[asterisk-commits] kpfleming: tag 1.2.14-netsec r48423 -
/tags/1.2.14-netsec/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Dec 12 16:01:02 MST 2006
Author: kpfleming
Date: Tue Dec 12 17:01:02 2006
New Revision: 48423
URL: http://svn.digium.com/view/asterisk?view=rev&rev=48423
Log:
importing files for 1.2.14-netsec release
Added:
tags/1.2.14-netsec/.lastclean (with props)
tags/1.2.14-netsec/.version (with props)
tags/1.2.14-netsec/ChangeLog (with props)
Added: tags/1.2.14-netsec/.lastclean
URL: http://svn.digium.com/view/asterisk/tags/1.2.14-netsec/.lastclean?view=auto&rev=48423
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--- tags/1.2.14-netsec/ChangeLog (added)
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@@ -1,0 +1,4270 @@
+2006-12-12 Kevin P. Fleming <kpfleming at digium.com>
+
+ * Asterisk 1.2.14 released
+
+2006-12-12 05:11 +0000 [r48403] Kevin P. Fleming <kpfleming at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+ * channels/chan_sip.c: Remove unused variable (rizzo)
+
+2006-11-22 02:19 +0000 [r47910] Steve Murphy <murf at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * doc/README.cdr: update clearly wrong documentation regarding
+ cdr_custom
+
+2006-11-16 20:29 +0000 [r47750-47761] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * apps/app_voicemail.c: Update copyright information in the ADSI
+ logo blob.
+
+2006-11-14 11:06 +0000 [r47596] Olle Johansson <oej at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at edvina.net>
+
+ * channels/chan_sip.c: Remove dialog properly at unload of module
+ (rizzo)
+
+2006-11-07 18:22 +0000 [r47274] Steve Murphy <murf at 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 at 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 at 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 at 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 at edvina.net>
+
+ * channels/chan_sip.c: Don't overwrite flags in the packet
+
+2006-11-02 13:55 +0000 [r46876] Russell Bryant <russell at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * cdr/cdr_odbc.c: Don't leak memory mmmk?
+
+2006-10-21 12:35 +0000 [r45808] Christian Richter <christian.richter at 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 at 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 at digium.com>
+
+ * Asterisk 1.2.13 released
+
+2006-10-17 20:37 +0000 [r45380] Joshua Colp <jcolp at 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 at digium.com>
+
+ * channels/chan_skinny.c: Fix an integer signedness problem.
+
+2006-10-17 17:22 +0000 [r45326] Kevin P. Fleming <kpfleming at digium.com>
+
+ * LICENSE: provide licensing language for IAXy firmware file
+
+2006-10-17 15:50 +0000 [r45306] Olle Johansson <oej at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * cli.c: proper fix for ast_group_t change
+
+2006-09-28 18:00 +0000 [r43924] Joshua Colp <jcolp at 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 at 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 at 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 at digium.com>
+
+ * cli.c: eliminate compiler warning introduced by recent changes
+
+2006-09-28 16:13 +0000 [r43891] Joshua Colp <jcolp at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at digium.com>
+
+ * configs/zapata.conf.sample: Clarified what "callwaiting" does in
+ zapata.conf.
+
+2006-09-18 15:05 +0000 [r43159] Joshua Colp <jcolp at 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 at mail.jeffandtilghman.com>
+
+ * apps/app_directory.c: Directory used the wrong context for
[... 3384 lines stripped ...]
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