:22 +0000 [r37571] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c: don't fail/abort if the message category - sound file cannot be played, just generate a warning message and - continue message playback - -2006-07-13 18:44 +0000 [r37546] Russell Bryant <russell@digium.com> - - * rtp.c: yeah, ummm... This frame pointer should not be static. - This situation only exists in 1.2 (pointed out by Constantine - Filin on the asterisk-dev mailing list) - -2006-07-13 16:44 +0000 [r37531] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: report address of peer trying to subscribe - to unknown hint - -2006-07-13 15:45 +0000 [r37458-37516] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * doc/README.enum: Bug 7532 - Typo in enum example - - * contrib/init.d/rc.mandrake.zaptel: Merge fixup for asterisk - startup script to zaptel startup script - -2006-07-12 15:53 +0000 [r37441-37442] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_voicemail.c: fix a weird case where a lock file could be - left (but would happen almost never) - - * app.c: fix a case where ast_lock_path() could leave a - randomly-named lock file hanging around make ast_unlock_path - actually report when unlocking fails - -2006-07-12 15:23 +0000 [r37439] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Add support to have maxauthreq as a global - option - -2006-07-12 13:54 +0000 [r37417-37419] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c, utils.c, res/res_agi.c, apps/app_zapras.c, - asterisk.c, channels/chan_modem.c, channels/chan_iax2.c: remove - some more bad examples of using printf - - * enum.c, pbx/pbx_config.c: get rid of some more printf's (although - most of these were ifdef-ed out) - -2006-07-12 03:55 +0000 [r37402] Matt O'Gorman <mogorman@digium.com> - - * app.c: GRRR no fprintf! - -2006-07-11 19:00 +0000 [r37378] Joshua Colp <jcolp@digium.com> - - * configs/iax.conf.sample, channels/chan_iax2.c: Add configuration - option for IAX2 users that will limit the amount of outstanding - AUTHREQs we are waiting for replies on. - -2006-07-10 21:01 +0000 [r37361] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: do masquerade-behind-proxy checking with better - control over locks - -2006-07-07 23:57 +0000 [r37307] Joshua Colp <jcolp@digium.com> - - * rtp.c: Change message regarding marker bit forcing when SSRC - changes to be shown only during debug so it doesn't overload high - capacity systems - -2006-07-06 21:41 +0000 [r37224] Matt O'Gorman <mogorman@digium.com> - - * channel.c: patch resolves issue with when to decide if its right - time to native bridge, feature redirect was not being checked. - patch from bug #7296 - -2006-07-06 20:38 +0000 [r37212] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_agent.c: Don't do weird things on a callback agent - that has attempted logoff while still on the phone. - -2006-07-06 15:48 +0000 [r37173] Joshua Colp <jcolp@digium.com> - - * channels/chan_sip.c: Instead of giving the scheduled item ID on a - peer expiration, give the time until they expire (issue #7455 - reported by slavon) - -2006-07-06 13:47 +0000 [r37143] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_db.c: Fix spelling/grammar (issue 7493) - -2006-07-05 15:31 +0000 [r36998] Joshua Colp <jcolp@digium.com> - - * channels/chan_oss.c: Spell extension correctly in documentation - for chan_oss dial (issue #7487 reported by flefoll) - -2006-07-04 14:45 +0000 [r36838-36911] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Tell clients based on old SIP standard that - we only support MD5 digest authentication... - - * channels/chan_sip.c: issue #7470 - Need larger buffer for - record-route headers... - -2006-07-03 05:12 +0000 [r36697-36751] Russell Bryant <russell@digium.com> - - * asterisk.c: fix a race condition that caused asterisk to log a - *ton* of warnings on mac osx about poll returning an error - because the polled file descriptor was bad. - - * channels/chan_mgcp.c, channels/chan_phone.c, - channels/chan_local.c, channels/chan_misdn.c, - channels/chan_sip.c, channels/chan_skinny.c, - channels/chan_agent.c, channels/chan_features.c, - channels/chan_h323.c, channels/chan_modem.c, - channels/chan_iax2.c: use ast_set_callerid to be more consistent - and to make sure that the "callerid" option in the conf files is - always handled the same way and sets ANI (issue #7285, gkloepfer) - - * dsp.c: fix the build with BUSYDETECT_TONEONLY defined (issue - #7414) - -2006-06-30 14:05 +0000 [r36290-36377] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_directory.c: Bug 7349 - Directory did not work correctly - when USE_ODBC_STORAGE was defined. - - * Makefile: Bug 7388 - compatibility changes for Solaris - -2006-06-29 07:19 +0000 [r36253-36254] Kevin P. Fleming <kpfleming@digium.com> - - * configs/queues.conf.sample: clarify documentation for - 'persistentmembers' setting - - * configs/sip.conf.sample: add documentation for peer-specific - 'outboundproxy' setting - -2006-06-28 14:12 +0000 [r36187] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Don't delete scheduled item twice in - sip_destroy (already fixed in svn trunk) - -2006-06-26 17:10 +0000 [r36078] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: ensure that two SIP channels that exist at - the same moment will not have the same channel names (issue - #7245, different fix) - -2006-06-26 15:27 +0000 [r36043] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 6997 maybe, but anyway - don't - retransmit responses to NON-invite requests. - -2006-06-25 15:10 +0000 [r35915] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Bug 7425 - Size of buffer is passed in by - len - -2006-06-23 11:30 +0000 [r35669] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: We should lock the queue before we go making - changes to member interface statuses. - -2006-06-21 19:25 +0000 [r35334] Joshua Colp <jcolp@digium.com> - - * configs/indications.conf.sample: Add Venezuelan indications - (issue #7402 reported by palillo) - -2006-06-20 15:05 +0000 [r35121] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * stdtime/private.h: Bug 7398 - Solaris puts its zoneinfo files in - a nonstandard place - -2006-06-20 10:27 +0000 [r35058] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6820 - Possible fix (already - implemented in trunk) - -2006-06-19 20:27 +0000 [r34911] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Call reset_user_pw upon changing the - password using externpass (issue #7395 reported by Ryan Cumming) - -2006-06-19 18:07 +0000 [r34875] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Issue 7357 - txt file left behind when - going to operator. Also, fix a possible file descriptor leak. - -2006-06-18 21:03 +0000 [r34627-34655] Russell Bryant <russell@digium.com> - - * pbx.c: don't set state to BUSY if the channel is already in the - UP state (issue #7376, backported from trunk) - - * configs/iax.conf.sample, channels/chan_iax2.c: don't store - multiple secrets delimited with semicolons for peers because this - is only valid for users. Instead, only keep the last specified - secret for a peer entry. Also, document how multiple secrets are - handled in the sample config. (Reported by PCadach on - #asterisk-bugs) - -2006-06-16 03:37 +0000 [r34400] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Zero out a declared structure so as to not - crash if it contains invalid data (reported by Qwell on - #asterisk-dev) - -2006-06-15 14:11 +0000 [r34306] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 7294 - patch by phsultan - Asterisk - sends Invite instead of BYE in some cases. - -2006-06-15 13:30 +0000 [r34274] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: don't use prefixed structure names for internal - structures don't use a plural structure name for a singular - object - -2006-06-15 12:40 +0000 [r34242] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: VoicemailMain exits on any key, when the - language is set to Italian, instead of properly handling the key - (issue 7353). - -2006-06-14 22:22 +0000 [r33841-34160] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: coding style cleanups on queue interface - handling code that was committed for the last release - - * channels/chan_iax2.c: use existing dial string parser for strings - supplied to iax2_devicestate, because they can be complete dial - strings, not just device names - - * include/asterisk/plc.h, jitterbuf.c, plc.c, apps/app_dumpchan.c, - apps/app_chanspy.c: clarify file headers that mention disclaimer - usage - - * file.c: don't output 'no format found' when we _did_ find the - format but couldn't open the desired file for some other reason - - * apps/app_mixmonitor.c: memory allocation optimizations - -2006-06-13 12:40 +0000 [r33753-33813] Russell Bryant <russell@digium.com> - - * pbx.c: remove duplicate mutex_unlock - - * apps/app_voicemail.c: fix various places where the code returns - without unlocking vmlock or destroying loaded configuration - - * apps/app_festival.c: add a missing close of an open fd, destroy - of open config, and removal of the calling channel from the - localusers list - - * asterisk.c: revert a change that caused more problems than it - fixed and fix the real problem in this code. fds was declared as - an array of zero size which caused some weird problems, some of - which would only be seen when compiling without optimizations. - (fixes issues #7071, #7326, and #7305) - -2006-06-12 21:34 +0000 [r33724] Joshua Colp <jcolp@digium.com> - - * include/asterisk/chanspy.h, apps/app_mixmonitor.c, channel.c: - Greatly simply the mixmonitor thread, and move channel reference - directly to spy structure so that the core can modify it. - -2006-06-12 20:40 +0000 [r33693] Russell Bryant <russell@digium.com> - - * res/res_agi.c: fix a place where a frame would be free'd twice - -2006-06-12 16:03 +0000 [r33638] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_local.c: only allow chan_local to masquerade the - outbound channel onto its owner, instead of the other way around - (this will ensure that group variables on the outbound channel are - preserved) - -2006-06-12 15:27 +0000 [r33615] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_agi.c: Move set priority up, because at this point in the - code, stdout is no longer the console. If we're unable to set - priority, the error goes to Asterisk as if it were an AGI command - (issue 7335). - -2006-06-11 21:21 +0000 [r33449-33548] Russell Bryant <russell@digium.com> - - * pbx.c: fix another place where a frame does not get free'd - - * apps/app_meetme.c: fix up five little places where frames would - not be free'd and remove an unnecessary mutex_unlock where there - is no way for it to be locked at that time - - * apps/app_ices.c: fix a place that would leak a frame (all of - these fixes are in applications that call ast_read() on a channel - but have code paths in them that would not free the frame) - - * apps/app_festival.c: fix a couple places that would leak a frame - - * apps/app_alarmreceiver.c: fix two places that would cause a frame - to be leaked - - * apps/app_url.c: fix a case where an HTML frame would be leaked - - * apps/app_test.c: Free frames read from the channel when measuring - noise. This resulted in about 9 or 10 seconds of leaked frames in - both the TestClient and TestServer applications - - * apps/app_zapbarge.c, apps/app_zapscan.c: backport a couple of - frame leak fixes from the trunk (revisions 33446, 33447) - -2006-06-09 18:52 +0000 [r33264-33300] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Allow the format outputted by meetme list to - be used for meetme commands (like kick) (issue #7322 reported by - darkskiez) - - * channels/chan_iax2.c: Remove an unneeded double lock (issue #7310 - reported by arkadia) - - * apps/app_dial.c: Handle hangup during recording of screened name - (issue #7304 reported by kulldominique) - - * apps/app_meetme.c: Add missing newlines (issue #7323 reported by - darkskiez) - -2006-06-09 15:53 +0000 [r33235] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Do not require a context on a domain= - setting - -2006-06-08 16:57 +0000 [r33036] Kevin P. Fleming <kpfleming@digium.com> - - * frame.c: handle out-of-memory conditions properly in - ast_frisolate() (reported by Slav Kenov on asterisk-dev mailing - list) - -2006-06-07 17:53 +0000 [r32818] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: fix some broken code with - BRIDGE_OPTIMIZATION defined (issue #7292) - -2006-06-06 16:55 +0000 [r32605] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 7287 - A too short voicemail with - ODBC_STORAGE will cause the first voicemail to be deleted - erroneously - -2006-06-06 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.9.1 released - -2006-06-06 16:02 +0000 [r32582] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * callerid.c: Bug 7268 - Callerid leaks memory on error - -2006-06-06 15:48 +0000 [r32566] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: clean up yesterday's security fix to not - cause breakage when video mini frames are received - -2006-06-03 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.9 released - -2006-06-05 19:53 +0000 [r32373] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: ensure that the received number of bytes is - included in all IAX2 incoming frame analysis checks (fixes a - known vulnerability) - -2006-06-04 03:43 +0000 [r31921] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: return bridge exit logic to what it was before - i broke it :-( - -2006-06-03 17:02 +0000 [r31775] Russell Bryant <russell@digium.com> - - * res/res_musiconhold.c: when using moh files mode, don't look for - a file past the number of files that have been loaded, or worse, - past the size of the files array - -2006-06-01 21:46 +0000 [r31321-31555] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_musiconhold.c: remove pointless forcing of the channel - into SLINEAR mode; the write format will be set later based on - the file that is chosen to be played to the channel - - * include/asterisk/channel.h, channel.c: handle Zap transfers - behind chan_agent properly so the agent doesn't get stuck waiting - for the call to hang up - - * configs/sip.conf.sample: remove a sample entry that never should - have been added (code to support it was not merged) - -2006-05-31 23:50 +0000 [r31194] Russell Bryant <russell@digium.com> - - * res/res_agi.c: if the connection to a FastAGI server fails - because of a timeout, log a more informative log message - -2006-05-31 22:26 +0000 [r31161] Kevin P. Fleming <kpfleming@digium.com> - - * rtp.c: silence a warning message that is not a warning - -2006-05-31 20:26 +0000 [r31127] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: fix misplaced manager event (issue #6866, - flefoll) - -2006-05-30 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.8 released - -2006-05-30 14:55 +0000 [r30770] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: Fix infinite loop scenario and add some sanity - checking to prevent segfault on a NULL parameter coming in (which - probably shouldn't happen, but just to be safe...) - -2006-05-26 17:09 +0000 [r30424-30546] BJ Weschke <bweschke@btwtech.com> - - * apps/app_queue.c: A new way to try and deal with deadlocks that - occur in app_queue at present. Using this approach, we only - manipulate the main queue mutexes when we get a dev state change - on a device that is actually a member of a queue. Backported from - /trunk for the "bug fix". - -2006-05-25 20:03 +0000 [r30373] Joshua Colp <jcolp@digium.com> - - * apps/app_meetme.c: Don't play the enter sound twice when a person - joins a conference after the leader has joined it. (issue #6138 - reported by shanermn) - -2006-05-25 17:39 +0000 [r30293-30296] Kevin P. Fleming <kpfleming@digium.com> - - * codecs/gsm/Makefile: don't try to use -march=s390 when building - on S/390 systems (reported via asterisk-users mailing list) - - * channels/chan_sip.c: allow SIPCHANINFO(peername) to work for - calls from users as well (issue #7215) - -2006-05-25 15:27 +0000 [r30239] Joshua Colp <jcolp@digium.com> - - * configs/extensions.conf.sample: Get rid of an incorrect SIP dial - string in the sample extensions.conf - I even tried variations... - no go (issue #7222 reported by arkadia) - -2006-05-24 21:24 +0000 [r30069-30098] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: oops... make sure to stop processing a - request once we have sent an authentication challenge (issue - #7220) - - * channels/chan_sip.c: don't send CANCEL on a pending INVITE if we - haven't received a provisional response yet... mark it pending - until the first response is received (issue #7079) - -2006-05-24 19:55 +0000 [r30037] Matt O'Gorman <mogorman@digium.com> - - * apps/app_meetme.c: app_meetme used the ast_max_exten instead of - path_max solves bug 6822 - -2006-05-24 19:44 +0000 [r30033-30035] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Merge branch for bug 6264 (Privacy option 2 - returns dial-status ANSWER / option_priority_jumping not - respected) (reported by jkoopmann and branch by murf) - - * logger.c: Fix deadlock caused by a race condition in the logger - when reloading (issue #7195 reported and fixed by softins) - -2006-05-24 16:59 +0000 [r29904-29973] Kevin P. Fleming <kpfleming@digium.com> - - * res/res_agi.c: support video recording via AGI 'RECORD FILE' - command (issue #7068) - - * apps/app_queue.c: fix various bugs related to exiting from queue - via keypress and moh handling (issue #6776, different fix) - - * channels/chan_zap.c: respect 'usecallingpres' in zapata.conf even - if CLID has not been set for the channel (issue #7123) - - * channels/chan_sip.c, configs/sip.conf.sample: add an option to - allow the admin to 'hide' SIP user/peer names from systems trying - to 'fish' names - -2006-05-23 21:44 +0000 [r29849] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: fix the sourceaddress option (issue #7213, - alphaque) - -2006-05-23 18:16 +0000 [r29764] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: simplify/fix lock retry, and fix comment - -2006-05-23 17:17 +0000 [r29733] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_sip.c: Sanity check code for an extended failure in - trying to obtain a channel lock that may have been obtained - elsewhere. Prevents the monitor thread of the SIP module from - going into an infinite loop, effectively, breaking SIP until you - restart Asterisk or the mutex is unlocked, whichever comes first. - -2006-05-23 17:15 +0000 [r29732] Kevin P. Fleming <kpfleming@digium.com> - - * dnsmgr.c, res/res_features.c, include/asterisk/linkedlists.h, - include/asterisk/lock.h, apps/app_sql_postgres.c, pbx.c: backport - some mutex initialization and linked list handling fixes from - trunk - -2006-05-23 15:58 +0000 [r29696] BJ Weschke <bweschke@btwtech.com> - - * res/res_features.c: Fix a potential leak and correct (hopefully) - a segfault under certain conditions. #6784 (vovan and perry - testing) - -2006-05-22 21:27 +0000 [r29464-29555] Joshua Colp <jcolp@digium.com> - - * apps/app_waitforsilence.c: Increase the silence threshold to 128 - to "fix" it, so I'm told. (issue #6595 reported by davetroy fixed - by casper) - - * res/res_features.c: Use the correct language when playing the - transfer sound (issue #7109 reported by casper) - - * channels/chan_local.c: Preserve presentation bit when going - through chan_local (issue #7002 reported by acunningham) - -2006-05-22 14:59 +0000 [r29394-29398] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_meetme.c: Bug 7194 - spelling fix - - * pbx.c: Bug 7196 - month range did not work - -2006-05-21 15:16 +0000 [r29196] BJ Weschke <bweschke@btwtech.com> - - * res/res_features.c: When an application that is executed via - applicationmap and exits non-zero, make sure that we pass through - the correct return value from the application to make sure a - segfault doesn't occur by a bridge trying to continue when it - should not. Also, when executing applications via applicationmap, - make sure that the application is executed against the channel - whose DTMF caused it to be fired off in the first place. (part - 1/2 of #7090 - this is the only fix that will be applied to both - 1.2 and /trunk) acunningham and blitzrage on testing... - -2006-05-20 19:50 +0000 [r29052] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: fix the possibility of writing one byte past - the end of a buffer. (issue #7189, Mithraen) - -2006-05-20 02:35 +0000 [r28968] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: don't allow queue member devices to ring longer - than the total queue timeout (issue #6423, reported and patched - by bcnit) - -2006-05-20 02:31 +0000 [r28966] Russell Bryant <russell@digium.com> - - * apps/app_sms.c: fix a case where code made assumptions about how - memory for variables is allocatted on the stack - this patch is - slightly different than the one that went in for the trunk - -2006-05-20 00:55 +0000 [r28794-28896] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: don't try to predict where the compiler - will place things on the stack... put them in the right place - explicitly (issues #7029 and #7100, maybe others) - - * channels/chan_sip.c: use the specified 'subscribecontext' for a - peer rather than the context found via the target domain (domain - contexts are for calls, not for subscriptions) (issue #7122, - reported by raarts) - -2006-05-19 19:18 +0000 [r28754-28790] Russell Bryant <russell@digium.com> - - * utils/smsq.c: fix the build of smsq with -Werror. I learned - something new about format strings from this patch! (issue #7141, - Mithraen) - - * asterisk.c: This explicit poll is only needed on mac. In fact, it - breaks some systems such as some versions of Fedora, causing - 'asterisk -rx' to never exit. This has been tested on systems - showing the asterisk -rx problem, as well as other unaffected - versions of linux, mac osx 10.4, and FreeBSD 6. (issue #7071) - -2006-05-19 17:04 +0000 [r28627-28698] Joshua Colp <jcolp@digium.com> - - * channels/chan_zap.c: Make the minidle option actually exist as - documented (issue #7159 reported by imran) - - * apps/app_voicemail.c: When forwarding messages use the context - that the active voicemail user was found in. (issue #7010) - - * enum.c: Backport of fix for issue #6654 that was fixed in trunk - but not here - - * apps/app_queue.c: Treat paused queue members as unreachable - (issue #7127 reported by peterh) - -2006-05-18 20:43 +0000 [r28335-28384] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: fix up a few more places to find the SDP - properly (fallout from fix for #7124) - - * channels/chan_sip.c: handle incoming multipart/mixed message - bodies in SIP and find the SDP, if present (issue #7124 reported - and patched by eborgstrom, but very different fix) - - * enum.c: use unsigned counters for handling answer/IE lengths - while processing DNS results (issue #7174) - - * channels/chan_sip.c: support 'inactive' tag for SDP media streams - (simple fix, proper fix will appear in 1.4 release) (issue #7130) - -2006-05-18 17:27 +0000 [r28257] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_hasnewvoicemail.c: Bug 7167 - HasNewVoicemail and - VMCOUNT() didn't work when USE_ODBC_STORAGE was defined - -2006-05-18 16:31 +0000 [r28169-28212] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Return -1 on error in ODBC messagecount and - 0 on success (issue #7133 reported by cfieldmtm) - - * apps/app_voicemail.c: Fix endless looping message by checking - value of res before doing retries stuff. (issue #7140 reported by - tanischen) - -2006-05-18 12:13 +0000 [r28125] Olle Johansson <oej@edvina.net> - - * apps/app_meetme.c: Video in meetme? Hmmm. Removed until we do - have some code for it. - -2006-05-17 22:34 +0000 [r27973] Joshua Colp <jcolp@digium.com> - - * channels/chan_iax2.c: Fix codec priority stuff during - authentication (issue #6194 reported by jkoopmann) - -2006-05-17 19:27 +0000 [r27927] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7176 - Crash in expire_register (We - need to find out what's causing peer to be undefined, so this is - just a bandaid, not a real fix) - -2006-05-17 17:07 +0000 [r27767-27847] Joshua Colp <jcolp@digium.com> - - * apps/app_voicemail.c: Priority jumping not working on VoiceMail - app with new syntax (issue #7164 reported and fixed by - alvaro_palma_aste) - - * apps/app_osplookup.c: OSPNext does not handle success/failure - correctly (issue #7147 reported and fixed by eborgstrom) - -2006-05-17 09:21 +0000 [r27723] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: chan_sip did not use the TRANSFER_CONTEXT - for transfers, like res_features. Now fixed. - -2006-05-17 02:19 +0000 [r27636] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 7125 - Fix race condition between - resequencing and leaving a message - -2006-05-16 23:31 +0000 [r27594] Joshua Colp <jcolp@digium.com> - - * apps/app_dial.c: Inherit channel variables during call forwards - when going through chan_local (issue #7095 reported by raarts) - -2006-05-16 20:05 +0000 [r27468] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: don't leak frames when deferring DTMF or dropping - duplicate ANSWER frames (issue #7041, slightly different fix, - reported/patched by clausf) - -2006-05-13 04:08 +0000 [r27093] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 7134 - File descriptor leak with ODBC - storage of voicemail - -2006-05-11 23:02 +0000 [r27051] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_logic.c: Bug 7086 - pbx_checkcondition substitution, - so that arbitrary strings are true (for regex) - -2006-05-11 09:05 +0000 [r26760-26773] Kevin P. Fleming <kpfleming@digium.com> - - * rtp.c: backport fix from trunk for bug #6934, ensuring that RTP - mark bit is changed when SSRC changes - - * channels/chan_sip.c: ensure that we send a response to REGISTER - requests that are successfully authenticated but contain invalid - Contact URIs - -2006-05-09 14:18 +0000 [r26050-26090] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_sip.c, doc/README.variables: Add the appropriate - jumping behavior that is the standard for 1.2.X to SIPGetHeader - that is now deprecated in /trunk. #7111 (blitzrage!!!) - - * apps/app_voicemail.c: Correct memory leak in find_user_realtime - #7118 (fnordian) - -2006-05-08 15:09 +0000 [r25608] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 7103 - mikma - The header is named - "Require" - Don't reply to ACK (Not using patch against trunk) - -2006-05-08 14:12 +0000 [r25518-25563] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_agent.c: Don't show agents as available when they - are in wrap-up time. #6726 (ZX81) - - * apps/app_queue.c: Make QueueStatusComplete event thread safe by - wrapping it inside the queue lock clause already there. #7013 - (bziherl reporting) - - * apps/app_queue.c: Don't recheck valid_exit() after getting the - result from say_position (which already checks it). Should - prevent another loop if the caller hits digits during the - position announcement. #6776 (tgj reporting) - -2006-05-08 11:16 +0000 [r25442] Joshua Colp <jcolp@digium.com> - - * res/res_features.c: Incorrect log statement when playing transfer - sounds (issue #7008 reported and fixed by nathan) - -2006-05-07 13:38 +0000 [r25288-25322] BJ Weschke <bweschke@btwtech.com> - - * apps/app_meetme.c: Fix playback behavior to exit correctly when - we receive a hangup during playback of the invalid pin message. - #7091 (AntD reporting) - - * asterisk.c: Reset the value of ast_mainpid if we fork so future - remote unix connections display the correct PID. #7098 (tzafrir - reporting) - -2006-05-06 02:32 +0000 [r25015-25165] Russell Bryant <russell@digium.com> - - * frame.c: fix a problem where the frame's data pointer is - overwritten by the newly allocated data buffer before the data - can be copied from it. This is in the ast_frisolate() function - which is rarely used. (issue #6732, stefankroon) - - * channels/chan_zap.c: ensure that the appropriate manager events - are sent in all of the places where alarms are detected or - cleared (issue #6866, flefoll) - - * channels/chan_h323.c: update chan_h323 to reflect the new - prototype for rtp_set_peer (issue #6560, casper) This was fixed a - couple months ago in the trunk, but never in 1.2. - -2006-05-05 20:44 +0000 [r25014] BJ Weschke <bweschke@btwtech.com> - - * apps/app_voicemail.c, include/asterisk/app.h, app.c: Voicemail - fixes along with an API change approved by russellb to fix the - bug(s). (jcollie and supczinskib) #7064 - -2006-05-05 17:39 +0000 [r24837-24911] Russell Bryant <russell@digium.com> - - * apps/app_while.c, apps/app_macro.c: use pbx_checkcondition() - instead of ast_true() to evaluate the condition for MacroIf and - WhileIf (issue #7086) - -2006-05-04 16:27 +0000 [r24706] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_queue.c: Bug 7023 - reload should not unpause members - -2006-05-04 11:17 +0000 [r24567-24669] BJ Weschke <bweschke@btwtech.com> - - * apps/app_verbose.c: Make sure that only the "|" is a recognized - delimiter for Verbose(), as the app documentation already - specifies. #7080 (alessiof reporting) - - * apps/app_dial.c: Correct application documentation to make users - aware that certain options cannot be used in conjunction with - others. #6666 (chotaire) - -2006-05-03 18:31 +0000 [r24496] Russell Bryant <russell@digium.com> - - * redhat/asterisk.spec: fix up "make rpm" - don't reference the - gzipped man page, because we don't store them compressed anymore - - add some files that currently were not listed (issue #6837) - -2006-05-03 12:39 +0000 [r24381] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #7074 - Problem with long contact - lines - -2006-05-02 19:39 +0000 [r24295] BJ Weschke <bweschke@btwtech.com> - - * file.c: Make certain ast_stopstream() sets the channel's stream - members to NULL after closing them. #7067 (jcomellas) - -2006-05-02 02:12 +0000 [r24019-24097] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_privacy.c: Prompt does not request '#' to end input, so - the application should not require it - - * apps/app_nbscat.c, apps/app_festival.c, apps/app_mp3.c, - apps/app_zapras.c, asterisk.c, apps/app_externalivr.c, - apps/app_ices.c, res/res_musiconhold.c, - include/asterisk/options.h: Bug 6864 - drop realtime priority on - ALL external processes - -2006-05-01 19:34 +0000 [r23985-23988] BJ Weschke <bweschke@btwtech.com> - - * apps/app_voicemail.c: Make sure that when someone 0's out while - recording a msg and then chooses to DELETE the recorded file, the - .txt file isn't left around by itself to cause problems later. - #7061 (dimitripietro reporting, blitzrage confirmed) - -2006-05-01 15:12 +0000 [r23951] Russell Bryant <russell@digium.com> - - * pbx.c: add missing locking of the dialplan functions list in the - "show functions" CLI command - -2006-05-01 10:45 +0000 [r23305-23899] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_skel.c: fix this to actually compile so people can learn - from it - - * cdr/cdr_sqlite.c: eliminate compiler warning - - * channels/chan_iax2.c: remove a pointless comparison, since the - buffer is smaller than the length being checked for - - * Makefile, editline/configure, cdr/Makefile, channels/Makefile, - db1-ast/Makefile: allow top-level OPTIMIZE setting to affect - builds in these subdirectories too - - * Makefile: let the compiler determine whether hardware or software - floating point should be used (like we do in the editline - subdirectory) - - * Makefile, apps/Makefile: remove extraneous -m64 flag that is not - needed remove old 'look' target which is no longer needed (these - are coming from Debian patches <G>) - - * editline/makelist: ensure that the script output is correctly - generated when the system's character set does not use the - English lowercase/uppercase character groups - - * Makefile: do installation in subdirs as a separate target (so - external modules can use the Makefile more easily) generate final - messages -after- running any post-install script that may be - present - -2006-04-28 16:40 +0000 [r23176] Russell Bryant <russell@digium.com> - - * configs/zapata.conf.sample, configs/mgcp.conf.sample, - configs/sip.conf.sample: note that group assignments must be from - 0 to 63 (issue #7048) - -2006-04-27 19:11 +0000 [r22954] Joshua Colp <jcolp@digium.com> - - * apps/app_queue.c: Queue(somequeue,,,,) -> interpreted as - Queue(somequeue,,,,0) (issue #7044 reported nathan fixed by - jsmith - sort of) - -2006-04-27 16:12 +0000 [r22866] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Fix buglet in channel reassignment on - SETUP_ACK - -2006-04-26 19:18 +0000 [r22596] Matt O'Gorman <mogorman@digium.com> - - * apps/app_voicemail.c: do not allow for users to forward voicemail - to themselves, patch from 7001 - -2006-04-21 22:39 +0000 [r22112-22113] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channel.c: Bug 7004 - release all threads waiting on a condition - prior to freeing it - -2006-04-19 21:10 +0000 [r21638] Kevin P. Fleming <kpfleming@digium.com> - - * contrib/scripts/safe_asterisk.8, contrib/scripts/safe_asterisk: - support system-specific scripts in safe_asterisk, before starting - Asterisk proper - -2006-04-19 18:43 +0000 [r21597] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * cdr/cdr_odbc.c: Bug 6553 - plug memory leaks when ODBC connection - is down - -2006-04-18 23:31 +0000 [r21237] Kevin P. Fleming <kpfleming@digium.com> - - * pbx.c: properly handle brace-wrapped strings in variable/function - references in the dialplan - -2006-04-18 06:26 +0000 [r20966-21037] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_random.c: Bug 6984 - off by one error in Random() - - * res/res_musiconhold.c: Bug 6544 - when we remove a music class, - the thread servicing it should die - -2006-04-14 17:21 +0000 [r20034-20037] Kevin P. Fleming <kpfleming@digium.com> - - * sounds.txt: uncomment files that actually do exist (oops) - - * sounds.txt: update text to match actual prompts being distributed - (thanks to Kinsey in the support department for reviewing all the - prompts!) - -2006-04-13 20:37 +0000 [r19891] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 6947 - Allow vm broadcasts to more than - 256 characters worth of mailboxes - -2006-04-13 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.7.1 released - -2006-04-13 17:40 +0000 [r19812] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_page.c: oops... let's not set a variable and then - immediately overwrite it while assuming its old value will - magically return - -2006-04-13 15:56 +0000 [r19768] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx.c: Bug 6957 - variable names beginning with CALLERID weren't - substituted correctly - -2006-04-12 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.7 released - -2006-04-11 22:39 +0000 [r19394-19397] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_dial.c: Bug 6490 - telco intercept should report - NOANSWER instead of CHANUNAVAIL - - * apps/app_voicemail.c: Bug 6061 - Fix ODBC storage of VM on PGSQL - and MSSQL - -2006-04-11 21:58 +0000 [r19353] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: don't create a 'voicemail' symlink in the sounds - directory; app_voicemail has not needed it since January of 2005 - (issue #6613) - -2006-04-11 21:55 +0000 [r19351] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * asterisk.c: Bug 6097 - possible descriptor leak - -2006-04-11 21:50 +0000 [r19345-19348] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_page.c: don't call the originating device as part of the - Page() operation (issue #6932) - - * channel.c: simplify spy queue flushing logic, and always force a - flush when one side gets full, even if the other side is not - empty (issue #6457) - - * pbx/pbx_config.c: don't destroy the entire dialplan during - 'reload', just atomically replace it like 'extensions reload' - does (issue #6047) - -2006-04-11 20:46 +0000 [r19303] Joshua Colp <joshnet@nbnet.nb.ca> - - * include/asterisk/linkedlists.h: Minor linked lists bug fix. When - you're dealing with swapping entries around a lot it can cause a - seg fault. - -2006-04-11 20:11 +0000 [r19301] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_dial.c: handle call time limit properly when warning is - requested _after_ call would hae already ended (issue #6356) - -2006-04-11 01:05 +0000 [r18866-19008] BJ Weschke <bweschke@btwtech.com> - - * apps/app_voicemail.c, app.c: When using the silence detector in - ast_play_and_record() and ast_play_and_prepend(), the truncation - code never gets called to remove the detected silence, because - the value of res is zero when control gets to that point. #6903 - w/some mods (softins) - - * res/res_features.c: Don't say that we can pass an 'exten' - argument in the documentation of Park() when we really cannot. - #6902 (opsys) - -2006-04-08 19:20 +0000 [r18436-18494] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 6914 - .txt file fails to rename on - operator out - - * formats/format_jpeg.c: Bug 6913 - fix for possible buffer - overflow - -2006-04-07 14:16 +0000 [r18250-18260] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: - Fix cause codes - Add cause code for - incompatible formats - - * channels/chan_sip.c: - Fix possible minor memory leak in chan_sip - - Return proper cause code on memory allocation error - -2006-04-06 22:15 +0000 [r18087-18089] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_meetme.c: fix typo - - * apps/app_meetme.c: small fix... don't try to check conference - details if it couldn't be created or found - - * apps/app_meetme.c: don't try to support 'i' or 'r' options if - chan_zap is not loaded, and warn the user when they attempt to - use them (issue #6675) update application help text to more - clearly define when Zaptel and chan_zap are required - -2006-04-06 17:24 +0000 [r17945] Russell Bryant <russell@digium.com> - - * apps/app_alarmreceiver.c: move continue out of block that checks - verbose level (issue #6880) - -2006-04-06 17:00 +0000 [r17702-17905] Joshua Colp <joshnet@nbnet.nb.ca> - - * pbx.c: Unlock channel on failure so that ast_mutex_destroy - doesn't throw a fit (issue #6647 reported by casper) - -2006-04-05 06:50 +0000 [r17335-17489] Olle Johansson <oej@edvina.net> - - * CREDITS, enum.c: Issue #6654: Enum crash on ADDRESS record, - possibly bad record, but still a crash - - * channels/chan_zap.c: Issue #6878 - Unhide DNDstate manager events - (thanks casper) - - * apps/app_queue.c: Issue #6882 - move "res=-1" out of verbosity - block, minor code cleanups (casper) - -2006-04-04 15:24 +0000 [r17283] Matt O'Gorman <mogorman@digium.com> - - * apps/app_senddtmf.c: Adds documentation to show what the w flag. - Patch from Ian Kinner at Digium. - -2006-04-03 20:38 +0000 [r17074-17150] Olle Johansson <oej@edvina.net> - - * configs/features.conf.sample: Issue 6870 - document that parking - lots need to be numeric - - * channels/chan_sip.c: Issue #6848 take two - Use the tag provided - by the SUBSCRIBE request when sending NOTIFY - - * channels/chan_sip.c: Ugly patch to avoid hangup causes in - non-final responses - -2006-03-31 19:11 +0000 [r16744-16771] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: move a NULL check to before the first time - the pointer is dereferenced (issue #6832) - - * channels/chan_iax2.c: fix the situation where bindport is - specified but bindaddr is not (issue #6616) - -2006-03-31 18:24 +0000 [r16742] Kevin P. Fleming <kpfleming@digium.com> - - * pbx.c: ensure that hint watchers (subscribers) cannot be added or - removed while the dialplan is being modified - -2006-03-30 22:56 +0000 [r16579-16581] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Bug 6853 - Manager fixes: 1) extra ActionID, - 2) missing colon - - * asterisk.c: Bug 6849 - trivial typo fix - -2006-03-30 21:44 +0000 [r16534-16559] Joshua Colp <joshnet@nbnet.nb.ca> - - * codecs/gsm/Makefile: Add another check for 64-bit goodness (issue - #6850 reported by evilbunny) - - * res/res_musiconhold.c: Do not exceed the array size for maximum - allowed moh files. (issue #6842) - -2006-03-30 01:34 +0000 [r16303-16346] Olle Johansson <oej@edvina.net> - - * res/res_features.c: Set initial value on adsipark - - * apps/app_groupcount.c: Typo fix. - - * configs/extensions.conf.sample: Typo (Issue 6839, casper) - -2006-03-29 19:11 +0000 [r16082-16192] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/pbx.h, apps/app_stack.c, pbx.c: Bug 6830 - Let - GosubIf work with the same conditions as a GotoIf (change in API - approved by Russell) - - * pbx.c: Bug 6835 - Updates to GotoIf help text - -2006-03-29 04:15 +0000 [r16008] Russell Bryant <russell@digium.com> - - * strcompat.c: tell unsetenv for solaris to return the result of - the setenv call - -2006-03-29 00:58 +0000 [r15898] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6823 - Portability issue with the - registration port number patch from yesterday. Be compatible with - more systems than OS/X :-) Thanks Rizzo for the advice. - -2006-03-29 00:32 +0000 [r15896] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/linkedlists.h: ensure that list traversal loops - which skip entries properly update the 'previous entry' pointer - so when entries _are_ removed the list does not get damaged - -2006-03-28 20:22 +0000 [r15703-15743] Russell Bryant <russell@digium.com> - - * agi/Makefile, strcompat.c, astmm.c: backport astmm + sparc fixes - from the trunk - - * channels/chan_iax2.c: fix Bus Error on sparc (issue #6354) - -2006-03-28 19:07 +0000 [r15699] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Fix breakage of NAT support for peers with - qualify=yes. Thanks Damin for access to your system, sorry folks. - -2006-03-28 18:09 +0000 [r15658] Russell Bryant <russell@digium.com> - - * pbx/pbx_ael.c: fix the order in which for loops are expanded - (issue #6810) - -2006-03-28 17:48 +0000 [r15615] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/init.d/rc.redhat.asterisk: Bug 6815 - Adding quotes to - make bash happy - -2006-03-27 23:45 +0000 [r15366-15381] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6736 - Use flags for OPTION messages. - Thanks Casper! - - * channels/chan_sip.c: Issue #6597 - sip show registry shows - incorrect port - - * channels/chan_sip.c: Issue #6409 - Use "s" extension when there's - no username in the URI - -2006-03-26 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.6 released - -2006-03-25 05:07 +0000 [r14821-14868] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/init.d/rc.redhat.asterisk: Bug 6601 - More configuration - abilities for the RH init script - - * apps/app_voicemail.c: Fix incorrect size of zeroing (left over - from when maxmsg was hardcoded at 100) - - * apps/app_voicemail.c: Bug 6783 - When context is specified, - voicemail should look for mailboxes in that context - -2006-03-24 14:48 +0000 [r14704] Russell Bryant <russell@digium.com> - - * image.c: use the correct variable in an error message (issue - #6791) - -2006-03-24 04:53 +0000 [r14610-14659] BJ Weschke <bweschke@btwtech.com> - - * apps/app_voicemail.c: Fix a typo in the app description - - * include/asterisk/sched.h: Doxygen comment typo corrections - -2006-03-23 21:51 +0000 [r14523] Joshua Colp <joshnet@nbnet.nb.ca> - - * res/res_features.c: Issue #6764 - Return BUSY signal when other - party is busy at Attended Transfer (Reported by mnachev) - -2006-03-23 21:44 +0000 [r14522] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Fix SETUP_ACK handling so that we change - channels if so requested - -2006-03-23 20:43 +0000 [r14467] BJ Weschke <bweschke@btwtech.com> - - * apps/app_meetme.c: Bug #5884 - fix a possible race state in - app_meetme when a channel has gone away and we are reading - continuously for more frames. (mneuhauser) - -2006-03-23 20:13 +0000 [r14462] Russell Bryant <russell@digium.com> - - * apps/app_readfile.c: don't crash when asked to read from a file - that doesn't exist (issue #6786) - -2006-03-22 22:18 +0000 [r14191-14276] Joshua Colp <joshnet@nbnet.nb.ca> - - * apps/app_voicemail.c: Fix a minor code issue - - * apps/app_voicemail.c: Issue #6781 - Verbose levels not enforced - in app_voicemail (Reported by flobi) - - * include/asterisk/cdr.h, cdr.c: Issue #5918 - Disposition showing - FAILED even though call is answered successfully (Reported by - tracinet) - - * pbx.c: Issue #6780 - ast_pbx_outgoing_cdr_failed description fix. - (Reported and fixed by casper) - -2006-03-22 09:10 +0000 [r14140] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6766 - fix ;user=phone functionality. - (Reported by alein, fix by russell - thanks!) - -2006-03-21 18:59 +0000 [r13814-13964] Russell Bryant <russell@digium.com> - - * configs/features.conf.sample: add a note explaining how to set - the DYNAMIC_FEATURES variable to allow the use of custom features - (issue #6747) - - * res/res_features.c: fix crash when using the ParkAndAnnounce - application. When using this application, there will be no peer - channel to play the parking announcement to. (issue #6756) - - * funcs/func_strings.c: fix REGEX on strings that contain quotes - (issue #6678) - - * sounds.txt: fix spelling of whiskey - - * apps/app_meetme.c: don't add conference participant if the user - hangs up while recording their name (issue #6661) - - * sample.call: re-add the Account parameter to the sample call file - since it's not really deprecated since the CDR function is no - longer built in - -2006-03-21 06:24 +0000 [r13707-13748] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 6714 - Workaround to avoid retrieving - incomplete voicemail message - - * editline/term.c: Do away with some warnings and fix some - indentation - -2006-03-20 17:36 +0000 [r13634] Olle Johansson <oej@edvina.net> - - * channels/chan_iax2.c: Do not overwrite ANI if it's set by IE - (sendani=yes in the peer) - -2006-03-19 09:59 +0000 [r13550] Russell Bryant <russell@digium.com> - - * apps/app_dial.c: revert the change made in revision 12927 in - favor of keeping the original behavior of the option. The - documentation has now been updated to reflect the actual - behavior. (issue #6523) - -2006-03-19 09:25 +0000 [r13547] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Reset global_rtautoclear at sip reload - -2006-03-16 20:05 +0000 [r13279] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * ast_expr2.y, ast_expr2.c: Bug 6737 - Fix compile warning on OS X - -2006-03-16 17:58 +0000 [r13239] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Issue #6690 - clarify progressinband - default setting - -2006-03-16 17:42 +0000 [r13237] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: always use the callerid signalling method - set in the zt_pvt strucutre as opposed to the last one read from - the config file (issue #6734, with mods) - -2006-03-16 06:56 +0000 [r13197] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: To quote giant developers: "Oops". Thanks, - Tony! - -2006-03-15 22:16 +0000 [r13161] Russell Bryant <russell@digium.com> - - * cdr.c: - remove some calculations that will always result in 0 - - if a CDR was never started, don't try to calculate a duration and - consider it failed - -2006-03-15 13:01 +0000 [r13026] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6728: Remove parameters to Event: - header on SUBSCRIBE requests - -2006-03-14 18:41 +0000 [r12925-12927] Russell Bryant <russell@digium.com> - - * apps/app_dial.c: when using the G() option to Dial, fix sending - the called channel to 1 priority beyond what was specified (issue - #6523) - - * apps/app_queue.c: fix a problem with not loading realtime queue - members by always reloading a realtime queue from the database - even if it is found in the list (issue #6680) - -2006-03-12 19:26 +0000 [r12646] Russell Bryant <russell@digium.com> - - * pbx.c: add locking to protect the list of global dialplan - variables - -2006-03-12 17:57 +0000 [r12577] Russell Bryant <russell@digium.com> - - * codecs/gsm/Makefile: fix build on parisc (issue #6704) - -2006-03-10 12:13 +0000 [r12477-12495] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #5937 - Make sure SIP CANCEL's are - re-transmitted - - * channels/chan_sip.c: Issue #6576 - SIP_CODEC not used for early - media (reported by gpapadop73) - -2006-03-08 10:51 +0000 [r12458] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue #6657 - Ignore 183 session progress - without SDP - -2006-03-07 00:05 +0000 [r12161-12195] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Bug 6020 - Race condition where packet could - be lost if first packet on list is acked - - * editline/np/vis.c, editline/readline.c: Bug 6664 - More fixes for - Solaris - -2006-03-06 14:23 +0000 [r12036-12072] Olle Johansson <oej@edvina.net> - - * channel.c: Revert earlier change - - * channel.c: Fix for astmm compilation - -2006-03-06 02:32 +0000 [r11946] Russell Bryant <russell@digium.com> - - * configs/zapata.conf.sample: fix a typo in the description of the - ringtimeout option - -2006-03-05 12:40 +0000 [r11849] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Clear page2 flags at reload too - -2006-03-04 11:45 +0000 [r11778] BJ Weschke <bweschke@btwtech.com> - - * apps/app_mixmonitor.c: Substitute variables in the post_process - string (if it exists) before those variables could possibly - disappear (channel hangup) #6462 - -2006-03-03 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.5 released - -2006-03-03 00:38 +0000 [r11607-11635] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * Makefile: Bug 6638 - Use POSIX command for Solaris - - * build_tools/make_build_h: Bug 6638 - Change from a historic BSD - command to a POSIX command for determining username - - * asterisk.c: Bug 6637 - Fixes for Solaris - - * Makefile: If debugging, the frame pointer is helpful - -2006-03-02 19:05 +0000 [r11528-11561] Russell Bryant <russell@digium.com> - - * res/res_monitor.c: fix inaccurate ack message to ChangeMonitor - action (issue #6630) - - * asterisk.sgml: make the terminology used in the synopsis match - the option description - - * asterisk.sgml: add the -L option to the synopsis on the man page - -2006-03-01 17:41 +0000 [r11479-11503] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * cdr/cdr_manager.c, cdr/cdr_tds.c, res/res_config_odbc.c, - include/asterisk/cdr.h, cdr/cdr_pgsql.c, cdr/cdr_odbc.c, cdr.c: - Bug 6615 - Fix 64bit conversion errors by using a long int - - * build_tools/make_svn_branch_name: Bug 6618 - Solaris - compatibility fix - -2006-02-28 19:46 +0000 [r11382-11410] Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: fix the output that indicates whether - qualify smoothing is on or not (issue #6608) - - * asterisk.c: adjust the keys directory when astvarlibdir is - specified in asterisk.conf (issue #6602) - - * res/res_agi.c: add a missing newline in the agi app description - (thanks wunderkin!) - -2006-02-27 15:20 +0000 [r11250-11281] Russell Bryant <russell@digium.com> - - * cli.c: don't try to print the help text for a CLI command when - RESULT_SHOWUSAGE is returned if there is no help text available - (issue #6604) - - * channels/chan_sip.c: fix finding realtime peers that are not - dynamic by ip address (issue #6093) - - * channel.c: don't hang up the channel if its state is set to UP - before we return from ast_call (issue #6569) - -2006-02-26 16:26 +0000 [r11165] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/logger.h, logger.c: Bug 5950 - reenable queue - log rotation; also, eliminate redundant code - -2006-02-25 19:54 +0000 [r11120] Matt Frederickson <creslin@digium.com> - - * translate.c: Backport of fix to translation optimizations. Thanks - again file! - -2006-02-25 05:08 +0000 [r11058-11089] Kevin P. Fleming <kpfleming@digium.com> - - * translate.c: factor the number of translation steps required into - translation path decisions, so that equal cost paths that require - fewer translations are preferred - - * translate.c: reformat code to fit guidelines remember which - translation paths are multi-step paths - - * channel.c: ensure that spy frame queueing is able to deal with - translation failing for any reason (issue #6546) - -2006-02-23 23:06 +0000 [r10952] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * Makefile: set PWD properly - -2006-02-23 14:57 +0000 [r10736-10863] Kevin P. Fleming <kpfleming@digium.com> - - * dnsmgr.c, include/asterisk/linkedlists.h: backport list handling - fix from trunk (solves memory leak problem in cdr variables and - device state watchers) remove unused variable to silence - compiler warning - - * configs/iax.conf.sample: add comment warning people about trying - to use hostnames/IPs in the sample config - -2006-02-20 23:01 +0000 [r10577] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * app.c: Would be nice to tell people to look in the right file to - increase a constant - -2006-02-20 06:17 +0000 [r10511-10535] Mark Spencer <markster@digium.com> - - * channels/chan_sip.c: Handle ACKing properly (remove gratuitous - -1) - - * channels/chan_iax2.c: Fix numerous places in jitter buffer where - freed memory is referenced - -2006-02-19 18:29 +0000 [r10462-10487] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * formats/format_sln.c: Okay, fseek doesn't return an offset - - * apps/app_voicemail.c: Fix possible lack of initialization of - useadsi - - * formats/format_sln.c: Bug 6539 - Division by two negates error - flag - -2006-02-18 00:17 +0000 [r10409] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * app.c: Bug 6529 - memory leak in ast_play_and_prepend - -2006-02-17 01:55 +0000 [r10301-10368] Russell Bryant <russell@digium.com> - - * jitterbuf.c: fix incorrent index calculation for jitterbuffer - history (issue #6517) - - * apps/app_voicemail.c: when executing the Directory application - from voicemail and a context is not specified, use the "default" - context, not the channel's current context (issue #6507) - -2006-02-15 01:21 +0000 [r10108-10137] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_agent.c: ensure that agents logged in via the - manager interface are stored in the persistence database (related - to issue #6301) - - * funcs/func_enum.c: handle longer ENUM lookup results (issue - #6476) - - * res/res_agi.c: ensure that FastAGI launcher can handle system - call interruption (issue #6449) - -2006-02-14 20:56 +0000 [r10021] Matt O'Gorman <mogorman@digium.com> - - * apps/app_meetme.c: bug fix from 6485 with musiconhold not being - turned off by app_meetme - -2006-02-14 20:20 +0000 [r10018] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_queue.c: don't double-increment abandon counter for - calls that are hung up while dialing members (issue #6289) - -2006-02-14 19:11 +0000 [r9990] Mark Spencer <markster@digium.com> - - * apps/app_meetme.c: Fix stopstream in menus (bug #6137) - -2006-02-14 18:50 +0000 [r9961-9964] BJ Weschke <bweschke@btwtech.com> - - * asterisk.c: #ifdef the include too. - - * asterisk.c: #ifdef'd the prctl fix to only try and compile on - linux systems. Thanks rizzo for pointing this out. - -2006-02-14 18:30 +0000 [r9953-9958] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: when answering INVITE, don't send codecs the - peer didn't offer (issue #6052) - - * rtp.c: revert yesterday's temporary fix for issue #6052 - -2006-02-14 04:45 +0000 [r9861-9870] BJ Weschke <bweschke@btwtech.com> - - * asterisk.c: Fixed my silly backport error from r9861 - - * asterisk.c: Merged changes from r9844 from /trunk. Make sure that - PR_SET_DUMPABLE is set to make certain that we still dump core if - Asterisk has setuid'd to run as non-root. - -2006-02-14 00:46 +0000 [r9818] Kevin P. Fleming <kpfleming@digium.com> - - * rtp.c: don't try to use peer's dynamic codec numbers, it leads to - duplication (issue #6052) - -2006-02-13 17:37 +0000 [r9756] Josh Roberson <josh@asteriasgi.com> - - * apps/app_meetme.c: Don't set the formats before we stop - indications. (issue #6380) - -2006-02-11 19:23 +0000 [r9581-9609] Russell Bryant <russell@digium.com> - - * channels/chan_mgcp.c, channels/chan_sip.c, pbx/pbx_dundi.c, - channels/chan_iax2.c: fix memory leak from not destroying the - scheduler context on module unload - - * apps/app_page.c: fix due to CDR changes - - * manager.c, pbx/pbx_spool.c, include/asterisk/channel.h, - include/asterisk/pbx.h, include/asterisk/manager.h, channel.c, - pbx.c: now that CDR is a loadable module, don't depend on it - elsewhere (issue #6460) - -2006-02-11 15:22 +0000 [r9528] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c, cdr.c: clean up my mess from thread-starting - change - -2006-02-11 06:29 +0000 [r9493] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_sip.c: kpfleming's fix from r9472 backported to 1.2 - -2006-02-10 20:38 +0000 [r9404] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_mgcp.c, dnsmgr.c, channels/chan_sip.c, - devicestate.c, channels/chan_modem.c, cdr.c: don't create monitor - threads in detached mode, when we need to be able to - pthread_join() them later if the module is unloaded (solve - crash-on-unload problem for these channel modules) - -2006-02-09 21:10 +0000 [r9323-9326] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Revert behavior change from previous commit - (fixes only) - - * apps/app_voicemail.c: Backport 5929 to 1.2 - -2006-02-09 02:31 +0000 [r9246-9262] Russell Bryant <russell@digium.com> - - * apps/Makefile: add another location for postgresql headers (issue - #6419) - - * channels/chan_iax2.c: reload peercontext on iax2 reload (issue - #6442) - -2006-02-08 22:34 +0000 [r9233] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * cdr/Makefile: Leave it to RH/CentOS to put the freetds headers in - a completely nonstandard location. - -2006-02-08 22:12 +0000 [r9232] Matt O'Gorman <mogorman@digium.com> - - * logger.c, channels/chan_oss.c: Make logger report - error,warning,notice if logger.conf not found, also updated - chan_oss to give correct error message if its config file is not - found. - -2006-02-05 17:10 +0000 [r9156] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_macro.c: Bug 6176 - Fix race condition - -2006-02-02 18:37 +0000 [r9086] Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: don't override ASTERISKVERSIONNUM to 000000 for non-svn - builds - -2006-02-02 16:12 +0000 [r9073] Matt Frederickson <creslin@digium.com> - - * res/res_odbc.c: Fix for (#6309), potential (highly unlikely) - memory leak in res_odbc - -2006-01-30 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.4 Released - -2006-01-30 17:08 +0000 [r8905] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: disable buggy PRI user-user code until it - can be fixed - -2006-01-28 13:52 +0000 [r8808] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 6182 - Don't remove scheduled event - until it's really done. (reported by malverian) - -2006-01-27 08:02 +0000 [r8785] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 6362 - Register without Contact: and - Expires: fails (reporter: op) - -2006-01-27 00:52 +0000 [r8758] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * ast_expr2.h, ast_expr2f.c, ast_expr2.c: Bug 6072 - Revisions to - the source bison and flex files don't auto-regenerate these files - -2006-01-26 19:42 +0000 [r8729] Russell Bryant <russell@digium.com> - - * channels/chan_zap.c: fix problem with dtmf on e&m (issue #6364) - -2006-01-26 14:39 +0000 [r8710] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 5898: Registrations does not get - deleted if there's an active SIP dialog - -2006-01-25 19:14 +0000 [r8666-8677] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: don't call ast_update_realtime with - uninitialized variables if we get a registration with an expirey - of 0 seconds (issue #6173) - - * channels/chan_features.c: fix memory leak (inspired by issue - #6351) - -2006-01-25 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.3 Released - -2006-01-25 09:46 +0000 [r8632] Olle Johansson <oej@edvina.net> - - * channel.c: Issue #6439 - the "timebomb" bug. Patch by Markster - over GPRS - -2006-01-25 05:38 +0000 [r8619] Russell Bryant <russell@digium.com> - - * utils/astman.c: don't leak almost 200 bytes for each new channel - (issue #6330) - -2006-01-25 01:50 +0000 [r8608] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_dial.c: ensure hangup cause code is handled properly - when channel does not return a frame (issue #6346) - -2006-01-24 22:55 +0000 [r8600] Russell Bryant <russell@digium.com> - - * asterisk.c: completely arbitrary whitespace change for testing - something with svnmerge ... - -2006-01-24 22:32 +0000 [r8588] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: ensure that channel cannot become zombie after we - check but before we try to start indications - -2006-01-24 20:37 +0000 [r8573] Matt Frederickson <creslin@digium.com> - - * channels/chan_zap.c: Backport fix for #6229, hangup on polarity - reversal - -2006-01-24 19:21 +0000 [r8537-8562] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Issue 6114: Don't hangup on BYE/ALSO with no - channel. - - * channels/chan_sip.c: Issue #6308 - never send response to ACK. - (Reported by whiskerp) - -2006-01-22 19:03 +0000 [r8437-8445] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: fix memory leak from not freeing the queue - member list when freeing an old queue - - * channel.c: fix MixMonitor crash (issue #6321, probably others) - -2006-01-22 15:13 +0000 [r8433] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_sip.c: Bug fix: Correct some scenarios where - CALL_LIMIT could not be getting adjusted properly allowing - chan_sip to send calls when it really shouldn't. Bug #6111 - -2006-01-22 08:52 +0000 [r8429] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_sip.c: Bug 6281 - Cannot set more than a single - header with SIPAddHeader - -2006-01-22 02:05 +0000 [r8412-8418] Russell Bryant <russell@digium.com> - - * pbx.c: add a modified fix to prevent writing outside of the - provided workspace when calculating a substring (issue #6271) - - * pbx.c: temporarily revert substring fix pending the result of the - discussion in issue #6271 - - * pbx.c: prevent the possibility of writing outside of the - available workspace (issue #6271) - -2006-01-21 18:29 +0000 [r8394] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_queue.c: Bug 5936 - AddQueueMember fails on realtime - queue, if queue not yet loaded - -2006-01-20 18:34 +0000 [r8347] Russell Bryant <russell@digium.com> - - * apps/app_queue.c: fix invalid value of prev_q (issue #6302) - -2006-01-20 01:00 +0000 [r8320] Matt O'Gorman <mogorman@digium.com> - - * channels/chan_iax2.c: solved problem with delayreject and iax - trunking bug 4291 - -2006-01-19 19:40 +0000 [r8281] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Enable "musicclass" setting for sip peers as - per the config sample. - -2006-01-19 19:14 +0000 [r8276] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * ast_expr2.y, ast_expr2.fl: Bug 6072 - Memory leaks in the - expression parser - -2006-01-19 04:56 +0000 [r8232-8242] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: fix Message-Account header to use the ip - address if the fromdomain isn't set (issue #6278) - - * apps/app_milliwatt.c: fix a seg fault due to assuming that space - gets allocatted on the stack in the same order that we declare - the variables (issue #6290) - -2006-01-18 21:02 +0000 [r8194] Matt O'Gorman <mogorman@digium.com> - - * apps/app_meetme.c: Solves issue with the login proccess in meetme - patch from 6136 - -2006-01-18 02:49 +0000 [r8173] Russell Bryant <russell@digium.com> - - * ChangeLog (removed): remove ChangeLog from the 1.2 branch. It - will only be present in the tags. - -2006-01-18 Russell Bryant <russell@digium.com> - - * Asterisk 1.2.2 Released - -2006-01-18 00:47 +0000 [r8140-8162] Matt O'Gorman <mogorman@digium.com> - - * loader.c: Changed order of autoload so that pbx_ comes before - channels, and in doing so cause bug 6002 to not be an issue - - * apps/app_festival.c: Stop any generators running on a channel - when festival is called as described in 5996 - -2006-01-17 18:29 +0000 [r8134] Matt Frederickson <creslin@digium.com> - - * res/res_features.c: Backport of fix for #6094 - -2006-01-17 16:55 +0000 [r8124] Matt O'Gorman <mogorman@digium.com> - - * logger.c: Fixed code ordering of logger_init and queue_log_init - bug 6263 - -2006-01-17 13:11 +0000 [r8112-8122] Kevin P. Fleming <kpfleming@digium.com> - - * asterisk.c: update CLI copyright notice - - * asterisk.c: do rlimit check _after_ reading config file, in case - 'dumpcore' is specified there - -2006-01-14 19:06 +0000 [r8074] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * funcs/func_strings.c: Bug 6238 - Fix segfault when delimiter not - specified - -2006-01-13 06:07 +0000 [r8047] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c: fix spelling errors (issue #6227) - -2006-01-12 06:14 +0000 [r7999] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c, configs/voicemail.conf.sample: Bug 6211 - - Add option deletevoicemail as equivalent to option delete for - Realtime - -2006-01-11 19:08 +0000 [r7965-7986] Russell Bryant <russell@digium.com> - - * channels/chan_agent.c: move variable to correct scope (issue - #6197) - - * apps/app_voicemail.c: fix temp greetings with ODBC storage (issue - #6078) - - * channels/chan_sip.c: fix mem leak on module unload (issue #6190) - - * app.c: don't override an error condition that occurred when - acting on the primary channel when stopping the autoservice on - the peer channel. (from issue #6087) - - * translate.c: lock list of translators *before* recalculating the - translation matrix - -2006-01-11 04:38 +0000 [r7963] Matt O'Gorman <mogorman@digium.com> - - * channel.c: Minor typo refrenced in 6191 - -2006-01-11 04:19 +0000 [r7957-7960] Russell Bryant <russell@digium.com> - - * pbx.c: fix locking error - lock instead of unlock - - * apps/app_dial.c: fix a little typo - -2006-01-11 01:30 +0000 [r7955] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 6192 - behave correctly when mailbox is - specified as argument - -2006-01-10 08:48 +0000 [r7939] Olle Johansson <oej@edvina.net> - - * doc/README.cdr: - Adding reference to README.tds - Reformatting - table - -2006-01-09 22:48 +0000 [r7917] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: re-initialize _all_ sequence numbers when - transfer completes - -2006-01-09 22:07 +0000 [r7915] Russell Bryant <russell@digium.com> - - * file.c: add missing unlock (issue #6112) - -2006-01-09 20:08 +0000 [r7904-7908] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_spool.c: Bug 6157 - Memory leak - - * doc/README.variables: Update variable documentation to match the - code - -2006-01-09 18:11 +0000 [r7898-7900] Kevin P. Fleming <kpfleming@digium.com> - - * asterisk.c: commit user/group-related changes from trunk - - * db.c: backport fix from revision 7856 of trunk - - * apps/app_voicemail.c: fix breakage introduced in revision 7871 - -2006-01-09 05:11 +0000 [r7870-7871] Russell Bryant <russell@digium.com> - - * apps/app_voicemail.c: fix seg fault when using greek syntax in - VoicemMailMain (issue #6142) - - * manager.c: backport fix for unnecessary unlock (issue #6171) - -2006-01-07 07:27 +0000 [r7848] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_spool.c: Bug 6156 - catch all threading errors, not just - simple failure - -2006-01-06 00:34 +0000 [r7831] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_config.c: Dumb error messages - "Context 'context' - already included in 'in' context" - -2006-01-06 00:21 +0000 [r7829] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_agent.c: update agent persistence when an agent - gets logged off by autologoff - -2006-01-05 23:53 +0000 [r7827] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/strings.h: Bug 6076 - Fix documentation of - ast_trim_blank return value - -2006-01-05 23:49 +0000 [r7825] Kevin P. Fleming <kpfleming@digium.com> - - * channel.c: eliminate rounding errors that caused call time limits - to be inaccurate (issue #5913) round 'time left' reported during - call limit warnings up to sound more accurate - -2006-01-05 23:07 +0000 [r7823] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * res/res_features.c: Bug 6081 - fix for memory leak, formatting - fixes - -2006-01-05 20:52 +0000 [r7819] Kevin P. Fleming <kpfleming@digium.com> - - * formats/format_pcm.c, formats/format_pcm_alaw.c: ensure that - variable is initialized - -2006-01-05 09:13 +0000 [r7812] Olle Johansson <oej@edvina.net> - - * res/res_features.c: Fix copyright of changed file - -2006-01-05 00:58 +0000 [r7799-7809] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_agent.c: send device state updates for auto-logoff - of agents as well - - * formats/format_pcm.c, formats/format_pcm_alaw.c: doh... fseek() - has no useful return value - - * formats/format_pcm.c, formats/format_pcm_alaw.c: use proper - fwrite() parameters and return value - - * formats/format_pcm.c, formats/format_pcm_alaw.c: return properly - after extending file - - * formats/format_pcm.c, formats/format_pcm_alaw.c: ensure that - ulaw/alaw sound files are filled with silence when extended (not - zeroes) - - * channel.c: make monitoring more tolerant of peers that deliver - frames in bursts - -2006-01-04 21:46 +0000 [r7792-7795] Olle Johansson <oej@edvina.net> - - * res/res_features.c: Issue #5980: Removing extra CR+LF in manager - events - needs port to trunk - - * channels/chan_sip.c: Fixing typo in XML for video updates. - -2006-01-04 07:06 +0000 [r7773] Russell Bryant <russell@digium.com> - - * funcs/func_moh.c: use a more correct way of determining the size - of the destination buffer - -2006-01-04 05:27 +0000 [r7771] BJ Weschke <bweschke@btwtech.com> - - * apps/app_privacy.c: Fix the 'if' clause to be true under the - right conditions. Bug #6126 - -2006-01-03 20:22 +0000 [r7746] Kevin P. Fleming <kpfleming@digium.com> - - * ast_expr.y (removed): remove unused 'old' expression parser - -2006-01-03 18:15 +0000 [r7743] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_stack.c: Bug 6121 - typo in application description - -2006-01-03 17:24 +0000 [r7736-7740] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/chanspy.h, apps/app_chanspy.c, - apps/app_mixmonitor.c, channel.c: revert incorrect fix for bug - #6048 from revision 7709 put in correct (simpler) fix add doxygen - docs for channel spy 'state' values - - * channels/chan_sip.c: backport rport scanning fix from trunk (bug - #6071) - - * ast_expr2f.c, ast_expr2.fl: don't leak memory for (most) - expression evaluations - -2006-01-02 07:31 +0000 [r7709] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_mixmonitor.c: Bug 6084 - MixMonitor after a 'cli stop - monitor' deadlocks - -2006-01-02 02:04 +0000 [r7706] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_sip.c, channels/chan_iax2.c: Fix compiler warnings. - -2005-12-30 14:54 +0000 [r7677] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channel.c: Bug 6091 - Fix race condition around uniqueid - -2005-12-28 17:35 +0000 [r7663-7665] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: fix memory leak in build_rpid (issue #6070) - - * apps/app_chanspy.c: backport fix for permissions of created - recordings (issue #6067) - -2005-12-27 00:07 +0000 [r7641] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_zap.c: backport fix to ensure that DSP is never - enabled on pseudo channels - -2005-12-26 20:32 +0000 [r7637] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * cdr/cdr_tds.c: Remove copy of code in libc, preferring code in - utils.c (public domain code) - -2005-12-26 18:19 +0000 [r7634] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c, channels/chan_agent.c, apps/app_sms.c, - asterisk.c, config.c, pbx/pbx_dundi.c, apps/app_externalivr.c, - apps/app_queue.c, channels/chan_iax2.c, cli.c, - apps/app_chanspy.c, res/res_monitor.c: cast time_t to an int in - printf/scanf (issue #5635) - -2005-12-23 06:38 +0000 [r7608] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_hasnewvoicemail.c: Bug 6051 - VMCOUNT should work as - documented and count all, not quit after finding 1 - -2005-12-23 03:01 +0000 [r7606] Kevin P. Fleming <kpfleming@digium.com> - - * asterisk.c: add license reference to copyright notice displayed - when CLI session begins add 'show warranty' and 'show license' - CLI commands (still need a complete list of non-GPL components - included in Asterisk) - -2005-12-23 00:00 +0000 [r7605] BJ Weschke <bweschke@btwtech.com> - - * apps/app_waitforsilence.c: Another app documentation tweak. - -2005-12-22 22:04 +0000 [r7601] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 6050 SQL requires the use of single - ticks to delimit values, not quotes - -2005-12-22 20:36 +0000 [r7595-7599] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c, configs/sip.conf.sample: revert changes to - videosupport to allow per-peer setting, since it isn't quite - complete and there is not an obvious fix at this point - - * channels/chan_sip.c: remove stray unlock (issue #5955) - -2005-12-21 22:23 +0000 [r7586] Josh Roberson <josh@asteriasgi.com> - - * channels/chan_sip.c: Actually put in the per-peer settings for - sip video, as they didn't make it in at astricon somehow, and - I've been too busy up until now to redo it. - -2005-12-21 20:01 +0000 [r7582] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * channels/chan_alsa.c: Allow a chan_alsa that failed to open sound - devices to be unloaded. - -2005-12-21 19:53 +0000 [r7580] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_agent.c: Bug #6040 - Documentation correction - -2005-12-21 19:23 +0000 [r7577] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * pbx/pbx_ael.c: Bug 5777 - Remove parentheses on Goto in AEL, so - that it parses correctly - -2005-12-20 20:21 +0000 [r7550-7557] Russell Bryant <russell@digium.com> - - * res/res_agi.c: check array bounds when parsing arguments to AGI - (issue #5868) - - * channels/chan_iax2.c: backport fix for reloading peer context - (issue #6007) - - * apps/app_directed_pickup.c: backport fix for segfault on directed - pickup when no CDR is available (issue #5998) - -2005-12-20 12:58 +0000 [r7546] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_meetme.c: backport fix for larger-than-20ms-frames from - trunk (bug #5697) - -2005-12-19 23:47 +0000 [r7529] Russell Bryant <russell@digium.com> - - * channels/chan_sip.c: I messed up and accidently committed this to - the trunk first ... - add note on required values of sip_methods - struct - remove duplicate function prototype - remove duplicate - ast_mutex_lock (issue #6025) - -2005-12-19 19:06 +0000 [r7521-7523] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * file.c: Bug 5988 - record append option not working - - * cdr.c: Bug 6026 - segfault for the sequence NoCDR(), - SetAMAFlags() - -2005-12-17 18:55 +0000 [r7517-7519] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * doc/README.ael: Document that curley braces must be on the same - line as the keyword. - - * apps/app_chanspy.c: Bug 6009 - off by one error - -2005-12-17 03:59 +0000 [r7510-7515] Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c: Max-Forwards headers must only be present on - requests, not responses - - * channels/chan_sip.c: forcibly expire previous subscriptions from - a peer when they resubscribe (keeps them from building up and - waiting for expiration, and stops us sending unwanted NOTIFY - messages to devices) - - * build_tools/make_svn_branch_name: fix some buglet when building - team branch version strings - -2005-12-17 01:02 +0000 [r7508] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/linkedlists.h: We want to check the previous - value, not the current value (which was just changed). - -2005-12-16 00:49 +0000 [r7497] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_cut.c: First field is truncated - -2005-12-15 10:52 +0000 [r7490] Christian Richter <christian.richter@beronet.com> - - * doc/README.misdn, channels/misdn/mISDNuser.patch (added), - channels/misdn/isdn_lib_intern.h, channels/misdn/mISDN.patch - (added), channels/misdn/isdn_lib.h, channels/chan_misdn.c, - channels/misdn/Makefile, channels/misdn/chan_misdn_config.h, - channels/misdn/ie.c, channels/chan_misdn_config.c, - channels/misdn/isdn_msg_parser.c, channels/Makefile, - channels/misdn/isdn_lib.c: * Added mISDN/mISDNuser Echo cancel - Patch * Fixed Makefiles so that chan_misdn can be compiled again - * added some hints, that mISDN cannot be compiled against gcc-4, - SMP, Spinlock Debug * fixed some Minor issues in chan_misdn, - regarding Type Of Number and Presentation - -2005-12-15 02:51 +0000 [r7482] BJ Weschke <bweschke@btwtech.com> - - * channel.c: Bug #6003 - Don't free the channel structure until - after having sent the manager event. - -2005-12-13 18:54 +0000 [r7435-7470] Kevin P. Fleming <kpfleming@digium.com> - - * doc/README.variables: clarify substring documentation - - * utils.c: correct broken math in tvfix() for timestamp values over - one million - - * apps/app_dial.c: restore ability of caller to hangup calls that - are still ringing (issue #5839) - - * channels/chan_sip.c, pbx.c: ensure that hangups while incoming - calls are in early state are handled properly (issue #5919) - - * channels/chan_agent.c: only report AGENT_IDLE for callback mode - agents when they are actually idle (issue #5902) - - * app.c: use the stream's current point when pausing/unpausing, - instead of elapsed time (which doesn't work when the stream has - been skipped forward or backward) (issue #5897) - - * apps/app_externalivr.c: set all the child file descriptors to - non-blocking so that we don't hang if the child fails to send a - newline-terminated command or error message - -2005-12-12 17:19 +0000 [r7433] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/linkedlists.h: Typo - -2005-12-11 06:08 +0000 [r7430] Russell Bryant <russell@digium.com> - - * utils/astman.c: silence a couple of compiler warnings about - pointer signedness - -2005-12-11 01:26 +0000 [r7427-7429] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * include/asterisk/linkedlists.h: Bug 5965 - major bug in - AST_LIST_REMOVE - - * apps/app_voicemail.c: Bug 5967 - -2005-12-10 18:10 +0000 [r7425] BJ Weschke <bweschke@btwtech.com> - - * channels/chan_zap.c: Bug #5877 Make sure the digit string from - E&M wink DNIS collection is properly null terminated as it grows. - -2005-12-08 23:45 +0000 [r7404-7406] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 5960 - - * configs/res_odbc.conf.sample: Documenting two keywords that were - previously missing - -2005-12-08 01:05 +0000 [r7382-7386] Kevin P. Fleming <kpfleming@digium.com> - - * pbx.c: initialize the buffer before using it... - - * pbx.c: ensure that hints are allowed to use global variable - references - -2005-12-06 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.1 Released - -2005-12-05 06:47 +0000 [r7335-7340] Russell Bryant <russell@digium.com> - - * Makefile: remove ASTERISKVERSIONNUM from the version string given - to doxygen - - * apps/app_queue.c: don't delete dynamic queue members when - reloading the static members from a realtime database (issue - #5922) - - * channels/chan_sip.c: fix the order of arguments to an error - message (issue #5927) - -2005-12-04 18:03 +0000 [r7329] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/make_svn_branch_name: use a more efficient way to get - the revision number, that will also report if the working copy - contains uncommitted modifications - -2005-12-03 19:55 +0000 [r7310] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * apps/app_voicemail.c: Bug 5925: check for "Unknown", as that's - what app_voicemail puts into the field for Unknown callerid Also, - remove useless res checks (initialized to 0; never set) - -2005-12-03 01:24 +0000 [r7299] Olle Johansson <oej@edvina.net> - - * configs/sip.conf.sample: Documenting the default registerattempts - setting as 0, continue hammering the server for ever and ever ;-) - -2005-12-02 21:12 +0000 [r7285] Tilghman Lesher <tilghman@mail.jeffandtilghman.com> - - * contrib/init.d/rc.debian.asterisk, - contrib/init.d/rc.mandrake.asterisk, - contrib/init.d/rc.redhat.asterisk, - contrib/init.d/rc.gentoo.asterisk, - contrib/init.d/rc.mandrake.zaptel, - contrib/init.d/rc.slackware.asterisk: Turn on executable bits for - startup scripts, and fix bash var interpolation for Mandrake - -2005-12-02 00:52 +0000 [r7275] Olle Johansson <oej@edvina.net> - - * channels/chan_sip.c: Bug #5907. Improve SIP INFO DTMF debugging - output. (1.2 & Trunk) - -2005-12-02 00:51 +0000 [r7266-7274] Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_page.c, pbx.c: inherit channel variables into channels - created by Page() application (issue #5888) - - * apps/app_voicemail.c, configs/voicemail.conf.sample, UPGRADE.txt: - allow previous context-searching behavior to be used if desired - (issue #5899) - - * apps/app_voicemail.c: properly handle password changes when - mailbox is last line of config file and not followed by a newline - (issue #5870) reformat password changing code to conform to - coding guidelines (issue #5870) - - * channels/chan_agent.c: protect agent_bridgedchannel() from - segfaulting when there is no bridged channel (issue #5879) - - * channels/chan_local.c: allow variables to exist on both 'halves' - of the Local channel (issue #5810) - - * apps/app_festival.c: don't block waiting for the Festival server - forever when it goes away (issue #5882) - - * channel.c: ensure channel's scheduling context is freed (issue - #5788) - - * Makefile, patches (removed): Makefile 'update' target now - supports updating from Subversion repositories (issue #5875) - remove support for 'patches' subdirectory, it's no longer useful - -2005-12-01 23:18 +0000 [r7261-7265] Olle Johansson <oej@edvina.net> - - * doc/README.misdn: Changing bug report address to the Asterisk - issue tracker - - * doc/README.jitterbuffer, doc/README.realtime: Removing references - to 1.1dev, replacing with 1.2, in documentation files. - - * doc/README.misdn: Fixing some spelling errors, as well as - changing "cvs" to "subversion" in misdn documentation. - -2005-12-01 19:25 +0000 [r7257] Kevin P. Fleming <kpfleming@digium.com> - - * build_tools/make_svn_branch_name: ensure that 'svn info' output - is in the expected language for the script to parse (issue #5880) - -2005-12-01 02:33 +0000 [r7228-7251] Russell Bryant <russell@digium.com> - - * apps/app_externalivr.c: use ast_app_separate_args to split - arguments (issue #5686) - - * apps/app_queue.c: fix queue weight feature - compare member - interfaces instead of pointers to the members, since each queue - has its own list of members. (issue #5863) - - * build_tools/make_svn_branch_name: use '=' instead of '==' for - string comparisons. /bin/bash is ok with this, but /bin/sh is - not. (issue #5885) - - * redhat/asterisk (removed), Makefile: remove outdated redhat init - script and provide the updated one in 'make rpm' (issue #5786) - - * contrib/init.d/rc.debian.asterisk, - contrib/init.d/rc.redhat.asterisk: Comment out LD_ASSUME_KERNEL - by default. Print error messages if the asterisk executable or - the asterisk configuration directory are not found. (issue #5785, - #5708) - - * apps/app_dial.c: fix DIALEDTIME when call has not been answered - (issue #5862) - - * rtp.c: do not allow an rtp message with zero type (issue #5749) - - * pbx.c: fix hint case sensitivity (issue #5856) - - * configs/sip.conf.sample: add description of the "fromdomain" - option (issue #5874) - -2005-11-30 03:52 +0000 [r7227] Josh Roberson <josh@asteriasgi.com> - - * apps/app_voicemail.c, UPGRADE.txt, ChangeLog: backport fix from - trunk - -2005-11-30 03:37 +0000 [r7219-7226] Kevin P. Fleming <kpfleming@digium.com> - - * doc/cdr.txt, doc/CODING-GUIDELINES, include/asterisk.h, - doc/README.mp3: remove remaining CVS references - - * channel.c: port memory leak fix from rev 7223 in trunk - - * include/asterisk/lock.h: do the multiple-lock check for cond_wait - properly... - -2005-11-29 06:12 +0000 [r7216-7218] Russell Bryant <russell@digium.com> - - * apps/app_cut.c: print an error message if invalid arguments are - specified - - * apps/app_skel.c: fix a couple of typos and a buglet - -2005-11-29 01:25 +0000 [r7199-7213] Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/lock.h: if the lock protected a pthread_cond is - held recursively, warn before waiting onthe condition - - * Makefile, build_tools/make_svn_branch_name (added): port version - string computation from trunk - - * / (added): branch renames remove unneeded branches - -2005-11-29 Josh Roberson <josh@asteriasgi.com> - - * apps/app_voicemail.c: Only look in 'default' context when no context defined to VoiceMailMain(). (issue #5887) - -2005-11-25 Russell Bryant <russell@digium.com> - - * apps/app_dial.c: Properly duplicate the string for ANI (issue #5850) - -2005-11-23 Russell Bryant <russell@digium.com> - - * configs/voicemail.conf.sample: Add note to indicate that #include should not be used for this file. (issue #5828) - - * indications.c: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826) - * configs/indications.conf.sample: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826) - * include/asterisk/indications.h: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826) - * res/res_indications.c: Fix spelling of "cadence", allowing the old misspelling for backwards compatability. (issue #5826) - - * apps/app_voicemail.c: Remove left over "yay!" debugging message. (issue #5829) - -2005-11-21 Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_cut.c: remove unnecessary include that causes spurious rebuilding - - * channels/chan_sip.c (build_peer): ensure that case changes made to peer names are not ignored during reload operations - (build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified - - * channels/chan_iax2.c (build_peer and build_user): ensure that case changes made to peer/user names are not ignored during reload operations - (build_peer): when a peer is changed from dynamic to static mode, reset the default port number if no other has been specified - -2005-11-21 Russell Bryant <russell@digium.com> - - * Makefile: Revert previous change for Darwin. - - * apps/app_osplookup.c: Properly populate the number of results. (issue #5789) - - * Makefile: Don't hard-code that poll functionality needs to be provided on Darwin. - * apps/Makefile: Fix incorrect portion of the patch to fix 'make install' on Solaris. - - * channels/chan_iax2.c (iax2_getpeername): Return non-zero to indicate that a peer was found when using realtime (issue #5815) - -2005-11-20 Russell Bryant <russell@digium.com> - - * Makefile apps/Makefile: Fix 'make install' for Solaris. (issue #5775) - - * apps/app_record.c: Don't leak a frame if writing it to the file fails. (issue #5787) - - * Makefile: Create the monitor spool directory when the other spool directories are created. - - * channels/chan_sip.c channels/chan_iax2.c: Change warning messages about the number of scheduled events happening all at once to debug messages. (issue #5794) - - * pbx/pbx_spool.c: Fix crash when a value is not specified with a variable on a Set: line in a call file. (issue #5806) - - * apps/app_meetme.c: Fix the 'X' option to the MeetMe application. (issue #5773) - - * apps/app_voicemail.c: Correct the use of a mailbox entered by the calling party instead of indicated as an argument to the Voicemail application. (issue #5774) - - * apps/app_controlplayback.c: Fix logic in checking for success when jumping to priority n+101. - * apps/app_md5.c: Fix logic in checking for success when jumping to priority n+101. - - * apps/app_hasnewvoicemail.c: Fix a typo in the application description. Also, fix the logic in checking for success when jumping to priority n+101. (issue #5795) - - * UPGRADE.txt: Add a note on a second way that the IAX2 channel naming convention has changed. (issue #5792) - * channels/chan_iax2.c: Fix alignment of the output for the "iax2 show peer <peer>" CLI command (issue #5792) - - * channels/Makefile: Re-add chan_oss to the default build. (issue #5799) - - * res/res_musiconhold.c: Fix incorrect argument for the buffer size to an ast_copy_string call (issue #5803) - - * funcs/func_enum.c: Shorten the module description (issue #5791) - -2005-11-17 Russell Bryant <russell@digium.com> - - * Makefile: Fix the output of Makefile generated variables to doxygen - - * channels/chan_sip.c: Add missing carriage return and line feed to the SDP line indicating that we don't support VAD (issue #5780) - -2005-11-16 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.0 released. - -2005-11-16 Jeremy McNamara <jj@nufone.net> - - * apps/app_voicemail.c (load_config): do not terminate asterisk if no voicemail config file - * channels/chan_skinny: Don't register channel type until ready, code formatting updates - -2005-11-16 Josh Roberson <josh@asteriasgi.com> - - * Makefile: Update to fix non-responsive remote console on Darwin (OSX)(issue #5757) - -2005-11-16 Kevin P. Fleming <kpfleming@digium.com> - - * channels/Makefile: don't build chan_modem and sub-modules by default - * configs/modules.conf.sample: explicitly 'noload' chan_modem.so and submodules, in case old versions exist - - * res/Makefile: issue mpg123 not-installed warning at 'make install' time, not 'make' - - * apps/app_forkcdr.c (forkcdr_exec): issue warning (and don't segfault) if ForkCDR is called on a channel that doesn't have a CDR (issue #5763) - - * channel.c (ast_queue_hangup): ensure that the channel lock is held before changing its fields... (issue #5770) - - * res/res_musiconhold.c: don't spit out incorrect log messages (and leak memory) during reload (issue #5766) - - * channels/chan_sip.c (process_sdp): don't pass video codec number into ast_getformatname(), it is not valid input for that function (issue #5764) - - * pbx/pbx_ael.c (match_assignment): properly parse equal signs surrounded by whitespace (issue #5761) - - * doc/README.realtime: document the limitations of using FreeTDS with Realtime (issue #5767) - -2005-11-15 Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: use -g3 for compiler to include macro information for debugger - - * astmm.c (__ast_vasprintf): don't re-use the ap list without copying it; that's not safe on some platforms (issue #5035) - - * doc/README.backtrace: add note about properly building Asterisk to be able to produce backtraces; wrap text and remove DOS line endings - - * channels/chan_sip.c (add_codec_to_sdp): add 'annexb=no' to G.729A SDP (issue #5539) - - * channels/chan_alsa.c (alsa_hangup): handle autohangup properly (issue #5672) - - * channels/chan_misdn.c (and other files): various fixes (issue #5739) - - * channels/chan_sip.c (handle_request_info): properly forward 'flash' events received via SIP INFO (issue #5751, different patch) - - * apps/app_disa.c (disa_exec): don't duplicate constant strings when not needed - - * apps/app_playback.c (playback_exec): use correct logic tests for options (issue #5752) - - * apps/app_disa.c (disa_exec): use standard arg parsing routines (issue #5736) - -2005-11-15 Russell Bryant <russell@digium.com> - - * manager.c: Don't crash on a SetVar action if the channel name is not set, or variable's value is not set (issue #5760) - - * doc/README.variables: Add application exit status variables - -2005-11-14 Josh Roberson <josh@asteriasgi.com> - - * manager.c: Fix crash on variable passing from AMI originate (issue #5737) - -2005-11-14 Russell Bryant <russell@digium.com> - - * many files: Merge doxygen documentation updates. (issue #5605) - - * apps/app_dial.c: Fix typo in RetryDial description. - -2005-11-12 Russell Bryant <russell@digium.com> - - * channels/chan_oss.c: Fix a typo in an error message. - -2005-11-11 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.0-rc2 released. - -2005-11-11 Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_sip.c (thread_safe_rand): ensure that threads don't get the same random number (issue #5712) - - * apps/app_voicemail.c (forward_message): correct bugs in message forwarding (issue #5718) - (copy_message): use correct path for locking (issue #5704) - - * apps/app_dial.c (wait_for_answer): correct flag copying for automon feature (issue #5720) - - * channels/chan_iax2.c: correct comment - - * apps/app_voicemail.c (close_mailbox): correct previous commit (issue #5663) - (vm_change_password): fix password change writing (issue #5721) - - * channels/chan_sip.c (transmit_invite): remove useless debug message; don't try to add OSP tokens to OPTIONS pings - - * apps/app_voicemail.c (close_mailbox): properly remove deleted messages at mailbox close time (issue #5663) - -2005-11-11 Mark Spencer <markster@digium.com> - - * channels/chan_zap.c (zt_bridge): only enable/disable DTMF detection on SUB_REAL channels - -2005-11-10 Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c: ensure that system headers that provide basic types are included first (issue #5713) - -2005-11-11 Russell Bryant <russell@digium.com> - - * many files in apps/: Clean up application descriptions. Clarify some wording and make sure they wrap at 80 characters. - -2005-11-10 Mark Spencer <markster@digium.com> - - * rtp.c (ast_rtp_raw_write): use unsigned int for return value from calc_txstamp() (issue #5595) - (calc_txstamp): never return a value that was less than zero before being turned into 'unsigned int' (issue #5595) - -2005-11-10 Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/chanspy.h: move spy-related stuff into separate header so chan_h323 can build (issue #5590) - - * include/asterisk/linkedlists.h (AST_LIST_HEAD_SET_NOLOCK): properly initialize tail pointer when list head is directly set (issue #5669) - - * app.c (ast_app_parse_options): ok, so we aren't all perfect... let's make the while loop actually work properly here (issue #5684) - - * apps/app_disa.c (disa_exec): correct password file parsing (issue #5676) - - * apps/app_meetme.c (conf_run): don't restrict admin users from joining a locked conference (issue #5680) - - * channels/chan_misdn.c: include stdio.h (issue #5671) - * channels/chan_misdn_config.c: fix prototype warning (issue #5671) - - * pbx.c: remove apps that were deprecated before 1.0 was released (issue #5673) - - * apps/app_striplsd.c, apps/app_substring.c: remove apps that were deprecated before 1.0 was released (issue #5673) - - * include/asterisk/lock.h (PTHREAD_MUTEX_RECURSIVE_NP): work around header problems on Cygwin (issue #5668) - - * pbx/pbx_ael.c: handle switch default cases inside macros properly (issue #5354) - - * configs/voicemail.conf.sample (format): add strong warning about changing format list when mailboxes contain messages (issue #5689) - - * many files: ensure that system headers are included before Asterisk headers (issue #5693) - - * channels/chan_iax2.c (complete_iax2_show_peer): don't return from function without releasing lock (issue #5685) - - * channels/iax2-provision.c (iax_provision_reload): don't leak memory (issue #5700) - - * pbx/pbx_ael.c (handle_macro): don't leak memory (issue #5701) - (handle_context): ditto - - * res/res_features.c (load_config): properly initialize referenced variable (issue #5703) - - * apps/app_queue.c (rqm_exec): correct segfault problem (issue #5705) - (aqm_exec): ditto - - * app.c (ast_app_parse_options): don't increment 's' until after checking for NULL (related to issue #5630) - - * apps/app_rpt.c: solve a memory leak (config structure was not freed) (issue #5706) - -2005-11-10 Russell Bryant <russell@digium.com> - - * app.c (ast_app_separate_args): Don't consider the open parenthesis as part of the arguments to an option. (issue #5630) - - * many files: Change all references to ast_separate_app_args to ast_app_separate_args - - * many files in apps/: Clean up some application descriptions. Make sure all descriptions in changed files are wrapped at 80 characters. - -2005-11-09 Russell Bryant <russell@digium.com> - - * pbx.c: Clean up descriptions of built-in dialplan applications. Changes include clearer wording and not referring to return values. - -2005-11-09 Kevin P. Fleming <kpfleming@digium.com> - - * channels/chan_iax2.c (update_registry): don't complain about unspecifed registration expiration intervals, just use the minimum - -2005-11-08 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.0-rc1 released. - - * include/asterisk/file.h: add test to ensure that stdio.h is included before this file (issue #5658) - - * res/res_odbc.c (odbc_prepare_and_execute): add new API call for use to properly handle prepared statements across server disconnects (issue #5563) - - * pbx.c (pbx_substitute_variables_helper_full): use already-substituted buffer for parsing variable name (issue #5664) - - * channels/chan_zap.c (zt_request): return AST_CAUSE_CONGESTION when a group-channel is requested and the group exists but all channels are busy (issue #3360, related fix) - * channels/chan_iax2.c (create_addr): treat UNREACHABLE as AST_CAUSE_UNREGISTERED so that it will generate CHANUNAVAIL from app_dial (issue #3360) - - * res/res_features.c (ast_bridge_call_thread_launch): set SCHED_RR separately from thread creation, so it won't fail when running as non-root (issue #5601, different fix) - - * pbx.c (pbx_builtin_pushvar_helper): add new API function for setting variables that can exist multiple times (issue #2720) - * apps/Makefile (APPS): add app_stack (issue #2720) - * apps/app_stack.c: new applications (issue #2720) - - * apps/app_meetme.c: fix two audio delay problems related to using non-Zap channels in conferences (issues #3599 and #4252) - * configs/meetme.conf.sample: add documentation of new 'audiobuffers' setting to control buffering on incoming audio from non-Zap channels - - * channels/chan_local.c (local_call): move channel variables from incoming to outgoing instead of inheriting them (issue #5604) - - * many files: add explicit include of stdio.h (issue #5650) - -2005-11-07 Kevin P. Fleming <kpfleming@digium.com> - - * UPGRADE.txt (Parking): add note about new parking behavior (issue #5532) - - * many files: more Cygwin compatibility, and proper getloadavg() prototype/macro (issue #5569) - - * include/asterisk/lock.h (__ast_pthread_mutex_lock): correct build with DETECT_DEADLOCKS defined (issue #5570) - -2005-11-07 Russell Bryant <russell@digium.com> - - * apps/app_queue.c: upgrade to new arg/option API and implement priority jumping control (issue #5580) - * many files: Add missing include of stdio.h, and remove some duplicate and unused header includes - - * include/asterisk/app.h: Increment the arg_index in the options structure to fix applicaiton options that have arguments to them - -2005-11-07 Kevin P. Fleming <kpfleming@digium.com> - - * cryptostub.c: include necessary headers - * include/asterisk/crypto.h: don't include unnecessary headers - - * manager.c (action_setvar): add support for setting global variables (issue #5571) - - * Makefile: correct cross-compilation issue introduced in Cygwin patches (issue #5572) - - * apps/app_voicemail.c: upgrade to new arg/option API and implement priority jumping control (issue #5649) - - * asterisk.c (main): setpriority() failure is not a reason to stop the process (issue #5581) - - * say.c (ast_say_date_with_format_da): say hours properly (issue #5576) - - * manager.c (astman_get_variables): restore old multiple-variable behavior for "Variable" header (issue #5585) - - * many files: don't check for NULL before calling ast_strlen_zero, it can do it itself (issue #5648) - - * pbx.c (handle_show_hints): use proper state-to-string function for hint state (issue #5583) - - * rtp.c: use unsigned format for debug packet output (issue #5595) - - * asterisk.c (main): force a dnsmgr background refresh after all other modules are initialized (issue #5599) - * dnsmgr.c: add ability to start a background refresh on demand (issue #5599) - - * apps/app_dial.c (HANDLE_CAUSE): set CDR disposition to match cause code (issue #5602) - - * asterisk.c: support 'runuser' and 'rungroup' options in asterisk.conf (issue #5621) - - * res/Makefile, apps/Makefile, channels/Makefile, Makefile: support WITHOUT_ZAPTEL define to forcibly avoid building Zaptel support (issue #5634) - - * Makefile: various fixes (issue #5633) - - * apps/app_osplookup.c: upgrade to new arg/option API and implement priority jumping control - - * channels/chan_misdn.c: various fixes (issue #5639) - * channels/misdn/isdn_lib.c: various fixes (issue #5639) - - * apps/app_playback.c: upgrade to new arg/option API and implement priority jumping control - - * apps/app_privacy.c: upgrade to new arg/option API and implement priority jumping control - - * apps/app_sendtext.c: upgrade to new arg/option API and implement priority jumping control - - * apps/app_transfer.c: upgrade to new arg/option API and implement priority jumping control - - * apps/app_txtcidname.c: upgrade to new arg/option API and implement priority jumping control - - * Makefile: restore function of 'dont-optimize' - - * config.c (config_text_file_load): don't generate log message when stat() fails - - * many files: clean up application documentation to not refer to return values, since they cannot be used in the dialplan (work done by Neil Lewis) - -2005-11-06 Russell Bryant <russell@digium.com> - - * many files: alphabetize options in applicaiton descriptions - - * channels/chan_iax2.c: Use an enum to define iax peer/user flags as well as the pvt structure state. Use the ast_flags macros for checking or setting the state. - - * sounds.txt: Add missing words from the description of the vm-opts prompt - - * apps/app_externalivr.c: Add a space that fixes building on older versions of gcc - - * many files: Add doxygen updates to categorize modules into groups. Convert a lot of comments over to doxygen style. Add some text giving a basic overview of channels. - - * many files: Update applications to add an exit status variable, make priority jumping optional, and use new args parsing macros - - * pbx.c cdr.c res/res_features.c apps/app_dial.c include/asterisk/cdr.h: Convert some built-in applications to use new args parsing macros. Change ast_cdr_reset to take a pointer to an ast_flags structure instead of an integer for flags. - - * channels/chan_agent.c: Don't loop forever on an invalid options string - - * apps/app_disa.c apps/app_forkcdr.c: Fix to use correct arguments to ast_cdr_reset - -2005-11-05 Kevin P. Fleming <kpfleming@digium.com> - - * Makefile: don't rebuild asterisk/build.h unless the asterisk binary is going to be relinked for some other reason (stops spurious recompile/link every time 'make' is issued); clean up variable substitutions to use consistent syntax - * asterisk.c: don't include asterisk/build.h (it's unnecessary) - * cli.c: don't include asterisk/build.h, use extern references to buildinfo.c - * buildinfo.c: new file to hold version info strings - -2005-11-04 Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_mixmonitor.c (mixmonitor_exec): correct app name in an error message - -2005-11-04 Russell Bryant <russell@digium.com> - - * channels/chan_iax2.c: Create a function that stores a peer's status in a given buffer. Use this function in "iax2 show peers" and "iax2 show peer <peername>". Also, add the peer's status as an option to the IAXPEER dialplan function. - -2005-11-04 Kevin P. Fleming <kpfleming@digium.com> - - * include/asterisk/compiler.h: don't try to use always_inline on old compilers - -2005-11-03 Russell Bryant <russell@digium.com> - - * res/res_agi.c: initialize buffer for result so that the contents are always valid in the response to GET FULL VARIABLE - -2005-11-03 Kevin P. Fleming <kpfleming@digium.com> - - * doc/README.variables: document DYNAMIC_FEATURES - - * res/res_features.c (ast_bridge_call): remove unused variables - - * apps/app_dial.c (dial_exec_full): simplify options and flag usage - - * include/asterisk/app.h: re-work application arg/option parsing APIs for consistent naming, add doxygen docs for option API - * many files: update to new APIs - -2005-11-02 Kevin P. Fleming <kpfleming@digium.com> - - * apps/app_dial.c (dial_exec_full): convert to use API calls for argument/option parsing - - * include/asterisk/channel.h: add doxygen docs for silence generator APIs - - * channel.c (ast_channel_bridge): simplify native-bridge return logic, remove 'unsuccessful' message since it causes too many questions :-) - -2005-11-01 Kevin P. Fleming <kpfleming@digium.com> - - * stdtime/localtime.c: fix build failure on uClibc systems (issue #5558) - * devicestate.c: same - - * many files: make chan_misdn actually build (issue #5566) - - * many files: more Cygwin build system support (issue #4678) - - * apps/app_parkandannounce.c (parkandannounce_exec): supply parent channel to ast_request_and_dial so channel variables can be inherited (issue #5564) - * include/asterisk/channel.h: add parent_channel field - * channel.c (__ast_request_and_dial): use parent_channel field to inherit variables into new channel - - * apps/app_cut.c (cut_internal): use ast_app_separate_args() instead of open code (issue #5560) - - * apps/app_mixmonitor.c (launch_monitor_thread): ast_strlen_zero can handle NULL input (issue #5561) - (mixmonitor_exec): same - - * res/res_features.c (ast_feature_request_and_dial): ensure that channel variables are inherited from the channel placing the call (issue #5499) - - * utils.c (getloadavg): change to using _BSD_SOURCE as the indicator for whether this function is present or not (issue #5549) - - * include/asterisk/utils.h (ast_slinear_saturated_add): force to be inlined whenever possible - (ast_slinear_saturated_multiply): same - (ast_slinear_saturated_divide): same - (inaddrcmp): same - * include/asterisk/strings.h (ast_strlen_zero): force to be inlined whenever possible - * include/asterisk/compiler.h (force_inline): add macro to force inlining of functions - - * app.c (ast_play_and_record): use ast_silence_generator during recording if requested - * asterisk.c: add global option to enable silence-during-record (issue #5135) - * channel.c (silence_generator_alloc): new - (silence_generator_release): new - (silence_generator_generate): new - (ast_channel_start_silence_generator): new API call to start generating silence on a channel - (ast_channel_stop_silence_generator): parallel call to stop silence generation - * apps/app_record.c (record_exec): use ast_silence_generator during recording if requested - -2005-11-01 Kevin P. Fleming <kpfleming@digium.com> - - * Asterisk 1.2.0-beta2 released. - |