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authorkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2006-12-14 13:18:34 +0000
committerkpfleming <kpfleming@f38db490-d61c-443f-a65b-d21fe96a405b>2006-12-14 13:18:34 +0000
commit22899319e425cc29fa13a234843da888ae699f15 (patch)
treed9835f1734f83ed1d461cb550e57fb63e45e9365
parent4fec88301b20c81e74d6711a6b95d0db56a8fb61 (diff)
parent2731a4087d5d2dfed32253ed8a69000895352be4 (diff)
Creating tag for the release of asterisk-1.2.14-netsec
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.2.14-netsec@48470 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--.lastclean1
-rw-r--r--.version1
-rw-r--r--ChangeLog4270
-rw-r--r--channel.c7
4 files changed, 5 insertions, 4274 deletions
diff --git a/.lastclean b/.lastclean
deleted file mode 100644
index ec635144f..000000000
--- a/.lastclean
+++ /dev/null
@@ -1 +0,0 @@
-9
diff --git a/.version b/.version
deleted file mode 100644
index 9a7b3f306..000000000
--- a/.version
+++ /dev/null
@@ -1 +0,0 @@
-1.2.14-netsec
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.
-
diff --git a/channel.c b/channel.c
index 539b3d9c1..4779bf8d9 100644
--- a/channel.c
+++ b/channel.c
@@ -39,7 +39,7 @@
#include <zaptel.h>
#endif /* __linux__ */
#ifndef ZT_TIMERPING
-#error "You need newer zaptel! Please cvs update zaptel"
+#error "You need newer zaptel! Please svn update zaptel"
#endif
#endif
@@ -3351,7 +3351,10 @@ static enum ast_bridge_result ast_generic_bridge(struct ast_channel *c0, struct
if (bridge_end.tv_sec) {
to = ast_tvdiff_ms(bridge_end, ast_tvnow());
if (to <= 0) {
- res = AST_BRIDGE_COMPLETE;
+ if (config->timelimit)
+ res = AST_BRIDGE_RETRY;
+ else
+ res = AST_BRIDGE_COMPLETE;
break;
}
} else