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authorlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-01-21 13:19:16 +0000
committerlmadsen <lmadsen@f38db490-d61c-443f-a65b-d21fe96a405b>2009-01-21 13:19:16 +0000
commit503ebba846b30a4d9e493a7e8578f9c40cf89bf4 (patch)
tree6cbe6c63e6e2dc3c25cd344c1cbee12b45143d54
parentd84f827ff6a3ac7853c45205604010ef13715276 (diff)
Creating tag for asterisk-1.4.23 (in the right location this time too!)
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.23@169668 f38db490-d61c-443f-a65b-d21fe96a405b
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-rw-r--r--1.4.23-rc4/res/res_clioriginate.c175
-rw-r--r--1.4.23-rc4/res/res_config_odbc.c564
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diff --git a/1.4.23-rc4/.cleancount b/1.4.23-rc4/.cleancount
new file mode 100644
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--- /dev/null
+++ b/1.4.23-rc4/.cleancount
@@ -0,0 +1 @@
+33
diff --git a/1.4.23-rc4/.lastclean b/1.4.23-rc4/.lastclean
new file mode 100644
index 000000000..bb95160cb
--- /dev/null
+++ b/1.4.23-rc4/.lastclean
@@ -0,0 +1 @@
+33
diff --git a/1.4.23-rc4/.version b/1.4.23-rc4/.version
new file mode 100644
index 000000000..dcce030a5
--- /dev/null
+++ b/1.4.23-rc4/.version
@@ -0,0 +1 @@
+1.4.23-rc4
diff --git a/1.4.23-rc4/BUGS b/1.4.23-rc4/BUGS
new file mode 100644
index 000000000..96b55e655
--- /dev/null
+++ b/1.4.23-rc4/BUGS
@@ -0,0 +1,22 @@
+Asterisk Bug Tracking Information
+=================================
+
+To learn about and report Asterisk bugs, please visit
+the official Asterisk Bug Tracker at:
+
+ http://bugs.digium.com
+
+For more information on using the bug tracker, or to
+learn how you can contribute by acting as a bug marshal
+please see:
+
+ http://www.asterisk.org/developers/bug-guidelines
+
+If you would like to submit a feature request, please
+resist the temptation to post it to the bug tracker.
+Feature requests should be posted to the asterisk-dev
+mailing list, located at:
+
+ http://lists.digium.com
+
+Thank you!
diff --git a/1.4.23-rc4/CHANGES b/1.4.23-rc4/CHANGES
new file mode 100644
index 000000000..699674279
--- /dev/null
+++ b/1.4.23-rc4/CHANGES
@@ -0,0 +1,370 @@
+Changes since Asterisk 1.2:
+
+ * over 4,000 commits since 1.2
+ * queue member naming
+ * CLI commands rework
+ o Change the way CLI commands are structured.
+ o Most commands are now <module> <verb> <args>
+ * chan_h323 update
+ * RTP packetization
+ * SLA (Shared Line Appearance) support
+ * T.38 Passthrough Support for faxing in SIP
+ * Generic channel jitterbuffer (spawned from RTP)
+ * Variable Length DTMF for better DTMF compatibility
+ * Improved chan_iax2 scalability by using multithreading
+ * AEL2 has replaced the original implementation of AEL. The "2" is removed. For more details,
+ read: http://www.voip-info.org/wiki/view/Asterisk+AEL2
+ AEL is no longer considered experimental.
+ * New sounds; English, Spanish, and French prompts, as well as music on hold files, in
+ multiple Asterisk native formats.
+ * IMAP storage of voicemail
+ * Jabber/GoogleTalk integration
+ * New speech recognition API for interfacing to different Voice Recognition software packages
+ * much more customizable and portable build system
+ o also for asterisk-addons
+ * Radius CDR logging
+ * SNMP support
+ * SMDI (Simplified Message Desk Interface) support
+ * Redesign of MusicOnHold configuration settings
+ * Manager over HTTP
+ * Significant chan_skinny updates
+ * Significant chan_misdn updates
+ * Improved SIP transfers
+ * SIP MWI subscription support
+ * Much improved support for SIP video
+ * Control over SIP transfers and subscriptions (enable/disable per device)
+ * ChanSpy whisper mode (Whisper Paging)
+ * Configurable language support for saying dates and times
+ * Significant architecture improvements for memory usage and performance
+ * Media-only IAX2 transfers
+ * Updates to the Radio Repeater app code
+ * Deprecation of AgentCallbackLogin in favor of a dialplan-based solution
+ * uClibc builds supported
+ * Work done for freeBSD portability
+ * Work done for Solaris portability
+ * FreeTDS-based database can be used with Realtime
+ * New internal data structure, stringfields, is implemented in IAX and SIP, reducing memory consumption by about 50%.
+ * Use of thread local storage for reduced memory allocation/freeing and lower stack consumption
+ * Reorganized files into docs/ main/ configs/, including name changes in some cases
+ * Much effort was expended in arranging documentation in source files in doxygen format
+ * Improved IP TOS support for IAX and SIP
+ * Builtin mini HTTP server
+ * Added support for Sigma Designs cards.
+ * Frame header caching to reduce memory allocation/freeing
+ * Passthrough and record/playback support for G.722 wideband audio
+ * using mpg123 to play MP3 files for music-on-hold will be deprecated in 1.4 (start using the "native support")
+ * New Apps:
+ 1. AMD() ;; Answering Machine Detection
+ 2. ChannelRedirect() ;; asynch goto, redirect chan to context/exten/priority
+ 3. ContinueWhile() ;; Addition to the While() suite. Acts like "continue".
+ 4. ExitWhile() ;; Addition to the While() suite. Acts like "break".
+ 5. ExtenSpy() ;; A close cousin to ChanSpy().
+ 6. FollowMe() ;; findme/followme call redirect app
+ 7. Log() ;; Send a message to the log, based on severity level.
+ 8. MacroExclusive() ;; No more than one invocation of this macro allowed at any one time.
+ 9. MorseCode() ;; turns strings into dits and dahs. A playground for ham radio licensees!
+ 10. OSPAuth() ;; OSP authentication
+ 11. QueueLog() ;; allows you to write your own events into the queue log
+ 12. SLAStation() ;; Shared Line Appearance
+ 13. SLATrunk() ;; Shared Line Appearance
+ 14. SpeechCreate() ;; Voice Recognition Engine interface...
+ 15. SpeechActivateGrammar()
+ 16. SpeechStart()
+ 17. SpeechBackground
+ 18. SpeechDeactivateGrammar()
+ 19. SpeechProcessingSound()
+ 20. SpeechDestroy()
+ 21. SpeechLoadGrammar()
+ 22. SpeechUnloadGrammar()
+ 23. StopMixMonitor() ;; to stop the MixMonitor App.
+ 24. TryExec() ;; execute dialplan app without fatal consequences
+ * Apps removed:
+ 1. CheckGroup -- do a comparison to ${GROUP()}
+ 2. Curl -- use the function CURL() instead
+ 3. Cut -- use the function CUT() instead
+ 4. DateTime -- use sayunixtime() app instead.
+ 5. DBget -- deprecated in 1.2, now removed.
+ 6. DBput -- deprecated in 1.2, now removed.
+ 7. Enumlookup -- use the function ENUMLOOKUP() instead
+ 8. Eval -- use the function EVAL() instead
+ 9. GetGroupCount -- use the function GROUP_COUNT() instead
+ 10. GetGroupMatchCount -- use the function GROUP_MATCH_COUNT() instead
+ 11. Intercom -- use the chan_oss module instead
+ 12. Math -- use the function MATH() instead
+ 13. MD5 -- use the function MD5() instead
+ 14. SetCIDname -- use the function CALLERID(name) instead
+ 15. SetCIDnum -- use the function CALLERID(number) instead
+ 16. SetGroup -- use Set(GROUP=group) instead
+ 17. SetRDNIS -- use the function CALLERID(rdnis) instead
+ 18. Sql_postgres -- was deprecated in 1.2, now removed
+ 19. Txtcidname -- use the function TXTCIDNAME instead
+ * New Dialplan Functions:
+ 1. ARRAY()
+ 2. BASE_64_DECODE()
+ 3. BASE_64_ENCODE()
+ 4. CHANNEL()
+ 5. CURL()
+ 6. CUT()
+ 7. DB_DELETE()
+ 8. FILTER()
+ 9. GLOBAL()
+ 10. IFTIME()
+ 11. KEYPADHASH()
+ 12. ODBC()
+ 13. QUOTE()
+ 14. RAND()
+ 15. REALTIME()
+ 16. SHA1()
+ 17. SORT()
+ 18. SPRINTF()
+ 19. SQL_ESC()
+ 20. STAT()
+ 21. STRPTIME()
+ 22. AUDIOHOOK_INHERIT()
+ * Apps that have changes to their interface:
+ 1. Authenticate() -- optional maxdigits argument added.
+ 2. ChanSpy() -- new options:
+ o w -- Enable 'whisper' mode, so the spying channel can talk to...
+ o W -- Enable 'private whisper' mode, so the spying channel can...
+ 3. DBdel() -- now marked as DEPRECATED in favor of the DB_DELETE func
+ 4. Dial()
+ o New Option: O([x]) for Zaptel operator mode
+ o New Option: K/k parking via dtmf tones
+ 5. Dictate() -- optional filename argument added.
+ 6. Directory() -- new option: e - In addition to the name, also read the extension number...
+ 7. ForkCDR() -- new options:
+ o 'a' -- update answer time on new cdr
+ o 'A' -- Lock the orig CDR answer time against changes.
+ o 'D' -- Copy the disposition from the orig to the new CDR.
+ o 'd' -- clear the dstcannel field in the new CDR.
+ o 'e' -- set the end time of the original CDR.
+ o 'R' -- do NOT reset the new CDR.
+ o 's' -- Add/change var in orig CDR.
+ o 'T' -- Force ast_cdr_end, answer to obey LOCKED flag for the orig. CDR.
+ -- ast_cdr_setvar will be forced also (used by the CDR() func in write mode)
+ 8. Meetme() -- new options:
+ o 'I' -- announce user join/leave without review
+ o 'l' -- set listen only mode (Listen only, no talking)
+ o 'o' -- set talker optimization - treats talkers who aren't speaking as...
+ o '1' -- do not play message when first person enters
+ 9. MeetmeAdmin() -- new options:
+ o 'r' -- Reset one user's volume settings
+ o 'R' -- Reset all users volume settings
+ o 's' -- Lower entire conference speaking volume
+ o 'S' -- Raise entire conference speaking volume
+ o 't' -- Lower one user's talk volume
+ o 'T' -- Lower all users talk volume
+ o 'u' -- Lower one user's listen volume
+ o 'U' -- Lower all users listen volume
+ o 'v' -- Lower entire conference listening volume
+ o 'V' -- Raise entire conference listening volume
+ 10. OSPFinish() : now also can return ERROR result.
+ 11. OSPLookup() : Sets more variables, also now returns ERROR result.
+ 12. Page() -- New option: r - record the page into a file (see 'r' for app_meetme)
+ 13. Pickup() -- multiple extensions, PICKUPMARK; read the description!
+ 14. Queue()
+ o New Argument: AGI
+ o New option: i
+ 15. Random() -- is now deprecated in 1.4
+ 16. Read() -- replace 'skip' and 'noanswer' options with 's', 'n', add 'i' option.
+ 17. Record() -- New option: 'x' : ignore all terminator keys (DTMF) and keep recording until hangup
+ 18. UserEvent() -- slight change in behavior. Read the description.
+ 19. VoiceMailMain() -- new a(#) option, goes to folder # directly.
+ 20. WaitForSilence() -- new optional 3rd arg, time delay before returning.
+ * Functions that have changes to their interfaces:
+ 1. CDR -- new options: u and s
+ 2. LANGUAGE -- Deprecated. Use CHANNEL(language) instead.
+ 3. MUSICCLASS -- Deprecated. Use CHANNEL(musicclass) instead.
+ * Configuration File Changes:
+ 1. NEW config files:
+ 1. amd.conf -- Answering Machine Detection parameters
+ 2. followme.conf -- parameters for the findme/followme call forwarding
+ 3. func_odbc.conf -- define sql access functions here
+ 4. gtalk.conf -- how to handle gtalk protocol calls
+ 5. h323.conf -- h323 configuration
+ 6. http.conf -- config for the builtin mini-http server in asterisk
+ 7. jabber.conf -- jabber interface
+ 8. jingle.conf -- jingle protocol interface config
+ 10. res_snmp.conf -- to enable snmp in asterisk, and define full/sub agent status
+ 11. say.conf -- define per-language rules for numbers, dates, etc.
+ 12. skinny.conf -- for those special skinny phones you want to use...
+ 13. sla.conf -- Shared Line Appearance config
+ 14. smdi.conf -- SMDI messaging config
+ 15. udptl.conf -- T38's udptl transport config
+ 16. users.conf -- user config
+ 2. Changes to Existing Config files:
+ 1. In General:
+ o Jitterbuffer support added to several channels. Usually adds these variables to a config file:
+ 1. jbenable
+ 2. jbmaxsize
+ 3. jbresyncthreshold
+ 4. jbimpl
+ 5. jblog
+ o MusicOnHold upgrade introduces two new variables:
+ 1. mohinterpret
+ 2. mohsuggest
+ 2. agents.conf
+ o multiplelogin variable added
+ o maxlogintries variable added
+ o autologoffunavail variable added
+ o endcall variable added
+ o goodbye variable added
+ o createlink variable REMOVED
+ 3. alsa.conf
+ o mohinterpret variable added
+ o Jitterbuffer variables added
+ 4. cdr.conf
+ o endbeforehexten variable added
+ o sections for csv and radius added, with variables usegmtime, loguniqueid,
+ loguserfield, and radiuscfg variables.
+ 5. cdr_tds.conf
+ o table variable added
+ 6. extensions.ael
+ o Many upgrades. See the info at http://www.voip-info.org/wiki/view/Asterisk+AEL2
+ 7. extensions.conf
+ o autofallthru now set to "yes" by default
+ o userscontext variable added
+ o added info/examples on paging and hints.
+ 8. features.conf
+ o parkedplay variable added (who to beep at)
+ o parkedmusicclass
+ o atxfernoanswertimeout variable added
+ o parkcall variable added (one step parking)
+ o improved documentation for dynamic feature declarations!
+ o added parkedcallltransfers option to control builtin transfers with parking
+ 9. iax.conf
+ o adsi variable added
+ o mohinterpret variable added
+ o mohsuggest variable added
+ o jitterbuffer updates
+ o iaxthreadcount variable added
+ o iaxmaxthreadcount variable added
+ o the way to specify TOS has changed.
+ o mailboxdetail variable has been REMOVED.
+ 10. indications.conf
+ o [bg] entry added (Bulgaria).
+ o [il] entry added (Israel)
+ o [in] entry added (India)
+ o [jp] entry added (Japan)
+ o [my] entry added (Malaysia)
+ o [th] entry added (Thailand)
+ 11. manager.conf
+ o webenabled variable added
+ o httptimeout variable added
+ o timestampevents variable added
+ 12. mgcp.conf
+ o Jitterbuffer support added
+ 13. misdn.conf
+ o l1watcher_timeout variable added
+ o pp_l2_check variable added
+ o echocancelwhenbridged variable added
+ o echotraining variable added
+ o max_incoming variable added
+ o max_outgoing variable added
+ 14. modules.conf
+ o a comment for preloading res_speech.so is added
+ o mention of global symbols is removed
+ o obsolesced entries for chan_modem_* and app_intercom have been removed
+ 15. musiconhold.conf
+ o the default is now to do native moh from /var/lib/asterisk/moh
+ 16. osp.conf
+ o authpolicy variable added
+ 17. oss.conf
+ o debug variable added
+ o device variable added
+ o mixer variable added
+ o boost variable added
+ o callerid variable added
+ o autohangup variable added
+ o queuesize variable added
+ o frags variable added
+ o JitterBuffer support
+ o sections to define alternate sound cards
+ 18. queues.conf
+ o autofill variable added
+ o monitor-type variable added
+ o musiconhold is now musicclass, with a difference in interpretation
+ o autofill variable added
+ o autopause variable added
+ o setinterfacevar variable added
+ o ringinuse variable added
+ 19. res_odbc.conf
+ o pooling variable added
+ 20. rpt.conf
+ o duplex variable added
+ o tailmessagetime variable added
+ o tailsquashedtime variable added
+ o tailmessages variable added
+ 21. rtp.conf
+ o rtcpinterval varaible added
+ 22. sip.conf
+ o allowguest variable can't be set to 'osp'
+ o allowoverlap variable added
+ o allowtransfer variable added
+ o limitonpeer variable added
+ o directrtpsetup variable added
+ o buggymwi variable added
+ o ospauth variable REMOVED
+ o notifyhold variable added
+ o autoframing variable added
+ o tos variable REMOVED
+ o tos_sip variable added
+ o tos_audio variable added
+ o tos_video variable added
+ o minexpiry variable added
+ o t1min variable added
+ o musicclass variable REMOVED
+ o mohinterpret variable added
+ o mohsuggest variable added
+ o allowsubscribe variable added
+ o videosupport variable added
+ o maxcallbitrate variable added
+ o g726nonstandard variable added
+ o dumphistory variable added
+ o t38pt_udptl variable added
+ o t38pt_rtp variable added
+ o t38pt_tcp variable added
+ o rfc2833compensate variable added
+ o matchexterniplocally variable added
+ o canreinvite variable can also now be set to 'nonat'
+ o rtsavesysname variable added
+ o JitterBuffer support added
+ o t38pt_usertpsource variable added
+ o regcontext variable can contains multiple contexts separated by an '&'
+ 23. skinny.conf
+ o port variable renamed to bindport
+ o JitterBuffer support added
+ o model variable REMOVED
+ o mohinterpret variable added
+ o mohsuggest variable added
+ o speeddial variable added
+ o addon variable added
+ 24. voicemail.conf
+ o userscontext variable added
+ o smdiport variable added
+ o attachfmt variable added
+ o volgain variable added
+ o tempgreetwarn variable added
+ 25. zapata.conf
+ o pritimer variable has improved documentation
+ o New signalling method: fgccama
+ o New signalling method: fgccamamf
+ o outsignalling variable added
+ o distinctiveringaftercid variable added
+ o cidsignalling now also accepts v23_jp, and smdi
+ o usesmdi variable added
+ o smdiport variable added
+ o mohinterpret variable added
+ o mohsuggest variable added
+ o JitterBuffer support added
+ * Removed Codecs/Channels:
+ 1. codec_g723 was removed because the actual codec implementation it was designed to use is not distributable
+ 2. chan_modem_* and related modules are gone because the kernel support for those interfaces is old, buggy and unsupported
+ * New Utils:
+ 1. aelparse -- compile .ael files outside of asterisk
+ * New manager events:
+ 1. OriginateResponse event comes to replace OriginateSuccess and OriginateFailure
+ * iLBC source code no longer included (see UPGRADE.txt for details)
+ * New CLI command "pri show version" that shows the current version of libpri
+ that the library was built against (requires a version of libpri since this API
+ feature was added).
diff --git a/1.4.23-rc4/COPYING b/1.4.23-rc4/COPYING
new file mode 100644
index 000000000..aa2ebac66
--- /dev/null
+++ b/1.4.23-rc4/COPYING
@@ -0,0 +1,341 @@
+
+ GNU GENERAL PUBLIC LICENSE
+ Version 2, June 1991
+
+ Copyright (C) 1989, 1991 Free Software Foundation, Inc.
+ 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ Everyone is permitted to copy and distribute verbatim copies
+ of this license document, but changing it is not allowed.
+
+ Preamble
+
+ The licenses for most software are designed to take away your
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+POSSIBILITY OF SUCH DAMAGES.
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+
+ How to Apply These Terms to Your New Programs
+
+ If you develop a new program, and you want it to be of the greatest
+possible use to the public, the best way to achieve this is to make it
+free software which everyone can redistribute and change under these terms.
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+ To do so, attach the following notices to the program. It is safest
+to attach them to the start of each source file to most effectively
+convey the exclusion of warranty; and each file should have at least
+the "copyright" line and a pointer to where the full notice is found.
+
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+ Copyright (C) 19yy <name of author>
+
+ This program is free software; you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
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+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
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+ Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+
+
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+ Gnomovision version 69, Copyright (C) 19yy name of author
+ Gnomovision comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
+ This is free software, and you are welcome to redistribute it
+ under certain conditions; type `show c' for details.
+
+The hypothetical commands `show w' and `show c' should show the appropriate
+parts of the General Public License. Of course, the commands you use may
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+mouse-clicks or menu items--whatever suits your program.
+
+You should also get your employer (if you work as a programmer) or your
+school, if any, to sign a "copyright disclaimer" for the program, if
+necessary. Here is a sample; alter the names:
+
+ Yoyodyne, Inc., hereby disclaims all copyright interest in the program
+ `Gnomovision' (which makes passes at compilers) written by James Hacker.
+
+ <signature of Ty Coon>, 1 April 1989
+ Ty Coon, President of Vice
+
+This General Public License does not permit incorporating your program into
+proprietary programs. If your program is a subroutine library, you may
+consider it more useful to permit linking proprietary applications with the
+library. If this is what you want to do, use the GNU Library General
+Public License instead of this License.
diff --git a/1.4.23-rc4/CREDITS b/1.4.23-rc4/CREDITS
new file mode 100644
index 000000000..a51638214
--- /dev/null
+++ b/1.4.23-rc4/CREDITS
@@ -0,0 +1,215 @@
+
+=== DEVELOPMENT SUPPORT ===
+We'd like to thank the following companies for helping fund development of
+Asterisk:
+
+Pilosoft, Inc. - for supporting ADSI development in Asterisk
+
+Asterlink, Inc. - for supporting broad Asterisk development
+
+GFS - for supporting ALSA development
+
+Telesthetic - for supporting SIP development
+
+Christos Ricudis - for substantial code contributions
+
+nic.at - ENUM support in Asterisk
+
+Paul Bagyenda, Digital Solutions - for initial Voicetronix driver development
+
+=== WISHLIST CONTRIBUTERS ===
+Jeremy McNamara - SpeeX support
+Nick Seraphin - RDNIS support
+Gary - Phonejack ADSI (in progress)
+Wasim - Hangup detect
+
+=== HARDWARE DONORS ===
+* Thanks to QuickNet Technologies for their donation of an Internet
+PhoneJack and Linejack card to the project. (http://www.quicknet.net)
+
+* Thanks to VoipSupply for their donation of Sipura ATAs to the project for
+T.38 testing. (http://www.voipsupply.com)
+
+* Thanks to Grandstream for their donation of ATAs to the project for
+T.38 testing. (http://www.grandstream.com)
+
+=== MISCELLANEOUS PATCHES ===
+Jim Dixon - Zapata Telephony and app_rpt
+ http://www.zapatatelephony.org/app_rpt.html
+
+Russell Bryant - Asterisk 1.0 maintainer and misc. enhancements
+ russelb@clemson.edu
+
+Anthony Minessale II - Countless big and small fixes, and relentless forward push
+ ChanSpy, ForkCDR, ControlPlayback, While/EndWhile, DumpChan, Dictate,
+ MacroIf, ExecIf, ExecIfTime, RetryDial, MixMonitor applications; many realtime
+ concepts and implementation pieces, including res_config_odbc; format_slin;
+ cdr_custom; several features in Dial including L(), G() and enhancements to
+ M() and D(); several CDR enhancements including CDR variables; attended
+ transfer; one touch record; native MOH; manager eventmask; command line '-t'
+ flag to allow recording/voicemail on nfs shares; #exec command and multiline
+ comments in config files; setvar in iax and sip configs.
+ anthmct@yahoo.com http://www.asterlink.com
+
+James Golovich - Innumerable contributions
+ You can find him and asterisk-perl at http://asterisk.gnuinter.net
+
+Andre Bierwirth - Extension hints and status
+
+Oliver Daudey - ISDN4Linux fixes
+
+Pauline Middelink - ISDN4Linux patches and some general patches.
+ She can be found at http://www.polyware.nl/~middelink/En/
+
+Jean-Denis Girard - Various contributions from the South Pacific Islands
+ jd-girard@esoft.pf http://www.esoft.pf
+
+William Jordan / Vonage - MySQL enhancements to Voicemail
+ wjordan@vonage.com
+
+Jac Kersing - Various fixes
+
+Steven Critchfield - Seek and Trunc functions for playback and recording
+ critch@basesys.com
+
+Jefferson Noxon - app_lookupcidname, app_db, and various other contributions
+
+Klaus-Peter Junghanns - in-band DTMF on SIP and MGCP
+
+Ross Finlayson - Dynamic RTP payload support
+
+Mahmut Fettahlioglu - Audio recording, music-on-hold changes, alaw file
+ format, and various fixes. Can be contacted at mahmut@oa.com.au
+
+James Dennis - Cisco SIP compatibility patches to work with SIP service
+ providers. Can be contacted at asterisk@jdennis.net
+
+Tilghman Lesher - ast_localtime(); ast_say_date_with_format();
+ GotoIfTime, Random, SayUnixTime, HasNewVoicemail applications;
+ CUT, SORT, EVAL, CURL, FIELDQTY, STRFTIME, QUEUEAGENT* functions;
+ and other innumerable bug fixes. http://asterisk.drunkcoder.com/
+
+Jayson Vantuyl - Manager protocol changes, various other bugs.
+ jvantuyl@computingedge.net
+
+Thorsten Lockert - OpenBSD, FreeBSD ports, making MacOS X port run on 10.3,
+ dialplan include verification, route lookup on OpenBSD, SNMP agent
+ support (res_snmp), various other bugs. tholo@sigmasoft.com
+
+Brian West - ODBC support and Bug Marshaling
+
+Josh Roberson - chan_zap reload support, Advanced Voicemail Features, other misc. patches,
+ and Bug Marshalling. - josh@asteriasgi.com, http://www.asteriasgi.com
+
+William Waites - syslog support, SIP NAT traversal for SIP-UA. ww@styx.org
+
+Rich Murphey - Porting to FreeBSD, NetBSD, OpenBSD, and Darwin.
+ rich@whiteoaklabs.com http://whiteoaklabs.com
+
+Simon Lockhart - Porting to Solaris (based on work of Logan ???)
+ simon@slimey.org
+
+Olle E. Johansson - SIP RFC compliance, documentation and testing, testing, testing
+ oej@edvina.net, http://edvina.net
+
+Steve Kann - new jitter buffer for IAX2
+ stevek@stevek.com
+
+Constantine Filin - major contributions to the Asterisk Realtime Architecture
+
+Steve Murphy - privacy support, $[ ] parser upgrade, AEL2 parser upgrade
+
+Claude Patry - bug fixes, feature enhancements, and bug marshalling
+ cpatry@gmail.com
+
+Miroslav Nachev, miro@space-comm.com COSMOS Software Enterprises, Ltd.
+ - for Variable for No Answer Timeout for Attended Transfer
+
+Slav Klenov & Vanheuverzwijn Joachim - development of the generic jitterbuffer
+ Securax Ltd. info@securax.be
+
+Roy Sigurd Karlsbakk - providing funding for generic jitterbuffer development
+ roy@karlsbakk.net, Briiz Telecom AS
+
+Voop A/S, Nuvio Inc, Inotel S.A and Foniris Telecom A/S - funding for rewrite of SIP transfers
+
+Philippe Sultan - RADIUS CDR module
+ INRIA, http://www.inria.fr/
+
+John Martin, Aupix - Improved video support in the SIP channel
+
+Steve Underwood - Provided T.38 pass through support.
+
+George Konstantoulakis - Support for Greek in voicemail added by InAccess Networks (work funded by HOL, www.hol.gr) gkon@inaccessnetworks.com
+
+Daniel Nylander - Support for Swedish and Norwegian languages in voicemail. http://www.danielnylander.se/
+
+Stojan Sljivic - An option for maximum number of messsages per mailbox in voicemail. Also an issue with voicemail synchronization has been fixed. GDS Partners www.gdspartners.com . stojan.sljivic@gdspartners.com
+
+Bartosz Supczinski - Support for Polish added by DIR (www.dir.pl) Bartosz.Supczinski@dir.pl
+
+James Rothenberger - Support for IMAP storage integration added by OneBizTone LLC Work funded by University of Pennsylvania jar@onebiztone.com
+
+Paul Cadach - Bringing chan_h323 up to date, bug fixes, and more!
+
+=== OTHER CONTRIBUTIONS ===
+John Todd - Monkey sounds and associated teletorture prompt
+Michael Jerris - bug marshaling
+Leif Madsen, Jared Smith and Jim van Meggelen - the Asterisk book
+ available under a Creative Commons License at http://www.asteriskdocs.org
+Brian M. Clapper - poll.c emulation
+ This product includes software developed by Brian M. Clapper <bmc@clapper.org>
+
+=== HOLD MUSIC ===
+Music provided by www.freeplaymusic.com
+
+=== OTHER SOURCE CODE IN ASTERISK ===
+Asterisk uses libedit, the lightweight readline replacement from NetBSD.
+The cdr_radius module uses libradiusclient-ng, which is also from NetBSD.
+They are BSD-licensed and require the following statement:
+
+ This product includes software developed by the NetBSD
+ Foundation, Inc. and its contributors.
+
+Digium did not implement the codecs in Asterisk. Here is the copyright on the
+GSM source:
+
+Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
+Technische Universitaet Berlin
+
+Any use of this software is permitted provided that this notice is not
+removed and that neither the authors nor the Technische Universitaet Berlin
+are deemed to have made any representations as to the suitability of this
+software for any purpose nor are held responsible for any defects of
+this software. THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.
+
+As a matter of courtesy, the authors request to be informed about uses
+this software has found, about bugs in this software, and about any
+improvements that may be of general interest.
+
+Berlin, 28.11.1994
+Jutta Degener
+Carsten Bormann
+
+And the copyright on the ADPCM source:
+
+Copyright 1992 by Stichting Mathematisch Centrum, Amsterdam, The
+Netherlands.
+
+ All Rights Reserved
+
+Permission to use, copy, modify, and distribute this software and its
+documentation for any purpose and without fee is hereby granted,
+provided that the above copyright notice appear in all copies and that
+both that copyright notice and this permission notice appear in
+supporting documentation, and that the names of Stichting Mathematisch
+Centrum or CWI not be used in advertising or publicity pertaining to
+distribution of the software without specific, written prior permission.
+
+STICHTING MATHEMATISCH CENTRUM DISCLAIMS ALL WARRANTIES WITH REGARD TO
+THIS SOFTWARE, INCLUDING ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND
+FITNESS, IN NO EVENT SHALL STICHTING MATHEMATISCH CENTRUM BE LIABLE
+FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
+WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
+ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT
+OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
diff --git a/1.4.23-rc4/ChangeLog b/1.4.23-rc4/ChangeLog
new file mode 100644
index 000000000..8b7482191
--- /dev/null
+++ b/1.4.23-rc4/ChangeLog
@@ -0,0 +1,22076 @@
+2009-01-16 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.23-rc4 released.
+
+2009-01-16 00:19 +0000 [r168745] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_ael.c: This patch fixes a problem where a goto (or jump,
+ in this case) fails a consistency check because it can't find a
+ matching extension. The problem was a missing instruction to end
+ the range notation in the code where it converts the pattern into
+ a regex and uses the regex code to determine the match. I tested
+ using the AEL code the user supplied, and now, the consistency
+ check passes. (closes issue #14141) Reported by: dimas
+
+2009-01-15 18:43 +0000 [r168721] Olle Johansson <oej@edvina.net>
+
+ * configs/extconfig.conf.sample: Meetme actually has realtime but
+ wasn't documented
+
+2009-01-15 18:22 +0000 [r168716] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c: Convert call to park_call_full to
+ masq_park_call_announce Since we removed the AST_PBX_KEEPALIVE
+ return value, we need to use masqueraded parking, otherwise we
+ will try to call ast_hangup() in __pbx_run() and in
+ do_parking_thread() and then promptly crash. (closes issue
+ #14215) Reported by: waverly360 Tested by: otherwiseguy (closes
+ issue #14228) Reported by: kobaz Tested by: otherwiseguy
+
+2009-01-15 01:20 +0000 [r168633] Tilghman Lesher <tlesher@digium.com>
+
+ * /: Blocked revision 168632 from /branches/1.2: 1.2 regression on
+ security fix AST-2009-001 (Closes issue #14238)
+
+2009-01-15 00:11 +0000 [r168628] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix some crashes from bad datastore handling in
+ app_queue.c * The queue_transfer_fixup function was searching for
+ and removing the datastore from the incorrect channel, so this
+ was fixed. * Most datastore operations regarding the
+ queue_transfer datastore were being done without the channel
+ locked, so proper channel locking was added, too. (closes issue
+ #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by
+ putnopvut (license 60) Tested by: ZX81, festr
+
+2009-01-14 21:48 +0000 [r168622] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn/isdn_lib.c: * Fixed create_process() allocation of
+ process ID values. The allocated process IDs could overflow their
+ respective NT and TE fields. Affects outgoing calls.
+
+2009-01-14 20:52 +0000 [r168614] Sean Bright <sean.bright@gmail.com>
+
+ * contrib/scripts/autosupport: Update autosupport script to supply
+ info for both Zaptel and DAHDI in 1.4 and be sure to run
+ dahdi_test in 1.6.x and trunk instead of zttest. (closes issue
+ #14132) Reported by: dsedivec Patches:
+ asterisk-1.4-autosupport.patch uploaded by dsedivec (license 638)
+ asterisk-trunk-autosupport.patch uploaded by dsedivec (license
+ 638)
+
+2009-01-14 19:34 +0000 [r168608] Steve Murphy <murf@digium.com>
+
+ * apps/app_page.c: app_page was failing to compile in dev-mode on
+ my gcc-4.2.4 system. This change gets rid of the warning.
+
+2009-01-14 19:02 +0000 [r168603] Tilghman Lesher <tlesher@digium.com>
+
+ * main/udptl.c: Don't read into a buffer without first checking if
+ a value is beyond the end. (closes issue #13600) Reported by:
+ atis Patches: 20090106__bug13600.diff.txt uploaded by Corydon76
+ (license 14) Tested by: atis
+
+2009-01-14 16:19 +0000 [r168598] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Fix a logic error I found while searching
+ through chan_agent.c I found that the allow_multiple_logins
+ function would never return 0 due to an incorrect comparison
+ being used when traversing the list of agents. While I was
+ modifying this function, I also did a little bit of coding
+ guidelines cleanup, too.
+
+2009-01-14 01:27 +0000 [r168593] Terry Wilson <twilson@digium.com>
+
+ * apps/app_page.c: Don't overflow when paging more than 128
+ extensions The number of available slots for calls in app_page
+ was hardcoded to 128. Proper bounds checking was not in place to
+ enforce this limit, so if more than 128 extensions were passed to
+ the Page() app, Asterisk would crash. This patch instead
+ dynamically allocates memory for the ast_dial structures and
+ removes the (non-functional) arbitrary limit. This issue would
+ have special importance to anyone who is dynamically creating the
+ argument passed to the Page application and allowing more than
+ 128 extensions to be added by an outside user via some external
+ interface. The patch posted by a_villacis was slightly modified
+ for some coding guidelines and other cleanups. Thanks,
+ a_villacis! (closes issue #14217) Reported by: a_villacis
+ Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch
+ uploaded by a (license 660) Tested by: otherwiseguy
+
+2009-01-13 19:13 +0000 [r168561] Russell Bryant <russell@digium.com>
+
+ * main/indications.c, main/channel.c, apps/app_read.c,
+ channels/chan_misdn.c, funcs/func_channel.c,
+ include/asterisk/indications.h, apps/app_disa.c, main/app.c,
+ res/snmp/agent.c, include/asterisk/channel.h,
+ res/res_indications.c: Revert unnecessary indications API change
+ from rev 122314
+
+2009-01-13 18:34 +0000 [r168551] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Don't pass a value with a side effect to a
+ macro (closes issue #14176) Reported by: paraeco Patches:
+ chan_sip.c.diff uploaded by paraeco (license 658)
+
+2009-01-13 17:48 +0000 [r168546] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_logic.c: If either conditional is NULL, don't try
+ copying it. (closes issue #14226) Reported by: caspy Patches:
+ 20090113__bug14226.diff.txt uploaded by Corydon76 (license 14)
+
+2009-01-12 21:42 +0000 [r168507-168516] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_agi.c: (closes issue #13881) Reported by: hoowa Update
+ the app CDR field for AGI commands that are not executing an
+ application via "exec".
+
+ * channels/chan_agent.c: (closes issue #12269) Reported by: IgorG
+ Tested by: denisgalvao This gits rid of the notion of an
+ owning_app allowing the request and hangup to be initiated by
+ different threads. Originating from an active agent channel
+ requires this. The implementation primarily changes __login_exec
+ to wait on a condition variable rather than a lock. Review:
+ http://reviewboard.digium.com/r/35/
+
+2009-01-12 14:58 +0000 [r168482] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: I am reverting the fix made in revision
+ 168128 (and its upward merges) after being contacted by Olle
+ Johansson and being shown how this fix is incorrect. Thanks to
+ Olle for clearing this up for me.
+
+2009-01-12 14:57 +0000 [r168480] Russell Bryant <russell@digium.com>
+
+ * configs/indications.conf.sample: s/ringdance/ringcadence/ for
+ Bulgaria
+
+2009-01-10 20:47 +0000 [r168267-168382] Kevin P. Fleming <kpfleming@digium.com>
+
+ * README: small commit to test new server
+
+ * README: small commit to test new server
+
+ * sounds/Makefile: update to use new sound file packages that
+ include license files
+
+2009-01-09 22:14 +0000 [r168198] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Make this compile for mvanbaak
+
+2009-01-09 21:28 +0000 [r168191] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: * Fix for JIRA AST-175/ABE-1757 *
+ Miscellaneous doxygen comments added.
+
+2009-01-09 20:08 +0000 [r168128] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add check_via calls to more request handlers
+ INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests were not
+ checking the topmost Via to determine where to send the response.
+ Adding check_via calls to those request handlers solves this.
+ (closes issue #13071) Reported by: baron Patches: check_via.patch
+ uploaded by baron (license 531) Tested by: baron
+
+2009-01-08 22:08 +0000 [r167840] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: Don't truncate database results at 255 chars.
+ (closes issue #14069) Reported by: evandro Patches:
+ 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14)
+
+2009-01-08 17:24 +0000 [r167620-167714] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: remove an unnecessary argument to
+ queue_request()
+
+ * channels/chan_sip.c: When a SIP request or response arrives for a
+ dialog with an associated Asterisk channel, and the lock on that
+ channel cannot be obtained because it is held by another thread,
+ instead of dropping the request/response, queue it for later
+ processing when the channel lock becomes available.
+ http://reviewboard.digium.com/r/117/
+
+2009-01-07 22:35 +0000 [r167432-167566] Russell Bryant <russell@digium.com>
+
+ * main/file.c: Fix the last couple of places where free() was
+ improperly used directly.
+
+ * main/file.c: Don't fclose() the file early, the filestream
+ destructor will handle it.
+
+ * main/file.c: Only try to close the file if one was actually
+ opened
+
+ * main/file.c: Don't use free() directly. This caused a crash since
+ ast_filestream is now an ao2 object. Reported by JunK-Y on IRC,
+ #asterisk-dev
+
+ * main/indications.c: Treat an empty string the same way as a NULL
+ country argument. In passing, simplify the handling of returning
+ a default tone zone.
+
+2009-01-06 21:35 +0000 [r167299] Mark Michelson <mmichelson@digium.com>
+
+ * main/db.c: Use the correct variable when creating the format
+ string (closes issue #14177) Reported by: nic_bellamy Patches:
+ asterisk-trunk-svn-r167242-ast_db_gettree.patch uploaded by nic
+ (license 299)
+
+2009-01-06 20:48 +0000 [r167260] Tilghman Lesher <tlesher@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 167259 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r167259 | tilghman | 2009-01-06 14:44:03 -0600 (Tue, 06
+ Jan 2009) | 2 lines Security fix AST-2009-001. ........
+
+2009-01-05 16:51 +0000 [r167179] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: A couple of changes to T.38 SDP attribute
+ handling There are some boolean attributes for T.38 such as
+ T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
+ T38FaxTranscodingJBIG. By simply being present, we should treat
+ these as a "true" value. The current code, however, was requiring
+ a 1 or 0 as the value of the attribute in order to parse it. This
+ is due to the fact that there are some T.38 endpoints and
+ gateways that also transmit this information incorrectly. This
+ patch follows the "be liberal in what you accept and strict in
+ what you send" philosophy by accepting both the correctly- and
+ incorrectly-formatted attributes, but only sending information as
+ it is supposed to be sent. It was also discovered that a
+ particular type of T.38 gateway sends some non-standard T.38 SDP
+ attributes. Instead of using T38FaxMaxDatagram and T38MaxBitRate,
+ it used T38MaxDatagram and T38FaxMaxRate respectively. We now
+ will properly accept these attributes as well. Note that there
+ are a lot of patches cited in the below commit message template.
+ This is because the person who submitted these patches is an
+ awesome person and wrote 1.4, 1.6.0, and 1.6.1 variants. (closes
+ issue #13976) Reported by: linulin Patches:
+ chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
+ chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded
+ by arcivanov (license 648)
+ chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov
+ (license 648) Tested by: arcivanov
+
+2009-01-01 00:01 +0000 [r166953-167095] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_alsa.c: Repeat attempts to write when we receive
+ -EAGAIN from the driver, as detailed in the ALSA sample code (see
+ http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html#a32)
+ Reported by: Jerry Geis (via the -users list) Fixed by: me
+ (license 14)
+
+ * channels/chan_local.c: Also inherit the musiconhold class.
+ (Closes #14153) Reported by: Jerry Geis, via the users list.
+ Patch by: me (license 14)
+
+2008-12-28 15:13 +0000 [r166772] Russell Bryant <russell@digium.com>
+
+ * channels/misdn_config.c: Use strncat() instead of an sprintf() in
+ which source and target buffers overlap
+ http://lists.digium.com/pipermail/asterisk-dev/2008-December/035919.html
+
+2008-12-23 15:35 +0000 [r166592] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, channels/chan_iax2.c: Compile, even if both
+ DAHDI and Zaptel are not installed. (Closes issue #14120)
+
+2008-12-23 15:16 +0000 [r166568] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Fix a crash resulting from a datastore with
+ inheritance but no duplicate callback The fix for this is to
+ simply set the newly created datastore's data pointer to NULL if
+ it is inherited but has no duplicate callback. (closes issue
+ #14113) Reported by: francesco_r Patches: 14113.patch uploaded by
+ putnopvut (license 60) Tested by: francesco_r
+
+2008-12-23 04:05 +0000 [r166509] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c: Use the integer form of condition for integer
+ comparisons. (closes issue #14127) Reported by: andrew
+
+2008-12-22 20:56 +0000 [r166380] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_dahdi.c: Fix a deadlock relating to channel locks
+ and autoservice It has been discovered that if a channel is
+ locked prior to a call to ast_autoservice_stop, then it is likely
+ that a deadlock will occur. The reason is that the call to
+ ast_autoservice_stop has a check built into it to be sure that
+ the thread running autoservice is not currently trying to
+ manipulate the channel we are about to pull out of autoservice.
+ The autoservice thread, however, cannot advance beyond where it
+ currently is, though, because it is trying to acquire the lock of
+ the channel for which autoservice is attempting to be stopped.
+ The gist of all this is that a channel MUST NOT be locked when
+ attempting to stop autoservice on the channel. In this particular
+ case, the channel was locked by a call to ast_read. A call to
+ ast_exists_extension led to autoservice being started and stopped
+ due to the existence of dialplan switches. It may be that there
+ are future commits which handle the same symptoms but in a
+ different location, but based on my looks through the code, it is
+ very rare to see a construct such as this one. (closes issue
+ #14057) Reported by: rtrauntvein Patches: 14057v3.patch uploaded
+ by putnopvut (license 60) Tested by: rtrauntvein Review:
+ http://reviewboard.digium.com/r/107/
+
+2008-12-22 17:22 +0000 [r166262-166297] Russell Bryant <russell@digium.com>
+
+ * main/utils.c: Fix up timeout handling in ast_carefulwrite().
+
+ * include/asterisk/strings.h, res/res_musiconhold.c: Re-work ref
+ count handling of MoH classes using astobj2 to resolve crashes.
+ (closes issue #13566) Reported by: igorcarneiro Tested by:
+ russell Review: http://reviewboard.digium.com/r/106/
+
+2008-12-19 23:34 +0000 [r166157] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, funcs/func_audiohookinherit.c (added),
+ channels/chan_sip.c, include/asterisk/audiohook.h,
+ main/audiohook.c, CHANGES: Backport of AUDIOHOOK_INHERIT for
+ Asterisk 1.4 (closes issue #13538) Reported by: mbit Patches:
+ 13538.patch uploaded by putnopvut (license 60) Tested by:
+ putnopvut
+
+2008-12-19 22:30 +0000 [r166093] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, res/res_features.c, include/asterisk/pbx.h,
+ apps/app_queue.c: This merges the masqpark branch into 1.4 These
+ changes eliminate the need for (and use of) the KEEPALIVE return
+ code in res_features.c; There are other places that use this
+ result code for similar purposes at a higher level, these appear
+ to be left alone in 1.4, but attacked in trunk. The reason these
+ changes are being made in 1.4, is that parking ends a channel's
+ life, in some situations, and the code in the bridge (and some
+ other places), was not checking the result code properly, and
+ dereferencing the channel pointer, which could lead to memory
+ corruption and crashes. Calling the masq_park function eliminates
+ this danger in higher levels. A series of previous commits have
+ replaced some parking calls with masq_park, but this patch puts
+ them ALL to rest, (except one, purposely left alone because a
+ masquerade is done anyway), and gets rid of the code that tests
+ the KEEPALIVE result, and the NOHANGUP_PEER result codes. While
+ bug 13820 inspired this work, this patch does not solve all the
+ problems mentioned there. I have tested this patch (again) to
+ make sure I have not introduced regressions. Crashes that
+ occurred when a parked party hung up while the parking party was
+ listening to the numbers of the parking stall being assigned, is
+ eliminated. These are the cases where parking code may be
+ activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to
+ parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip
+ xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via
+ manager. The interesting testing cases for parking are: I. A
+ calls B, A parks B a. B hangs up while A is getting the numbers
+ announced. b. B hangs up after A gets the announcement, but
+ before the parking time expires c. B waits, time expires, A is
+ redialed, A answers, B and A are connected, after which, B hangs
+ up. d. C picks up B while still in parking lot. II. A calls B, B
+ parks A a. A hangs up while B is getting the numbers announced.
+ b. A hangs up after B gets the announcement, but before the
+ parking time expires c. A waits, time expires, B is redialed, B
+ answers, A and B are connected, after which, A hangs up. d. C
+ picks up A while still in parking lot. Testing this throroughly
+ involves acting all the permutations of I and II, in situations
+ 1,2,3, and 4. Since I added a few more changes (ALL references to
+ KEEPALIVE in the bridge code eliimated (I missed one earlier), I
+ retested most of the above cases, and no crashes. H-extension
+ weirdness. Current h-extension execution is not completely
+ correct for several of the cases. For the case where A calls B,
+ and A parks B, the 'h' exten is run on A's channel as soon as the
+ park is accomplished. This is expected behavior. But when A calls
+ B, and B parks A, this will be current behavior: After B parks A,
+ B is hung up by the system, and the 'h' (hangup) exten gets run,
+ but the channel mentioned will be a derivative of A's... Thus, if
+ A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on
+ channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info
+ will be those relating to Channel A. And, in the case where A is
+ reconnected to B after the park time expires, when both parties
+ hang up after the joyful reunion, no h-exten will be run at all.
+ In the case where C picks up A from the parking lot, when either
+ A or C hang up, the h-exten will be run for the C channel. CDR's
+ are a separate issue, and not addressed here. As to WHY this
+ strange behavior occurs, the answer lies in the procedure
+ followed to accomplish handing over the channel to the parking
+ manager thread. This procedure is called masquerading. In the
+ process, a duplicate copy of the channel is created, and most of
+ the active data is given to the new copy. The original channel
+ gets its name changed to XXX<ZOMBIE> and keeps the PBX
+ information for the sake of the original thread (preserving its
+ role as a call originator, if it had this role to begin with),
+ while the new channel is without this info and becomes a call
+ target (a "peer"). In this case, the parking lot manager thread
+ is handed the new (masqueraded) channel. It will not run an
+ h-exten on the channel if it hangs up while in the parking lot.
+ The h exten will be run on the original channel instead, in the
+ original thread, after the bridge completes. See bug 13820 for
+ our intentions as to how to clean up the h exten behavior.
+ Review: http://reviewboard.digium.com/r/29/
+
+2008-12-19 19:48 +0000 [r165991] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/dahdi_compat.h, main/asterisk.c, main/channel.c,
+ apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_meetme.c,
+ apps/app_dahdiscan.c, codecs/codec_dahdi.c,
+ res/res_musiconhold.c, channels/chan_iax2.c: (closes issue
+ #13480) Reported by: tzafrir Replace a bunch of if defined checks
+ for Zaptel/DAHDI through several new defines in dahdi_compat.h.
+ This removes a lot of code duplication. Example from bug: #ifdef
+ HAVE_ZAPTEL fd = open("/dev/zap/pseudo", O_RDWR); #else fd =
+ open("/dev/dahdi/pseudo", O_RDWR); #endif is replaced with: fd =
+ open(DAHDI_FILE_PSEUDO, O_RDRW);
+
+2008-12-19 15:03 +0000 [r165796-165889] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: Ensure that the chanspy datastore is fully
+ initialized. This patch resolved some random crash issues
+ observed by a user on a BSD system (closes issue #14111) Reported
+ by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281)
+
+ * main/utils.c: Make ast_carefulwrite() be more careful. This patch
+ handles some additional cases that could result in partial writes
+ to the file description. This was done to address complaints
+ about partial writes on AMI. (issue #13546) (more changes needed
+ to address potential problems in 1.6) Reported by: srt Tested by:
+ russell Review: http://reviewboard.digium.com/r/99/
+
+2008-12-18 21:14 +0000 [r165767] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Add mutexes around accesses to the IMAP
+ library interface. This prevents certain crashes, especially when
+ shared mailboxes are used. (closes issue #13653) Reported by:
+ howardwilkinson Patches:
+ asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by
+ howardwilkinson (license 590) Tested by: jpeeler
+
+2008-12-18 18:52 +0000 [r165661] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Set the process group ID on the MOH
+ process so that all children will get killed (closes issue
+ #14099) Reported by: caspy Patches:
+ res_musiconhold.c.patch.killpg.try2 uploaded by caspy (license
+ 645)
+
+2008-12-18 17:11 +0000 [r165537-165591] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Only care about a compatible codec for early bridging
+ if we are actually bridging to another channel. If we are not we
+ actually want to bring the audio back to us. (closes issue
+ #13545) Reported by: davidw
+
+ * apps/app_followme.c: Do not crash if we are not passed in a
+ followme id. (closes issue #14106) Reported by: ys Patches:
+ app_followme.c.2.diff uploaded by ys (license 281)
+
+2008-12-17 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.23-rc3 released.
+
+2008-12-17 21:14 +0000 [r165317] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_macro.c: Reverse the fix from issue #6176 and add proper
+ handling for that issue. (Closes issue #13962, closes issue
+ #13363) Fixed by myself (license 14)
+
+2008-12-17 20:51 +0000 [r164977-165255] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_meetme.c, apps/app_realtime.c, apps/app_directory.c,
+ apps/app_queue.c: Fix some memory leaks found while looking at
+ how realtime configs are handled. Also cleaned up some coding
+ guidelines violations in app_realtime.c, mostly related to
+ spacing
+
+ * channels/chan_sip.c: After looking through SIP registration code
+ most of the day, this is one of the few things I could find that
+ was just plain wrong. Even though it probably isn't possible for
+ it to happen, it seems weird to have code that checks if a
+ pointer is NULL and then immediately dereferences that pointer if
+ it was NULL.
+
+2008-12-16 21:38 +0000 [r164672-164881] Russell Bryant <russell@digium.com>
+
+ * main/utils.c: Fix an issue where DEBUG_THREADS may erroneously
+ report that a thread is exiting while holding a lock. If the last
+ lock attempt was a trylock, and it failed, it will still be in
+ the list of locks so that it can be reported. (closes issue
+ #13219) Reported by: pj
+
+ * apps/app_macro.c: Do not dereference the channel if
+ AST_PBX_KEEPALIVE has been returned. This is a bug I noticed
+ while looking at the code for app_macro. This return code means
+ that another thread has assumed ownership of the channel and it
+ can no longer be touched. (I hate this return code with a
+ passion, by the way.)
+
+ * main/manager.c: Add "restart gracefully" to the AMI blacklist of
+ CLI commands. "module unload" was already identified as a command
+ that can not be used from the AMI. "restart gracefully"
+ effectively unloads all modules, and will run in to the same
+ problems. (closes issue #13894) Reported by: kernelsensei
+
+ * include/asterisk/threadstorage.h, main/threadstorage.c: Fix
+ memory leak and invalid reporting issues with DEBUG_THREADLOCALS.
+ One issue was that the ast_mutex_* API was being used within the
+ context of the thread local data destructors. We would go off and
+ allocate more thread local data while the pthread lib was in the
+ middle of destroying it all. This led to a memory leak. Another
+ issue was an invalid argument being provided to the the
+ object_add API call. (closes issue #13678) Reported by: ys Tested
+ by: Russell
+
+ * channels/chan_sip.c: Fix a memory leak related to the use of the
+ "setvar" configuration option. The problem was that these
+ variables were being appended to the list of vars on the sip_pvt
+ every time a re-registration or re-subscription came in. Since
+ it's just a waste of memory to put them there unless the request
+ was an INVITE, then the fix is to check the request type before
+ copying the vars. (closes issue #14037) Reported by: marvinek
+ Tested by: russell
+
+2008-12-16 15:15 +0000 [r164634] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: I added a sentence to clarify why - and ' ' are
+ ignored in patterns as per bug 14076. Leif says he'll put some
+ stuff about it in the extensions.conf sample, etc.
+
+2008-12-16 14:28 +0000 [r164605] Russell Bryant <russell@digium.com>
+
+ * res/res_musiconhold.c: Don't try to change working directory if a
+ directory was not configured. (closes issue #14089) Reported by:
+ caspy
+
+2008-12-15 19:53 +0000 [r164416-164422] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/pbx.h: Add the deadlock note to
+ ast_spawn_extension as well
+
+ * include/asterisk/channel.h, include/asterisk/pbx.h: Add notes to
+ autoservice and pbx doxygen regarding a potential deadlock
+ scenario so that it is avoided in the future
+
+2008-12-15 18:11 +0000 [r164204-164350] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Do not try to unlock a non-existant channel
+ if the transfer fails. (closes issue #13800) Reported by: dwagner
+ Patches: asterisk-1.4.22-chan-sip-nullp.patch uploaded by tweety
+ (license 608)
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ include/asterisk/channel.h: Use autoconf logic to determine
+ whether the system has timersub or not. Do not blindly assume
+ Solaris does not. (closes issue #13838) Reported by: ano
+
+ * apps/app_dial.c: Can we try not to assign an unsigned int to -1?
+ (closes issue #14074) Reported by: wetwired
+
+2008-12-15 14:31 +0000 [r164201] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, res/res_features.c: Handle a case where a call
+ can be bridged to a channel that is still ringing. The issue that
+ was reported was about a case where a RINGING channel got
+ redirected to an extension to pick up a call from parking. Once
+ the parked call got taken out of parking, it heard silence until
+ the other side answered. Ideally, the caller that was parked
+ would get a ringing indication. This patch fixes this case so
+ that the caller receives ringback once it comes out of parking
+ until the other side answers. The fixes are: - Make sure we
+ remember that a channel was an outgoing channel when doing a
+ masquerade. This prevents an erroneous ast_answer() call on the
+ channel, which causes a bogus 200 OK to be sent in the case of
+ SIP. - Add some additional comments to explain related parts of
+ code. - Update the handling of the ast_channel visible_indication
+ field. Storing values that are not stateful is pointless. Control
+ frames that are events or commands should be ignored. - When a
+ bridge first starts, check to see if the peer channel needs to be
+ given ringing indication because the calling side is still
+ ringing. - Rework ast_indicate_data() a bit for the sake of
+ readability. (closes issue #13747) Reported by: davidw Tested by:
+ russell Review: http://reviewboard.digium.com/r/90/
+
+2008-12-13 23:22 +0000 [r164082] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c: Change the default calldurationlimit from the
+ special value 0 to -1, so we can better detect an exceptional
+ case. This follows on to the changes made in revision 156386.
+ Related to issue #13851. (closes issue #13974) Reported by:
+ paradise Patches: 20081208__bug13974.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: file, blitzrage, ZX81
+
+2008-12-12 22:20 +0000 [r163785] Russell Bryant <russell@digium.com>
+
+ * /: Set the reviewboard:url property on 1.4, as well
+
+2008-12-12 22:03 +0000 [r163761] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, main/editline/read.c: Simple fix for Ctrl-C not
+ immediately exiting Asterisk, but also add a pointer inside
+ editline to look back to asterisk.c, so others don't spend as
+ much time as I did looking (in the wrong place) for the
+ appropriate function. Reported by: ZX81, via the #asterisk-users
+ channel Fixed by: me (license 14)
+
+2008-12-12 14:40 +0000 [r163448-163511] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_dundi.c: Specify uint32_t for variables storing a CRC32
+ so that it is actually 32 bits on 64-bit machines, as well.
+ (inspired by issue #13879)
+
+ * main/channel.c, main/autoservice.c, include/asterisk/channel.h:
+ Resolve issues that could cause DTMF to be processed out of
+ order. These changes come from team/russell/issue_12658 1) Change
+ autoservice to put digits on the head of the channel's frame
+ readq instead of the tail. If there were frames on the readq that
+ autoservice had not yet read, the previous code would have
+ resulted in out of order processing. This required a new API call
+ to queue a frame to the head of the queue instead of the tail. 2)
+ Change up the processing of DTMF in ast_read(). Some of the
+ problems were the result of having two sources of pending DTMF
+ frames. There was the dtmfq and the more generic readq. Both were
+ used for pending DTMF in various scenarios. Simplifying things to
+ only use the frame readq avoids some of the problems. 3) Fix a
+ bug where a DTMF END frame could get passed through when it
+ shouldn't have. If code set END_DTMF_ONLY in the middle of digit
+ emulation, and a digit arrived before emulation was complete,
+ digits would get processed out of order. (closes issue #12658)
+ Reported by: dimas Tested by: russell, file Review:
+ http://reviewboard.digium.com/r/85/
+
+2008-12-11 23:35 +0000 [r163383] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: When a Ctrl-C or Ctrl-D ends a remote console,
+ on certain shells, the terminal is messed up. By intercepting
+ those events with a signal handler in the remote console, we can
+ avoid those issues. (closes issue #13464) Reported by: tzafrir
+ Patches: 20081110__bug13464.diff.txt uploaded by Corydon76
+ (license 14) Tested by: blitzrage
+
+2008-12-11 22:44 +0000 [r163316] Matt Nicholson <mnicholson@digium.com>
+
+ * pbx/pbx_dundi.c: Clean up the dundi cache every 5 minutes.
+ (closes issue #13819) Reported by: adomjan Patches:
+ pbx_dundi.c-clearcache.patch uploaded by adomjan (license 487)
+ dundi_clearecache3.diff uploaded by mnicholson (license 96)
+ Tested by: adomjan
+
+2008-12-11 21:46 +0000 [r163092-163253] Russell Bryant <russell@digium.com>
+
+ * funcs/func_strings.c, funcs/func_cut.c: Fix some observed
+ slowdowns in dialplan processing. The change is to remove
+ autoservice usage from dialplan functions that do not need it
+ because they do not perform operations that potentially block.
+ (closes issue #13940) Reported by: tbelder
+
+ * res/res_features.c: Fix an issue that made it so you could only
+ have a single caller executing a custom feature at a time. This
+ was especially problematic when custom features ran for any
+ appreciable amount of time. The fix turned out to be quite
+ simple. The dynamic features are now stored in a read/write list
+ instead of a list using a mutex. (closes issue #13478) Reported
+ by: neutrino88 Fix suggested by file
+
+2008-12-11 16:51 +0000 [r163088] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: Don't wait forever, if there's a specified
+ recording timeout. (closes issue #13885) Reported by: bamby
+ Patches: res_agi.c.patch uploaded by bamby (license 430)
+
+2008-12-11 16:46 +0000 [r163080-163084] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Revert this cast to long. Using time_t here
+ causes build failures on a FreeBSD 32-bit build.
+
+ * apps/app_queue.c: Fix a potential crash due to unsafe datastore
+ handling. This patch also contains a conversion from using long
+ to time_t for representing times for a queue, as well as some
+ whitespace fixes. (closes issue #14060) Reported by: nivek
+ Patches: datastore_fixup.patch.corrected uploaded by nivek
+ (license 636) with slight modification from me Tested by: nivek
+
+2008-12-10 22:52 +0000 [r162874-162926] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_musiconhold.c: Oops, inverted logic for a strcasecmp
+ check. Pointed out by mmichelson, thanks!
+
+ * res/res_musiconhold.c: (closes issue #13229) Reported by:
+ clegall_proformatique Ensure that moh_generate does not return
+ prematurely before local_ast_moh_stop is called. Also, the sleep
+ in mp3_spawn now only occurs for http locations since it seems to
+ have been added originally only for failing media streams.
+
+2008-12-10 19:01 +0000 [r162738-162804] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Fix subscription based MWI up a bit. We only
+ want to put sip: at the beginning of the URI if it is not already
+ there and revert code to ignore destination check if subscribing
+ for MWI. (closes issue #12560) Reported by: vsauer Patches:
+ patch001.diff uploaded by ramonpeek (license 266)
+
+ * channels/chan_sip.c: When a SIP peer unregisters set the expiry
+ time back to 0 so that the 200 OK contains an expires of 0.
+ (closes issue #13599) Reported by: hjourdain Patches:
+ chan_sip.c.diff uploaded by hjourdain (license 583)
+
+2008-12-10 16:45 +0000 [r162671] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/ael/ael.flex: (closes issue #14022)
+ Reported by: wetwired Tested by: murf I checked, and I added a
+ mod to the trunk version of Asterisk that would make it 8-bit
+ transparent on 27 Nov 2007, but I made no such updates to 1.4. My
+ best guess is that 1.4 was released, and it was not appropriate
+ to commit an enhancement. But I'm going to add the same fixes to
+ 1.4 now, for the following reasons: 1. wetwired is correct; 1.4
+ is **mostly** 8-bit transparent now. This is because the lexical
+ token forming rules use . in most 'word' state continuances. It's
+ just the beginning of a 'word' that is picky. 2. Accepting 8-bit
+ chars in some places and not others leads to bug reports like
+ this.
+
+2008-12-10 16:44 +0000 [r162659-162670] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/stringfields.h: Update to stringfield handling
+ so that side-effects on parameters are not evaluated multiple
+ times. An example where this caused a problem was in chan_sip.c,
+ with the line ast_string_field_set(p, fromdomain, ++fromdomain);
+ This patch was originally uploaded to issue #13783 by jamessan.
+ While the issue was closed for other reasons, this patch is valid
+ and fixes a separate problem, and is thus being committed.
+
+ * channels/chan_sip.c: Revert fix for issue 13570. It has caused
+ more problems than it helped to fix. (closes issue #13783)
+ Reported by: navkumar (closes issue #14025) Reported by: ffs
+
+ * doc/misdn.txt: Add missing documentation to misdn.txt (closes
+ issue #14052) Reported by: festr Patches: misdn.txt.patch
+ uploaded by festr (license 443)
+
+2008-12-10 16:05 +0000 [r162653] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Increment the sequence number on the end packets for
+ RFC2833. After reading the RFC some more and doing some testing I
+ agree with this change. (closes issue #12983) Reported by: vt
+ Patches: dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license
+ 520)
+
+2008-12-09 23:08 +0000 [r162463] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Oops, should be "tz", not "zonetag".
+
+2008-12-09 22:17 +0000 [r162413] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, include/asterisk/utils.h, main/utils.c: Remove
+ the test_for_thread_safety() function completely. The test is not
+ valid. Besides, if we actually suspected that recursive mutexes
+ were not working, we would get a ton of LOG_ERROR messages when
+ DEBUG_THREADS is turned on. (inspired by a discussion on the
+ asterisk-dev list)
+
+2008-12-09 21:53 +0000 [r162348] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: We appear to have documented tz= in the
+ [general] section of voicemail.conf, without actually having
+ implemented it. Oops. (Reported by Olivier on the -users list)
+
+2008-12-09 21:14 +0000 [r162341] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_directed_pickup.c: Add 'down' as a valid state for
+ directed call pickup. This creeps up when we receive session
+ progress when dialing a device and not ringing. (closes issue
+ #14005) Reported by: ddl
+
+2008-12-09 20:57 +0000 [r162286] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: Fix an issue where callers on an incoming call
+ on an SLA trunk would not hear ringback. We need to make sure
+ that we don't start writing audio to the trunk channel until
+ we're actually ready to answer it. Otherwise, the channel driver
+ will treat it as inband progress, even though all they are
+ getting is silence. (closes issue #12471) Reported by: mthomasslo
+
+2008-12-09 20:44 +0000 [r162273] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_festival.c: Fix double declaration of 'x' on the PPC
+ platform. (closes issue #14038) Reported by: ffloimair
+
+2008-12-09 20:28 +0000 [r162265] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: If we fail to start a thread for the pbx to run in,
+ we need to be sure to decrease the number of active calls on the
+ system. This fix may relate to ABE-1713, but it is not certain
+ yet.
+
+2008-12-09 20:20 +0000 [r162264] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/ael/ael.flex: In discussion with
+ seanbright on #asterisk-dev, I have added a default rule, and an
+ option to suppress the default rule from being generated in the
+ flex output, for the sake of those OS's where they didn't tweak
+ flex's ECHO macro, and the compiler doesn't like it. The
+ regressions are OK with this.
+
+2008-12-09 19:47 +0000 [r162188-162204] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: Make sure that the timestamp for DTMF is not the same
+ as the previous voice frame and do not send audio when
+ transmitting DTMF as this confuses some equipment. (closes issue
+ #13209) Reported by: ip-rob Patches: 13209.diff uploaded by file
+ (license 11) Tested by: ip-rob, bujones
+
+ * main/rtp.c: Take video into account when early bridging RTP.
+ (closes issue #13535) Reported by: davidw
+
+2008-12-09 18:13 +0000 [r162136] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/ael/ael.flex: Previous fix used ast_malloc
+ and ast_copy_string and messed up the standalone stuff. Fixed.
+
+2008-12-09 17:07 +0000 [r162071] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_phone.c: For some reason, after a distclean, gcc
+ started returning 'value computed is not used'. Fixing (for
+ --enable-dev-mode).
+
+2008-12-09 16:46 +0000 [r162014] Russell Bryant <russell@digium.com>
+
+ * apps/app_disa.c: Allow DISA to handle extensions that start with
+ #. (closes issue #13330) Reported by: jcovert
+
+2008-12-09 16:31 +0000 [r162013] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/pbx_ael.c, include/asterisk/ael_structs.h,
+ pbx/ael/ael.flex: (closes issue #14019) Reported by: ckjohnsonme
+ Patches: 14019.diff uploaded by murf (license 17) Tested by:
+ ckjohnsonme, murf This crash was the result of a few small errors
+ that would combine in 64-bit land to result in a crash. 32-bit
+ land might have seen these combine to mysteriously drop the args
+ to an application call, in certain circumstances. Also, in trying
+ to find this bug, I spotted a situation in the flex input, where,
+ in passing back a 'word' to the parser, it would allocate a
+ buffer larger than necessary. I changed the usage in such
+ situations, so that strdup was not used, but rather, an
+ ast_malloc, followed by ast_copy_string. I removed a field from
+ the pval struct, in u2, that was never getting used, and set in
+ one spot in the code. I believe it was an artifact of a previous
+ fix to make switch cases work invisibly with extens. And, for
+ goto's I removed a '!' from before a strcmp, that has been there
+ since the initial merging of AEL2, that might prevent the proper
+ target of a goto from being found. This was pretty harmless on
+ its own, as it would just louse up a consistency check for users.
+ Many thanks to ckjohnsonme for providing a simplified and
+ complete set of information about the bug, that helped
+ considerably in finding and fixing the problem. Now, to get
+ aelparse up and running again in trunk, and out of its "horribly
+ broken" state, so I can run the regression suite!
+
+2008-12-09 14:52 +0000 [r161948] Russell Bryant <russell@digium.com>
+
+ * main/app.c: Fix a problem with GROUP() settings on a masquerade.
+ The previous code carried over group settings from the old
+ channel to the new one. However, it did nothing with the group
+ settings that were already on the new channel. This patch removes
+ all group settings that already existed on the new channel. I
+ have a more complicated version of this patch which addresses
+ only the most blatant problem with this, which is that a channel
+ can end up with multiple group settings in the same category.
+ However, I could not think of a use case for keeping any of the
+ group settings from the old channel, so I went this route for
+ now. (closes AST-152)
+
+2008-12-08 17:52 +0000 [r161725] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Make the usereqphone option work again.
+ (closes issue #13474) Reported by: mmaguire Patches:
+ 20080912_bug13474.diff uploaded by mmaguire (license 571)
+
+2008-12-05 21:02 +0000 [r161426] Sean Bright <sean.bright@gmail.com>
+
+ * main/astobj2.c, /, include/asterisk/astobj2.h: Merged revisions
+ 161421 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r161421 | seanbright | 2008-12-05 15:50:23 -0500 (Fri, 05 Dec
+ 2008) | 8 lines Fix build errors on FreeBSD (uint -> unsigned
+ int). (closes issue #14006) Reported by: alphaque Patches:
+ astobj2.h-patch uploaded by alphaque (license 259) (Slightly
+ modified by seanbright) ........
+
+2008-12-05 16:51 +0000 [r161354] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * utils/smsq.c: kill a warning
+
+2008-12-05 14:12 +0000 [r161287] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Fix a NULL format string warning found by buildbot.
+
+2008-12-04 18:30 +0000 [r161013] Jeff Peeler <jpeeler@digium.com>
+
+ * main/rtp.c: (closes issue #13835) Reported by: matt_b Tested by:
+ jpeeler This mirrors a check that was present in ast_rtp_read to
+ also be in ast_rtp_raw_write to not schedule sending the receiver
+ report if the remote RTCP endpoint address isn't present in the
+ RTCP structure. Closes AST-142.
+
+2008-12-04 16:44 +0000 [r160943] Mark Michelson <mmichelson@digium.com>
+
+ * main/callerid.c: Fix a callerid parsing issue. If someone
+ formatted callerid like the following: "name <number>" (including
+ the quotation marks), then the parts would be parsed as name:
+ "name number: number This is because the closing quotation mark
+ was not discovered since the number and everything after was
+ parsed out of the string earlier. Now, there is a check to see if
+ the closing quote occurs after the number, so that we can know if
+ we should strip off the opening quote on the name. Closes AST-158
+
+2008-12-03 21:54 +0000 [r160770] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Some compilers warn on null format strings;
+ some don't (caught by buildbot)
+
+2008-12-03 21:38 +0000 [r160764] Jason Parker <jparker@digium.com>
+
+ * channels/chan_agent.c: Only show this warning when we want to
+ show it. (closes issue #13982) Reported by: coolmig Patches:
+ chan_agent.c.patch uploaded by coolmig (license 621)
+
+2008-12-03 20:41 +0000 [r160703] Steve Murphy <murf@digium.com>
+
+ * funcs/func_callerid.c: (closes issue #13597) Reported by:
+ john8675309 Patches: patch.13597 uploaded by murf (license 17)
+ Tested by: murf, john8675309 This patch causes the setcid func to
+ update the CDR clid after setting the channel field.
+
+2008-12-03 17:55 +0000 [r160480-160570] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: During bridge code, the channel bridge may
+ return a retry code, if a transfer was initiated but not yet
+ completed. If the bridge is immediately retried, then we may send
+ a storm of TXREQ packets, even though the first set is sent
+ reliably (retransmitted). Fixes AST-137.
+
+ * pbx/pbx_spool.c: If an entry is added to the directory during a
+ scan when another entry expires, then that new entry will not be
+ processed promptly, but must wait for either a future entry to
+ start or a current entry's retry to occur. If no other entries
+ exist in the directory (other than the new entries) when a bunch
+ expire, then the new entries must wait until another new entry is
+ added to be processed. This was a rather weird race condition,
+ really. Fixes AST-147.
+
+ * pbx/pbx_spool.c: Don't start scanning the directory until all
+ modules are loaded, because some required modules (channels,
+ apps, functions) may not yet be in memory yet. Fixes AST-149.
+
+ * channels/chan_sip.c: Jon Bonilla (Manwe) pointed out on the -dev
+ list: "I guess that having only ip-phones in mind is not a good
+ approach. Since it is possible to have a sip proxy connected to
+ asterisk we could receive a 407 (unauthorized) or 483 (too many
+ hops) as response and dialog ending would not be a good
+ behavior." So modified.
+
+2008-12-02 23:58 +0000 [r160390-160411] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c: Channel is masqueraded, don't keep alive
+
+ * res/res_features.c: A situation like A calls B, A builtin_atxfers
+ B to C, C parks B would lead to a crash. Thanks to file for
+ telling me how to fix it! (closes issue #13854) Reported by: Adam
+ Lee Tested by: otherwiseguy
+
+2008-12-02 17:42 +0000 [r160297] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: When the text does not match exactly (e.g.
+ RTP/SAVP), then the %n conversion fails, and the resulting
+ integer is garbage. Thus, we must initialize the integer and
+ check it afterwards for success. (closes issue #14000) Reported
+ by: folke Patches: asterisk-sipbg-sscanf-1.4.22.diff uploaded by
+ folke (license 626) asterisk-sipbg-sscanf-1.6.0.1.diff uploaded
+ by folke (license 626) asterisk-sipbg-sscanf-trunk-r159896.diff
+ uploaded by folke (license 626)
+
+2008-12-02 01:16 +0000 [r160266] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/astmm.h: make compile with dev mode and malloc
+ debug
+
+2008-12-02 00:25 +0000 [r160207] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/stringfields.h, apps/app_voicemail.c,
+ main/pbx.c, main/frame.c: Ensure that Asterisk builds with
+ --enable-dev-mode, even on the latest gcc and glibc.
+
+2008-12-01 Tilghman Lesher <tilghman@digium.com>
+
+ * Released 1.4.23-rc2
+
+2008-12-01 17:27 +0000 [r160003] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Apply some logic used in iax2_indicate() to
+ iax2_setoption(), as well, since they both have the potential to
+ send control frames in the middle of call setup. We have to wait
+ until we have received a message back from the remote end before
+ we try to send any more frames. Otherwise, the remote end will
+ consider it invalid, and we'll get stuck in an INVAL/VNAK storm.
+
+2008-12-01 16:08 +0000 [r159976] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/manager.c: Get rid of the useless format string and argument
+ in the Bogus/ manager channelname. Noted by kpfleming and name
+ Bogus/manager suggested by eliel
+
+2008-12-01 14:52 +0000 [r159900] Russell Bryant <russell@digium.com>
+
+ * .cleancount: Force a "make clean" to avoid a bizarre build issue
+ ...
+
+2008-12-01 14:05 +0000 [r159897] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/manager.c: make manager compile on OpenBSD. The last (10th)
+ argument to ast_channel_alloc here should be a pointer and NULL
+ is not really a pointer.
+
+2008-11-29 16:58 +0000 [r159808] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/enum.c, utils/frame.c, configure, res/res_agi.c,
+ include/asterisk/module.h, main/logger.c, main/dns.c,
+ include/asterisk/threadstorage.h, include/asterisk/utils.h,
+ include/asterisk/devicestate.h, channels/chan_sip.c,
+ include/asterisk/dundi.h, main/jitterbuf.c,
+ channels/chan_agent.c, configure.ac, utils/astman.c,
+ include/asterisk/cli.h, include/asterisk/channel.h,
+ include/jitterbuf.h, include/asterisk/manager.h,
+ main/ast_expr2.c, Makefile, include/asterisk/logger.h,
+ include/asterisk/res_odbc.h, main/srv.c, channels/chan_misdn.c,
+ include/asterisk/linkedlists.h, include/asterisk/lock.h,
+ include/asterisk/strings.h, makeopts.in,
+ include/asterisk/stringfields.h, utils/check_expr.c,
+ channels/chan_vpb.cc, res/res_features.c, channels/chan_iax2.c:
+ update dev-mode compiler flags to match the ones used by default
+ on Ubuntu Intrepid, so all developers will see the same warnings
+ and errors since this branch already had some printf format
+ attributes, enable checking for them and tag functions that
+ didn't have them format attributes in a consistent way
+
+2008-11-26 20:21 +0000 [r159476-159571] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_oss.c, channels/busy.h (removed),
+ channels/ring_tone.h (added), channels/chan_alsa.c,
+ channels/ringtone.h (removed), channels/busy_tone.h (added),
+ channels/Makefile: rename these files so as to avoid conflicts
+ when users update their working copies and have unversioned files
+ already in place
+
+ * channels, agi/Makefile, utils/Makefile, channels/busy.h (added),
+ Makefile.moddir_rules, Makefile.rules, channels/ringtone.h
+ (added), channels/Makefile: simplify (and slightly bug-fix) the
+ recent developer-oriented COMPILE_DOUBLE mode add channels/busy.h
+ and channels/ringtone.h to the repository instead of generating
+ them repeatedtly; most users do not change the settings to build
+ them, but the Makefile rules are still there if they wish to do
+ so ensure that 'make clean' removes dependency files for .i files
+ that are created in COMPILE_DOUBLE mode
+
+2008-11-25 22:41 +0000 [r159316] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, channels/chan_iax2.c: (closes issue #12694) Reported
+ by: yraber Patches: 12694.2nd.diff uploaded by murf (license 17)
+ Tested by: murf, laurav Thanks to file (Joshua Colp) for his IAX
+ fix. the change to cdr.c allows no-answer to percolate up into
+ CDR's, and feels like the right place to locate this fix; if BUSY
+ is done here, no-answer should be, too.
+
+2008-11-25 21:56 +0000 [r159246-159269] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Don't try to send a response on a NULL pvt.
+ (closes issue #13919) Reported by: barthpbx Patches:
+ chan_iax2.c.patch uploaded by eliel (license 64) Tested by:
+ barthpbx
+
+ * /, channels/chan_iax2.c: Merged revisions 159245 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25
+ Nov 2008) | 7 lines Regression fix for last security fix. Set the
+ iseqno correctly. (closes issue #13918) Reported by: ffloimair
+ Patches: 20081119__bug13918.diff.txt uploaded by Corydon76
+ (license 14) Tested by: ffloimair ........
+
+2008-11-25 17:34 +0000 [r159158] Russell Bryant <russell@digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: Add ao2_trylock() to
+ go along with ao2_lock() and ao2_unlock()
+
+2008-11-25 16:23 +0000 [r159096] Terry Wilson <twilson@digium.com>
+
+ * apps/app_festival.c: Add missing variable declaration in the PPC
+ code
+
+2008-11-25 04:50 +0000 [r159025] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_rpt.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac: System call ioperm is non-portable, so check for
+ its existence in autoconf. (Closes issue #13863)
+
+2008-11-22 00:04 +0000 [r158629] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c: (closes
+ issue #13786) Reported by: tzafrir When compiling against Zaptel
+ dahdi_compat will now only define all the DAHDI defines if the
+ Zaptel define is present. Also, there is no such thing as
+ DAHDI_PRI.
+
+2008-11-21 23:14 +0000 [r158603] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: In reference to the fix made for 13871, I was
+ merging the fix into 1.6.0 and realized I missed the code in the
+ h-exten block, and didn't catch it because my test case had the
+ h-exten commented out. So, this corrects the code I missed, as a
+ preventative against another crash report. Tested with the
+ h-exten defined, all is well.
+
+2008-11-21 23:07 +0000 [r158600] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: The passed extension may not be the same in the list
+ as the current entry, because we strip spaces when copying the
+ extension into the structure. Therefore, use the copied item to
+ place the item into the list. (found by lmadsen on -dev, fixed by
+ me)
+
+2008-11-21 22:05 +0000 [r158539] Russell Bryant <russell@digium.com>
+
+ * main/astobj2.c, include/asterisk/astobj2.h: When compiling with
+ DEBUG_THREADS, report the real file/func/line for
+ ao2_lock/ao2_unlock
+
+2008-11-21 21:19 +0000 [r158483] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: (closes issue #13871) Reported by: mdu113
+ This one is totally my fault. The code doesn't even create a
+ bridge if the channel CDR has POST_DISABLED. I didn't check for
+ that at the end of the bridge. Fixed with a few small insertions.
+ Tested. Looks good. No cdr generated, no crash, no unnecc. data
+ objects created either.
+
+2008-11-21 15:24 +0000 [r158053-158306] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: This change had somehow gotten reverted due to
+ a completely unrelated commit. Thanks to Theo Belder on the
+ Asterisk-dev list for pointing this out.
+
+ * include/asterisk/file.h, main/frame.c, main/file.c,
+ include/asterisk/frame.h: There was an issue when attempting to
+ reference an embedded frame in a freed ast_filestream. This patch
+ makes use of the ao2 functions to make sure that we do not free
+ an ast_filestream structure until the embedded ast_frame has been
+ "freed" as well. (closes issue #13496) Reported by: fst-onge
+ Patches: filestream_frame_1_4.diff uploaded by putnopvut (license
+ 60) Tested by: putnopvut Closes AST-89
+
+ * channels/chan_sip.c: We don't handle 4XX responses to BYE well.
+ According to section 15 of RFC 3261, we should terminate a dialog
+ if we receive a 481 or 408 in response to our BYE. Since I am
+ aware of at least one phone manufacturer who may sometimes send a
+ 404 as well, I am being liberal and saying that any 4XX response
+ to a BYE should result in a terminated dialog. (closes issue
+ #12994) Reported by: pabelanger Patches: 12994.patch uploaded by
+ putnopvut (license 60) Closes AST-129
+
+ * apps/app_dial.c, channels/chan_sip.c: Make sure to set the hangup
+ cause on the calling channel in the case that ast_call() fails.
+ For incoming SIP channels, this was causing us to send a 603
+ instead of a 486 when the call-limit was reached on the
+ destination channel. (closes issue #13867) Reported by: still_nsk
+ Patches: 13867.diff uploaded by putnopvut (license 60) Tested by:
+ blitzrage
+
+2008-11-20 01:46 +0000 [r158010] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Merged revision 157977 from
+ https://origsvn.digium.com/svn/asterisk/team/group/issue8824
+ ........ Fixes JIRA ABE-1726 The dial extension could be empty if
+ you are using MISDN_KEYPAD to control ISDN provider features.
+
+2008-11-19 21:34 +0000 [r157859] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/stdtime/Makefile, codecs/gsm/src, main/db1-ast/btree,
+ channels/misdn/Makefile, main/db1-ast/recno, pbx/ael, channels,
+ main/db1-ast/Makefile, main/stdtime, main/db1-ast/hash,
+ codecs/gsm/Makefile, main/db1-ast/db, Makefile.moddir_rules,
+ channels/misdn, main/db1-ast/mpool, pbx/Makefile, Makefile.rules,
+ res/snmp, res/Makefile: the gcc optimizer frequently finds broken
+ code (use of uninitalized variables, unreachable code, etc.),
+ which is good. however, developers usually compile with the
+ optimizer turned off, because if they need to debug the resulting
+ code, optimized code makes that process very difficult. this
+ means that we get code changes committed that weren't adequately
+ checked over for these sorts of problems. with this build system
+ change, if (and only if) --enable-dev-mode was used and
+ DONT_OPTIMIZE is turned on, when a source file is compiled it
+ will actually be preprocessed (into a .i or .ii file), then
+ compiled once with optimization (with the result sent to
+ /dev/null) and again without optimization (but only if the first
+ compile succeeded, of course). while making these changes, i did
+ some cleanup work in Makefile.rules to move commonly-used
+ combinations of flag variables into their own variables, to make
+ the file easier to read and maintain
+
+2008-11-18 22:47 +0000 [r157503] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Add some missing invite state changes
+ necessary in the sip_write function. Not setting the invite state
+ correctly on the call was resulting in the Record application
+ leaving empty files. I also have updated the doxygen comment next
+ to the declaration of the INV_EARLY_MEDIA constant to reflect
+ that we also use this state when we *send* a 18X response to an
+ INVITE. (closes issue #13878) Reported by: nahuelgreco Patches:
+ sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco
+ (license 162) Tested by: putnopvut
+
+2008-11-18 19:13 +0000 [r157365] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c: (closes issue #13899) Reported by: akkornel
+ This fix is the result of a bug fix in ast_app_separate_args
+ r124395. If an argument does not exist it should always be set to
+ a null string rather than a null pointer.
+
+2008-11-18 18:25 +0000 [r157305] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c, channels/chan_local.c, res/res_features.c,
+ include/asterisk/channel.h, apps/app_followme.c: Fix a crash in
+ the end_bridge_callback of app_dial and app_followme which would
+ occur at the end of an attended transfer. The error occurred
+ because we initially stored a pointer to an ast_channel which
+ then was hung up due to a masquerade. This commit adds a "fixup"
+ callback to the bridge_config structure to allow for
+ end_bridge_callback_data to be changed in the case that a new
+ channel pointer is needed for the end_bridge_callback.
+
+2008-11-15 19:31 +0000 [r157104-157163] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile, Makefile.rules: when an individual directory dist-clean
+ is run, run clean in that directory first, and when running
+ top-level dist-clean, do not run subdirectory clean operations
+ twice
+
+ * Makefile.moddir_rules: dist-clean should remove dependency
+ information files as well
+
+ * contrib/asterisk-ng-doxygen: major update to doxygen
+ configuration file: 1) update to doxygen 1.5.x style file, as
+ used in trunk 2) tell doxygen where are header files are, so
+ include-file processing can be done 3) make all macros that are
+ used to define variables/functions be expanded, so that doxygen
+ will properly document the resulting variable/function 4) make
+ all macros that are used to provide the contents of a variable
+ (structure) be expanded, so that doxygen will be able to document
+ the resulting fields 5) suppress compiler attributes
+ (__attribute__(xxx)) from being seen by doxygen, so it will
+ properly match up function definition and usage (for an example
+ of th effect of this, look at the doxygen docs for ast_log() from
+ before and afte this commit)
+
+2008-11-14 15:18 +0000 [r156816] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: If the prompt to reenter a voicemail
+ password timed out, it resulted in the password not being saved,
+ even if the input matched what you gave when first prompted to
+ enter a new password. This is because the return value of
+ ast_readstring was checked, but not checked properly. This bug
+ was discovered by Jared Smith during an Asterisk training course.
+ Thanks for reporting it!
+
+2008-11-14 00:41 +0000 [r156688-156755] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_while.c: ast_waitfordigit() requires that the channel be
+ up, for no good logical reason. This prevents While/EndWhile from
+ working within the "h" extension. Reported by: jgalarneau (for
+ ABE C.2) Fixed by: me
+
+ * main/manager.c: Provide more space for all the data which can
+ appear in an originating channel name. (closes issue #13398)
+ Reported by: bamby Patches: manager.c.diff uploaded by bamby
+ (license 430)
+
+2008-11-13 11:58 +0000 [r156485-156510] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, autoconf/ast_gcc_attribute.m4: revert this change...
+ non-functional changes don't belong here
+
+ * configure, autoconf/ast_gcc_attribute.m4: correct minor syntax
+ error... no functional change
+
+2008-11-12 21:18 +0000 [r156386] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c: When using call limits under 1 second, infinite
+ call lengths are allowed, instead. (closes issue #13851) Reported
+ by: ruddy
+
+2008-11-12 19:36 +0000 [r156297] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: It turns out that the 0x0XX00 codes being returned
+ for N, X, and Z are off by one, as per conversation with jsmith
+ on #asterisk-dev; he was teaching a class and disconcerted that
+ this published rule was not being followed, with patterns _NXX,
+ _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should
+ have been. This change, tested on these 3 patterns now picks the
+ proper one. However, this change may surprise users who set up
+ dialplans based on previous behavior, which has been there for
+ what, 2 and half years or so now.
+
+2008-11-12 19:26 +0000 [r156294] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: If the SLA thread is not started, then reload
+ causes a memory leak. (closes issue #13889) Reported by: eliel
+ Patches: app_meetme.c.patch uploaded by eliel (license 64)
+
+2008-11-12 19:10 +0000 [r156289] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c: For whatever reason, gcc only warned me about
+ the possible use of an uninitialized variable when compiling
+ 1.6.1.
+
+2008-11-12 18:39 +0000 [r156229] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Revert revision 132506, since it
+ occasionally caused IAX2 HANGUP packets not to be sent, and
+ instead, schedule a task to destroy the iax2 pvt structure 10
+ seconds later. This allows the IAX2 HANGUP packet to be queued,
+ transmitted, and ACKed before the pvt is destroyed. (closes issue
+ #13645) Reported by: dzajro Patches:
+ 20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: vazir Reviewed: http://reviewboard.digium.com/r/51/
+
+2008-11-12 17:53 +0000 [r156178] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_meetme.c: (closes issue #13173) Reported by: pep This
+ change adds an announce_thread responsible for playing
+ announcements to an existing conference. This allows all
+ announcing to be immediately stopped if necessary but more
+ importantly allows other threads that need to play something to
+ not block. There are multiple benefits to this, but the actual
+ bug is for solving the scenario for a channel to be unusable
+ after hang up for the entire duration of the parting
+ announcement. The parting announcement can be extremely long
+ depending on what the user recorded upon joining the conference.
+ Reviewed by Russell on Review Board:
+ http://reviewboard.digium.com/r/25/
+
+2008-11-12 17:38 +0000 [r156167] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: When doing some tests, I was having a crash at
+ the end of every call if an attended transfer occurred during the
+ call. I traced the cause to the CDR on one of the channels being
+ NULL. murf suggested a check in the end bridge callback to be
+ sure the CDR is non-NULL before proceeding, so that's what I'm
+ adding.
+
+2008-11-12 17:29 +0000 [r156164] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Move the sanity check that makes sure "always
+ fork" is not set along with the console option to be after the
+ code that reads options from asterisk.conf. This resolves a
+ situation where Asterisk can start taking up 100% when
+ misconfigured. (Thanks to Bryce Porter (x86 on IRC) for letting
+ me log in to his system to figure out what was causing the 100%
+ CPU problem.)
+
+2008-11-10 21:07 +0000 [r155861] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Channel drivers assume that when their
+ indicate callback is invoked, that the channel on which the
+ callback was called is locked. This patch corrects an instance in
+ chan_agent where a channel's indicate callback is called directly
+ without first locking the channel. This was leading to some
+ observed locking issues in chan_local, but considering that all
+ channel drivers operate under the same expectations, the generic
+ fix in chan_agent is the right way to go. AST-126
+
+2008-11-10 20:49 +0000 [r155803] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/valgrind.txt: I got tired of saying this in every single
+ bugnote referring to this file.
+
+2008-11-09 01:08 +0000 [r155553] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h,
+ apps/app_followme.c: Use static functions here instead of nested
+ ones. This requires a small change to the ast_bridge_config
+ struct as well. To understand the reason for this change, see the
+ following post:
+ http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html
+
+2008-11-07 22:27 +0000 [r155398] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Clarify error message. (closes issue #13809)
+ Reported by: denke Patches: 20081104__bug13809.diff.txt uploaded
+ by Corydon76 (license 14) Tested by: denke
+
+2008-11-06 19:45 +0000 [r155011] Mark Michelson <mmichelson@digium.com>
+
+ * configs/voicemail.conf.sample: The documentation listed the
+ ability to set 'maxmsg' per context. The truth is that you can
+ only set this in the general section or per mailbox. Thus I am
+ updating the sample config file to be more accurate. Thanks to
+ sasargen on IRC for bringing up this issue.
+
+2008-11-05 16:44 +0000 [r154724] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: The logic of a strcasecmp call was
+ reversed (closes issue #13841) Reported by: clegall_proformatique
+
+2008-11-05 16:06 +0000 [r154685] Steve Murphy <murf@digium.com>
+
+ * main/channel.c: This fix was prompted by communication from user,
+ who was seeing thousands of error logs... looks like EAGAIN. Made
+ such uninteresting.
+
+2008-11-04 20:49 +0000 [r154365] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: On busy systems, it's possible for the
+ values checked within a single line of code to change, unless the
+ structure is locked to ensure a consistent state. (closes issue
+ #13717) Reported by: kowalma Patches: 20081102__bug13717.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: kowalma
+
+2008-11-04 19:01 +0000 [r154266] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: JIRA ABE-1703 mISDN sets the channel to
+ the wrong state when it receives the indication
+ AST_CONTROL_RINGING.
+
+2008-11-04 18:58 +0000 [r154060-154263] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_h323.c: Make the monitor thread non-detached, so it
+ can be joined (suggested by Russell on -dev list).
+
+ * apps/app_voicemail.c: Attempting to expunge a mailbox when the
+ mailstream is NULL will crash Asterisk. (Closes issue #13829)
+ Reported by: jaroth Patch by: me (modified jaroth's patch)
+
+ * main/rtp.c: Remove the potential for a division by zero error.
+ (Closes issue #13810)
+
+2008-11-03 13:01 +0000 [r153823] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_oss.c, channels/chan_dahdi.c, funcs/func_odbc.c,
+ main/file.c, main/http.c, main/utils.c, pbx/pbx_config.c,
+ res/res_jabber.c: somehow missed a bunch of gcc 4.3.x warnings in
+ this branch on the first pass
+
+2008-11-02 19:51 +0000 [r153651] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/features.h: features.h depends on linkedlists.h,
+ so include it
+
+2008-11-01 18:22 +0000 [r153337] Kevin P. Fleming <kpfleming@digium.com>
+
+ * utils/frame.c, main/cli.c, utils/stereorize.c, main/channel.c,
+ funcs/func_enum.c, channels/chan_dahdi.c, main/manager.c,
+ channels/chan_skinny.c, main/ast_expr2f.c, res/res_agi.c,
+ pbx/ael/ael_lex.c, main/http.c, channels/chan_alsa.c,
+ pbx/ael/ael.flex, formats/format_gsm.c, apps/app_adsiprog.c,
+ formats/format_wav.c, apps/app_festival.c,
+ main/db1-ast/hash/hash_page.c, main/translate.c,
+ res/res_crypto.c, agi/eagi-test.c, formats/format_ogg_vorbis.c,
+ utils/astman.c, channels/chan_oss.c, agi/eagi-sphinx-test.c,
+ pbx/ael/ael.tab.c, main/file.c, pbx/ael/ael.tab.h,
+ apps/app_sms.c, pbx/pbx_dundi.c, res/res_indications.c,
+ utils/streamplayer.c, apps/app_chanspy.c, main/asterisk.c,
+ apps/app_voicemail.c, utils/muted.c, pbx/ael/ael.y,
+ apps/app_authenticate.c, formats/format_wav_gsm.c,
+ res/res_musiconhold.c, channels/chan_iax2.c: fix a bunch of
+ potential problems found by gcc 4.3.x, primarily bare strings
+ being passed to printf()-like functions and ignored results from
+ read()/write() and friends
+
+2008-10-31 22:36 +0000 [r153270] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c, apps/app_followme.c: Add end_bridge_callback
+ for app_follome and add AUTOLOOP flag to res_features
+
+2008-10-31 Tilghman Lesher <tlesher@digium.com>
+
+ * Asterisk 1.4.23-rc1 released.
+
+2008-10-31 16:30 +0000 [r153114] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Turn off qualify on uncached realtime peers.
+ (Closes issue #13383)
+
+2008-10-31 15:45 +0000 [r153095] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, res/res_features.c, include/asterisk/channel.h:
+ Recent CDR fixes moved execution of the 'h' exten into the
+ bridging code, so variables that were set after ast_bridge_call
+ was called would not show up in the 'h' exten. Added a callback
+ function to handle setting variables, etc. from w/in the bridging
+ code. Calls back into a nested function within the function
+ calling ast_bridge_call (closes issue #13793) Reported by:
+ greenfieldtech
+
+2008-10-30 20:58 +0000 [r152992] Sean Bright <sean.bright@gmail.com>
+
+ * bootstrap.sh: The -I argument to aclocal needs a space before the
+ include directory name.
+
+2008-10-30 20:33 +0000 [r152922-152958] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_h323.c: Cannot join detached threads. See
+ http://www.opengroup.org/onlinepubs/000095399/functions/pthread_join.html
+ (Closes issue #13400)
+
+ * channels/chan_local.c: Unlock before returning, when extension
+ doesn't exist. (closes issue #13807) Reported by: eliel Patches:
+ chan_local.c.patch uploaded by eliel (license 64)
+
+2008-10-30 16:53 +0000 [r152811] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/cdr.c: instead of comparing the string pointer to 0, let's
+ compare the value that was actually parsed out of the string
+ (found by sparse)
+
+2008-10-29 05:23 +0000 [r152539] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix an incorrect usage of sizeof() (closes
+ issue #13795) Reported by: andrew53 Patches:
+ chan_sip_sizeof.patch uploaded by andrew53 (license 519)
+
+2008-10-29 05:19 +0000 [r152535-152538] Steve Murphy <murf@digium.com>
+
+ * configs/features.conf.sample, apps/app_dial.c, apps/app_queue.c:
+ A little documentation cross-ref between features and dial and
+ queue... I wasted some time (stupidly) trying to get the
+ one-touch parking stuff working, because it didn't occur to me
+ that I had to also have the corresponding options in the dial
+ command! Duh! (In all this time, I never set this up before!) So,
+ to keep some poor fool from suffering the same fate, I made the
+ features.conf.sample file mention the corresponding opts in
+ dial/queue; and the docs for dial/app specifically mention the
+ corresponding decls in the feature.conf file. I hope this doesn't
+ spoil some vast, eternal plan...
+
+ * apps/app_dial.c, res/res_features.c, funcs/func_channel.c,
+ include/asterisk/pbx.h, apps/app_queue.c: The magic trick to
+ avoid this crash is not to try to find the channel by name in the
+ list, which is slow and resource consuming, but rather to pay
+ attention to the result codes from the ast_bridge_call, to which
+ I added the AST_PBX_NO_HANGUP_PEER_PARKED value, which now are
+ returned when a channel is parked. If you get AST_PBX_KEEPALIVE,
+ then don't touch the channel pointer. If you get
+ AST_PBX_NO_HANGUP_PEER, or AST_PBX_NO_HANGUP_PEER_PARKED, then
+ don't touch the peer pointer. Updated the several places where
+ the results from a bridge were not being properly obeyed, and
+ fixed some code I had introduced so that the results of the
+ bridge were not overridden (in trunk). All the places that
+ previously tested for AST_PBX_NO_HANGUP_PEER now have to check
+ for both AST_PBX_NO_HANGUP_PEER and
+ AST_PBX_NO_HANGUP_PEER_PARKED. I tested this against the 4 common
+ parking scenarios: 1. A calls B; B answers; A parks B; B hangs up
+ while A is getting the parking slot announcement, immediately
+ after being put on hold. 2. A calls B; B answers; A parks B; B
+ hangs up after A has been hung up, but before the park times out.
+ 3. A calls B; B answers; B parks A; A hangs up while B is getting
+ the parking slot announcement, immediately after being put on
+ hold. 4. A calls B; B answers; B parks A; A hangs up after B has
+ been hung up, but before the park times out. No crash. I also ran
+ the scenarios above against valgrind, and accesses looked good.
+
+2008-10-28 22:32 +0000 [r152368-152463] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Quoting in the wrong direction (Fixes
+ AST-107)
+
+ * apps/app_dial.c: Reset all DIAL variables back to blank, in case
+ Dial is called multiple times per call (which could otherwise
+ lead to inconsistent status reports). (closes issue #13216)
+ Reported by: ruddy Patches: 20081014__bug13216.diff.txt uploaded
+ by Corydon76 (license 14) Tested by: ruddy
+
+2008-10-27 23:28 +0000 [r152286] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Buffer policy setting for half is not
+ needed.
+
+2008-10-27 21:32 +0000 [r152215] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: Inherit ALL elements of CallerID across a
+ local channel. (closes issue #13368) Reported by: Peter Schlaile
+ Patches: 20080826__bug13368.diff.txt uploaded by Corydon76
+ (license 14)
+
+2008-10-26 20:23 +0000 [r152059] Sean Bright <sean.bright@gmail.com>
+
+ * funcs/func_strings.c: Since passing \0 as the second argument to
+ strchr is valid (and will match the trailing \0 of a string) we
+ need to check that first, otherwise we end up with incorrect
+ results. Fix suggested by reporter. (closes issue #13787)
+ Reported by: meitinger
+
+2008-10-25 10:59 +0000 [r151905] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Move AMI initialization to occur after loading
+ modules. This prevents a deadlock when someone tries to initiate
+ a module reload from the AMI just as Asterisk is starting.
+ (closes issue #13778) Reported by: hotsblanc Fix suggested by
+ hotsblanc
+
+2008-10-23 16:04 +0000 [r151763] Terry Wilson <twilson@digium.com>
+
+ * configs/features.conf.sample, res/res_features.c, CHANGES:
+ Backport fix from 1.6.0 that allows you to set
+ parkedcalltransfers=no|caller|callee|both, but default to both
+ which would be the equivalent of the existing behaviour. The
+ problem was that if someone parked a call, the callee and caller
+ would both get assigned the builtin transfer feature, which would
+ not only be potentially giving someone the ability to transfer
+ themselves when they shouldn't have it, but would also dissallow
+ reinviting the media off of the call. (closes issue #12854)
+ Reported by: davidw Patches: parkingfix4.diff.txt uploaded by
+ otherwiseguy Tested by: davidw, otherwiseguy
+
+2008-10-20 04:57 +0000 [r151240-151241] Kevin P. Fleming <kpfleming@digium.com>
+
+ * autoconf/ast_check_pwlib.m4, autoconf/ast_check_openh323.m4,
+ configure.ac: rename this macro to properly reflect what it does
+
+ * autoconf/ast_check_pwlib.m4 (added), autoconf (added),
+ autoconf/acx_pthread.m4 (added), autoconf/ast_func_fork.m4
+ (added), configure, autoconf/ast_gcc_attribute.m4 (added),
+ bootstrap.sh, autoconf/ast_check_gnu_make.m4 (added),
+ autoconf/ast_ext_lib.m4 (added), autoconf/ast_prog_ld.m4 (added),
+ autoconf/ast_c_compile_check.m4 (added),
+ autoconf/ast_c_define_check.m4 (added),
+ autoconf/ast_prog_egrep.m4 (added),
+ autoconf/ast_check_openh323.m4 (added),
+ autoconf/ast_prog_ld_gnu.m4 (added), autoconf/ast_prog_sed.m4
+ (added), acinclude.m4 (removed): break up acinclude.m4 into
+ individual files, which will make it easier to maintain, easier
+ to add new macros (less patching) and will ease maintenance of
+ these macros across Asterisk branches
+
+2008-10-19 19:51 +0000 [r151100-151167] BJ Weschke <bweschke@btwtech.com>
+
+ * main/asterisk.c: As per kpfleming's comments to the prior commit,
+ I'm reverting some of the changes here. A comment was made in bug
+ #13726 "3. The same mistake as in (2) is done in a few other
+ places in the code that check for: #if defined(HAVE_ZAPTEL) ||
+ defined(HAVE_DAHDI) Harmless, but still incorrect." In the case
+ of main/asterisk.c, this is not incorrect because without
+ HAVE_ZAPTEL defined, we're missing the include for ioctl and the
+ namespace that defines DAHDI_TIMERCONFIG which is still required
+ when using Zaptel with the 1.4 branch.
+
+ * main/asterisk.c: Fix the 1.4 branch compile again broken with
+ r150557 when using with Zaptel and not DAHDI (closes issue
+ #13740) reported by: jmls patch by: bweschke
+
+2008-10-18 01:42 +0000 [r150816] BJ Weschke <bweschke@btwtech.com>
+
+ * main/manager.c: Using the GetVar handler in AMI is potentially
+ dangerous (insta-crash [tm]) when you use a dialplan function
+ that requires a channel and then you don't provide one or provide
+ an invalid one in the Channel: parameter. We'll handle this
+ situation exactly the same way it was handled in pbx.c back on
+ r61766. We'll create a bogus channel for the function call and
+ destroy it when we're done. If we have trouble allocating the
+ bogus channel then we're not going to try executing the function
+ call at all and run the risk of crashing. (closes issue #13715)
+ reported by: makoto patch by: bweschke
+
+2008-10-17 17:18 +0000 [r150637] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: Interesting crash. In this case, you exit the
+ bridge with peer completely GONE. I moved the channel find call
+ up to cover the whole peer CDR reset code segment. This appears
+ to solve the crash without changing the logic at all.
+
+2008-10-17 15:31 +0000 [r150557] Jason Parker <jparker@digium.com>
+
+ * main/asterisk.c, main/channel.c, channels/chan_dahdi.c,
+ configure, configure.ac: Correctly allow chan_dahdi to compile
+ against older versions of Zaptel. Don't always define
+ HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..) Minor
+ cleanup to make things clear. (closes issue #13726) Reported by:
+ tzafrir Patches: dahdi_def.diff uploaded by tzafrir (license 46)
+
+2008-10-16 23:40 +0000 [r150298-150304] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c: Reverting changes from commits 150298 and 150301
+ since I was mistakenly under the assumption that dialplan
+ functions *always* required that a channel be present. I need to
+ go home earlier, I think :)
+
+ * main/manager.c: And don't forget to return on the error condition
+
+ * main/manager.c: Don't try to call a dialplan function's read
+ callback from the manager's GetVar handler if an invalid channel
+ has been specified. Several dialplan functions, including CHANNEL
+ and SIP_HEADER, do not check for NULL-ness of the channel being
+ passed in. (closes issue #13715) Reported by: makoto
+
+2008-10-16 15:56 +0000 [r150124] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/chan_misdn.c: Fix memory leak found by customer
+
+2008-10-16 15:26 +0000 [r150056] Steve Murphy <murf@digium.com>
+
+ * cdr/cdr_odbc.c: This patch is relevant to: ABE-1628 and
+ RYM-150398 and AST-103 in internal Digium bug trackers. These
+ fixes address a really subtle memory corruption problem that
+ would happen in machines heavily loaded in production
+ environments. The corruption would always take the form of the
+ STMT object getting nulled out and one of the unixODBC calls
+ would crash trying to access statement->connection. It isn't
+ fully proven yet, but the server has now been running 2.5 days
+ without appreciable memory growth, or any gain of %cpu, and no
+ crashes. Whether this is the problem or not on that server, these
+ fixes are still warranted. As it turns out, **I** introduced
+ these errors unwittingly, when I corrected another crash earlier.
+ I had formed the build_query routine, and failed to remove
+ mutex_unlock calls in 3 places in the transplanted code. These
+ unlocks would only happen in error situations, but unlocking the
+ mutex early set the code up for a catastrophic failure, it
+ appears. It would happen only once every 100K-200K or more calls,
+ under heavy load... but that is enough. If another crash occurs,
+ with the same MO, I'll come back and remove my confession from
+ the log, and we'll keep searching, but the fact that we have
+ Asterisk dying from an asynchronous wiping of the STMT object,
+ only on some connection error, and that the server has lived for
+ 2.5 days on this code without a crash, sure make it look like
+ this was the problem! Also, in several points, Statement handles
+ are set to NULL after SQLFreeHandle. This was mainly for
+ insurance, to guarantee a crash. As it turns out, the code does
+ not appear to be attempting to use these freed pointers. Asterisk
+ owes a debt of gratitude to Federico Alves and Frediano Ziglio
+ for their untiring efforts in finding this bug, among others.
+
+2008-10-15 21:34 +0000 [r149683-149840] BJ Weschke <bweschke@btwtech.com>
+
+ * CHANGES: Another documentation fix. (closes issue #13708)
+
+ * configs/agents.conf.sample: An update to the
+ documentation/example of agents.conf.sample with the correct
+ parameter for this feature as defined in chan_agent.c (closes
+ issue #13709)
+
+2008-10-15 10:30 +0000 [r149452] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: fix some problems when parsing SIP messages
+ that have the maximum number of headers or body lines that we
+ support
+
+2008-10-14 23:43 +0000 [r149130-149266] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Change this warning to an error message.
+ Suggestion comes from Sean Bright. Thanks Sean!
+
+ * channels/chan_sip.c: Call register_peer_exten even in the case
+ that the peer's IP/port does not change. (closes issue #13309)
+ Reported by: dimas Patches: v2-13309.patch uploaded by dimas
+ (license 88)
+
+ * include/asterisk/audiohook.h, main/audiohook.c: Add a tolerance
+ period for sync-triggered audiohooks so that if packetization of
+ audio is close (but not equal) we don't end up flushing the
+ audiohooks over small inconsistencies in synchronization. Related
+ to issue #13005, and solves the issue for most people who were
+ experiencing the problem. However, a small number of people are
+ still experiencing the problem on long calls, so I am not closing
+ the issue yet
+
+ * apps/app_queue.c: Update the queue with the correct number of
+ calls and whether the call was completed within the service level
+ when a transfer takes place. This way, we do not "break" the
+ leastrecent and fewestcalls strategies by not logging a call
+ until after the transferred call has ended. (closes issue #13395)
+ Reported by: Marquis Patches: app_queue.c.transfer.patch uploaded
+ by Marquis (license 32)
+
+ * channels/chan_sip.c: Don't allow reserved characters to be used
+ in register lines in sip.conf. (closes issue #13570) Reported by:
+ putnopvut
+
+2008-10-14 20:09 +0000 [r149061] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_waitforsilence.c: Check correct values in the return of
+ ast_waitfor(); also, get rid of a possible memory leak. (closes
+ issue #13658) Reported by: explidous Patch by: me
+
+2008-10-14 19:05 +0000 [r148990] Leif Madsen <lmadsen@digium.com>
+
+ * CHANGES: Add in some missing updates to the CHANGES file for
+ sip.conf (closes issue #13100) Reported and patch by:
+ gknispel_proformatique
+
+2008-10-14 19:03 +0000 [r148916-148987] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Some compilers warn, some don't. Fixing.
+
+ * apps/app_voicemail.c: Ensure that mail headers are 7-bit clean,
+ even when UTF-8 characters are used in headers like 'Subject' and
+ 'To'. Closes AST-107.
+
+2008-10-14 17:33 +0000 [r148912] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: Deadlock prevention in chan_local. (closes
+ issue #13676) Reported by: tacvbo Patches: 13676.patch uploaded
+ by putnopvut (license 60) Tested by: tacvbo
+
+2008-10-14 10:30 +0000 [r148611-148736] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: on Ubuntu (at least), recent versions of ld in binutils
+ delete all debugging symbols when -x is supplied; since the
+ reasons why -x is being passed are lost in the mists of time,
+ remove it so debugging will work properly
+
+ * main/translate.c: it would be nice if this message printing code
+ had actually been tested before it was committed...
+
+2008-10-10 16:25 +0000 [r147997-148257] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: User not notified of temporary greeting, if
+ ODBC storage is in use. (closes issue #13659) Reported by:
+ moliveras Patches: 20081009__bug13659.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: moliveras
+
+ * apps/app_voicemail.c: When blank, callerid name and number should
+ display "unknown caller" in voicemail emails. (Closes issue
+ #13643)
+
+2008-10-09 18:56 +0000 [r147941] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_features.c: (closes issue #13139) Reported by: krisk84
+ Tested by: krisk84 This change prevents a call that is placed in
+ the parkinglot to be picked up before the PBX is finished. If
+ another extension dials the parking extension before the PBX
+ thread has completed at minimum warnings will occur about the PBX
+ not properly being terminated. At worst, a crash could occur.
+
+2008-10-08 22:22 +0000 [r147681] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: when parsing a text configuration option,
+ ensure that the buffer on the stack is actually large enough to
+ hold the legal values of that option, and also ensure that
+ sscanf() knows to stop parsing if it would overrun the buffer
+ (without these changes, specifying "buffers=...,immediate" would
+ overflow the buffer on the stack, and could not have worked as
+ expected)
+
+2008-10-08 14:51 +0000 [r147517] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_speech_utils.c: If we receive DTMF make sure that the
+ state of the speech structure goes back to being not ready.
+ (issue #LUMENVOX-8)
+
+2008-10-07 23:14 +0000 [r147429-147430] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: revert this change until i can understand
+ why it results in locking order changes
+
+ * channels/chan_dahdi.c: don't start a PBX on incoming PRI call
+ channels until after we're done setting channel variables and
+ other things on the channel, otherwise the channel might go away
+ (if the dialplan hangs up quickly) before we are done, which
+ results in a spectacular crash
+
+2008-10-07 16:48 +0000 [r147193] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_voicemail.c: Make 'imapsecret' an alias to
+ 'imappassword' in voicemail.conf.
+
+2008-10-06 20:52 +0000 [r146711-146799] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_callerid.c, apps/app_speech_utils.c,
+ funcs/func_curl.c, funcs/func_groupcount.c, res/res_smdi.c,
+ channels/chan_sip.c, funcs/func_timeout.c, funcs/func_odbc.c,
+ funcs/func_cdr.c, funcs/func_math.c, channels/chan_iax2.c:
+ Dialplan functions should not actually return 0, unless they have
+ modified the workspace. To signal an error (and no change to the
+ workspace), -1 should be returned instead. (closes issue #13340)
+ Reported by: kryptolus Patches: 20080827__bug13340__2.diff.txt
+ uploaded by Corydon76 (license 14)
+
+ * channels/chan_local.c: Check whether an extension exists in the
+ _call method, rather than the _alloc method, because we need to
+ evaluate the callerid (since that data affects whether an
+ extension exists). (closes issue #13343) Reported by: efutch
+ Patches: 20080915__bug13343.diff.txt uploaded by Corydon76
+ (license 14) Tested by: efutch
+
+2008-10-06 15:57 +0000 [r146643] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: ensure that the private structure for
+ pseudo channels is created without 'leaking' configuration data
+ from other configured channels (closes issue #13555) Reported by:
+ jeffg Patches: issue_13555.patch uploaded by kpfleming (license
+ 421) Tested by: jeffg
+
+2008-10-05 21:17 +0000 [r146448] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: Fix silly formatting.
+
+2008-10-03 22:51 +0000 [r146244] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_rpt.c: Change some preprocessor macros to struct
+ definitions so that we get app_rpt to build with DAHDI. (closes
+ issue #13576) Reported by: blitzrage
+
+2008-10-03 20:44 +0000 [r146129] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/features.h, res/res_features.c, res/res_agi.c:
+ (closes issue #13425) Reported by: mdu113 Tested by: mdu113
+ Similar to r143204, masquerade the channel in the case of Park
+ being called from AGI.
+
+2008-10-03 17:12 +0000 [r146026] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: (closes issue #13579) Reported by: dwagner
+ (closes issue #13584) Reported by: dwagner Tested by: murf,
+ putnopvut The thought occurred to me that the res= from the
+ extension spawn was ending up being returned from the bridge.
+ "Thou shalt not poison the return value". Made the change and it
+ appears to allow blind xfers to work as normal. If I'm wrong,
+ reopen the bugs. But it looks good to me! Many thanks to
+ putnopvut for helping me reproduce this!
+
+2008-10-02 16:39 +0000 [r145751-145839] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_odbc.c: Backport support for some of the keyword
+ modifications used in 1.6 (while warning that some options aren't
+ really supported) and add some warning messages. Some credit to
+ oej, who was complaining in #asterisk-dev.
+
+ * res/res_odbc.c: Some sanity checks that may have led to prior
+ crashes, found by codefreeze-lap (murf) on IRC. Also some cleanup
+ of incorrectly-used constants.
+
+2008-10-01 17:18 +0000 [r145479] Leif Madsen <lmadsen@digium.com>
+
+ * contrib/scripts/realtime_pgsql.sql: Update the realtime_pgsql.sql
+ script to create the setinterfacevar column. (closes issue
+ #13549) Reported by: fiddur
+
+2008-10-01 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.22 released.
+
+2008-09-09 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.22-rc5 released.
+
+2008-09-09 15:40 +0000 [r142063] Russell Bryant <russell@digium.com>
+
+ * res/res_features.c: Ensure that the stored CDR reference is still
+ valid after the bridge before poking at it. Also, keep the
+ channel locked while messing with this CDR. (fixes crashes
+ reported in issue #13409)
+
+2008-09-08 21:10 +0000 [r141809] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix pedantic mode of chan_sip to only check
+ the remote tag of an endpoint once a dialog has been confirmed.
+ Up until that point, it is possible and legal for the far-end to
+ send provisional responses with a different To: tag each time.
+ With this patch applied, these provisional messages will not
+ cause a matching problem. (closes issue #11536) Reported by: ibc
+ Patches: 11536v2.patch uploaded by putnopvut (license 60)
+
+2008-09-08 21:02 +0000 [r141806] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: When doing an async goto, detect if the channel is
+ already in the middle of a masquerade. This can happen when
+ chan_local is trying to optimize itself out. If this happens,
+ fail the async goto instead of bursting into flames. (closes
+ issue #13435) Reported by: geoff2010
+
+2008-09-08 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.22-rc4 released.
+
+2008-09-08 20:15 +0000 [r141741] Jason Parker <jparker@digium.com>
+
+ * Makefile, redhat (removed): Remove RPM package targets from
+ Makefile (and all associated parts). This has never worked in
+ 1.4, and we decided that it makes no sense to be done here. There
+ are many distros out there that already have "proper" spec files
+ that can be (re)used. Closes issue #13113 Closes issue #10950
+ Closes issue #10952
+
+2008-09-08 16:26 +0000 [r141678] Russell Bryant <russell@digium.com>
+
+ * configure, configure.ac: Actually use Zaptel CFLAGS if using
+ Zaptel instead of DAHDI This fixes building against Zaptel when
+ using a custom path
+
+2008-09-06 20:13 +0000 [r141565] Steve Murphy <murf@digium.com>
+
+ * channels/chan_sip.c: This fix comes from Joshua Colp The
+ Brilliant, who, given the trace, came up with a solution. This
+ will most likely will close 13235 and 13409. I'll wait till
+ Monday to verify, and then close these bugs.
+
+2008-09-06 15:23 +0000 [r141503] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: Reverting behavior change (AGI should not exit
+ non-zero on SUCCESS) (closes issue #13434) Reported by:
+ francesco_r
+
+2008-09-05 21:10 +0000 [r141217-141366] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Agent's should not try to call a channel's
+ indicate callback if the channel has been hung up. It will likely
+ crash otherwise ABE-1159
+
+ * apps/app_voicemail.c: Since greetings are not stored in IMAP, we
+ should not be DISPOSE'ing of them the same way we do with other
+ messages. (closes issue #13414) Reported by: mthomasslo Patches:
+ 13414v2.patch uploaded by putnopvut (license 60) Tested by:
+ mthomasslo
+
+ * channels/chan_sip.c: Commit 140417 had a logic flaw in it which
+ caused port 5060 to always be used when dialing a peer if no
+ explicit port was specified. This broke the behavior of
+ implicitly using the port from which the peer registered if no
+ port is specified. This commit fixes the logic flaw. (closes
+ issue #13424) Reported by: mdu113 Patches: 13424.patch uploaded
+ by putnopvut (license 60) Tested by: mdu113
+
+2008-09-05 14:15 +0000 [r141094-141156] Steve Murphy <murf@digium.com>
+
+ * main/channel.c: A small change to prevent double-posting of
+ CDR's; thanks to Daniel Ferrer for bringing it to our attention
+
+ * pbx/ael/ael-test/ref.ael-vtest25 (added),
+ pbx/ael/ael-test/ael-vtest25/extensions.ael (added),
+ pbx/ael/ael-test/ael-vtest25 (added), pbx/ael/ael_lex.c,
+ pbx/ael/ael-test/ref.ael-test6, pbx/ael/ael.flex: (closes issue
+ #13357) Reported by: pj Tested by: murf (closes issue #13416)
+ Reported by: yarns Tested by: murf If you find this message
+ overly verbose, relax, it's probably not meant for you. This
+ message is meant for probably only two people in the whole world:
+ me, or the poor schnook that has to maintain this code because
+ I'm either dead or unavailable at the moment. This fix solves two
+ reports, both having to do with embedding a function call in a
+ ${} construct. It was tricky because the funccall syntax has
+ parenthesis () in it. And up till now, the 'word' token in the
+ flex stuff didn't allow that, because it would tend to steal the
+ LP and RP tokens. To be truthful, the "word" token was the
+ trickiest, most unstable thing in the whole lexer. I was lucky it
+ made this long without complaints. I had to choose every
+ character in the pattern with extreme care, and I knew that
+ someday I'd have to revisit it. Well, the day has come. So, my
+ brilliant idea (and I'm being modest), was to use the surrounding
+ ${} construct to make a state machine and capture everything in
+ it, no matter what it contains. But, I have to now treat the word
+ token like I did with comments, in that I turn the whole thing
+ into a state-machine sort of spec, with new contexts
+ "curlystate", "wordstate", and "brackstate". Wait a minute,
+ "brackstate"? Yes, well, it didn't take very many regression
+ tests to point out if I do this for ${} constructs, I also have
+ to do it with the $[] constructs, too. I had to create a separate
+ pcbstack2 and pcbstack3 because these constructs can occur inside
+ macro argument lists, and when we have two state machines
+ operating on the same structures we'd get problems otherwise. I
+ guess I could have stopped at pcbstack2 and had the brackstate
+ stuff share it, but it doesn't hurt to be safe. So, the pcbpush
+ and pcbpop routines also now have versions for "2" and "3". I had
+ to add the {KEYWORD} construct to the initial pattern for "word",
+ because previously word would match stuff like "default7",
+ because it was a longer match than the keyword "default". But,
+ not any more, because the word pattern only matches only one or
+ two characters now, and it will always lose. So, I made it the
+ winner again by making an optional match on any of the keywords
+ before it's normal pattern. I added another regression test to
+ make sure we don't lose this in future edits, and had to fix just
+ one regression, where it no longer reports a 'cascaded' error,
+ which I guess is a plus. I've given some thought as to whether to
+ apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I
+ decided to put it in 1.4 because one of the bug reports was
+ against 1.4; and it is unexpected that AEL cannot handle this
+ situation. It actually reduced the amount of useless "cascade"
+ error messages that appeared in the regressions (by one line,
+ ehhem). There is a possible side-effect in that it does now do
+ more careful checking of what's in those ${} constructs, as far
+ as matching parens, and brackets are concerned. Some users may
+ find a an insidious problem and correct it this way. This should
+ be exceedingly rare, I hope.
+
+2008-09-04 17:00 +0000 [r141028] Jeff Peeler <jpeeler@digium.com>
+
+ * res/res_features.c, res/res_agi.c: (closes issue #11979) Fixes
+ multiple parking problems: Crash when executing a park on an
+ extension dialed by AGI due to not returning the proper return
+ code. Crash when using a builtin feature that was a subset of a
+ enabled dynamic feature. Crash due to always hanging up the peer
+ despite the fact that the peer was supposed to be parked.
+
+2008-09-03 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.22-rc3 released.
+
+2008-09-03 14:29 +0000 [r140850] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Fix voicemail forwarding when using ODBC
+ storage. (closes issue #13387) Reported by: moliveras Patches:
+ 13387.patch uploaded by putnopvut (license 60) Tested by:
+ putnopvut, moliveras
+
+2008-09-03 13:24 +0000 [r140816] Russell Bryant <russell@digium.com>
+
+ * main/poll.c: Don't freak out if the poll emulation receives NULL
+ for the pollfds array (closes issue #13307) Reported by: jcovert
+
+2008-09-02 23:47 +0000 [r140751] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: After adding the context checking to
+ app_voicemail for IMAP storage, I left out a crucial place to
+ copy the context to the vm_state structure. This is the
+ correction.
+
+2008-09-02 23:36 +0000 [r140670-140747] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c: I am turning the warnings generated in ast_cdr_free
+ and post_cdr into verbose level 2 messages. Really, they matter
+ little to end users. You either get the CDR's you wanted, or you
+ don't, and it is a bug.
+
+ * main/channel.c: After reconsidering, with respect to 13409,
+ ast_cdr_detach should be OK, better in fact, than ast_cdr_free,
+ which generates lots of useless warnings that will undoubtably
+ generate complaints.
+
+ * main/channel.c, main/pbx.c: (closes issue #13409) Reported by:
+ tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by
+ tomaso (license 564) I basically spent the day, verifying that
+ this patch solves the problem, and doesn't hurt in non-problem
+ cases. Why valgrind did not plainly reveal this leak absolutely
+ mystifies and stuns me. Many, many thanks to tomaso for finding
+ and providing the fix.
+
+2008-09-02 18:14 +0000 [r140605] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_iax2.c: Make sure to use the correct length of the
+ mohinterpret and mohsuggest buffers when copying configuration
+ values. (closes issue #13336) Reported by:
+ decryptus_proformatique Patches:
+ chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded
+ by decryptus (license 555)
+
+2008-08-29 17:34 +0000 [r140417-140488] Mark Michelson <mmichelson@digium.com>
+
+ * main/manager.c, apps/app_queue.c, channels/chan_iax2.c: After
+ working on the ao2_containers branch, I noticed something a bit
+ strange. In all cases where we provide a callback function to
+ ao2_container_alloc, the callback function would only return 0 or
+ CMP_MATCH. After inspecting the ao2_callback() code carefully, I
+ found that if you're only looking for one specific item, then you
+ should return CMP_MATCH | CMP_STOP. Otherwise, astobj2 will
+ continue traversing the current bucket until the end searching
+ for more matches. In cases like chan_iax2 where in 1.4, all the
+ peers are shoved into a single bucket, this makes for potentially
+ terrible performance since the entire bucket will be traversed
+ even if the peer is one of the first ones come across in the
+ bucket. All the changes I have made were for cases where the
+ callback function defined was passed to ao2_container_alloc so
+ that calls to ao2_find could find a unique instance of whatever
+ object was being stored in the container.
+
+ * apps/app_voicemail.c: Add context checking when retrieving a
+ vm_state. This was causing a problem for people who had
+ identically named mailboxes in separate voicemail contexts. This
+ commit affects IMAP storage only. (closes issue #13194) Reported
+ by: moliveras Patches: 13194.patch uploaded by putnopvut (license
+ 60) Tested by: putnopvut, moliveras
+
+ * channels/chan_sip.c: Fix SIP's parsing so that if a port is
+ specified in a string to Dial(), it is not ignored. (closes issue
+ #13355) Reported by: acunningham Patches: 13355v2.patch uploaded
+ by putnopvut (license 60) Tested by: acunningham
+
+2008-08-27 19:49 +0000 [r140299] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix tag checking in get_sip_pvt_byid_locked
+ when in pedantic mode. The problem was that the wrong tags would
+ be compared depending on the direction of the call. (closes issue
+ #13353) Reported by: flefoll Patches:
+ chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll
+ (license 244)
+
+2008-08-26 16:49 +0000 [r140115] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: add HAVE_PRI if define around
+ dahdi_close_pri_fd
+
+2008-08-26 16:07 +0000 [r140060] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Fix some bogus scheduler usage in chan_sip.
+ This code used the return value of a completely unrelated
+ function to determine whether the scheduler should be run or not.
+ This would have caused the scheduler to not run in cases where it
+ should have. Also, leave a note about another scheduler issue
+ that needs to be addressed at some point.
+
+2008-08-26 15:57 +0000 [r140056] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: (closes issue #12071) Reported by: tzafrir
+ Patches: dahdi_close.diff uploaded by tzafrir (license 46) Tested
+ by: tzafrir, jpeeler This patch fixes closing open file
+ descriptors in the case of an error.
+
+2008-08-26 15:27 +0000 [r140051] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix a race condition with the IAX scheduler
+ thread. A lock and condition are used here to allow newly
+ scheduled tasks to wake up the scheduler just in case the new
+ task needs to run sooner than the current wakeup time when the
+ thread is sleeping. However, there was a race condition such that
+ a newly scheduled task would not properly wake up the scheduler
+ or affect the wake up period. The order of execution would have
+ been: 1) Scheduler thread determines wake up time of N ms. 2)
+ Another thread schedules a task and signals the condition, with
+ an execution time of < N ms. 3) Scheduler thread locks and goes
+ to sleep for N ms. By moving the sleep time determination to
+ inside the critical section, this possibility is avoided.
+
+2008-08-26 15:22 +0000 [r140050] Terry Wilson <twilson@digium.com>
+
+ * Makefile: sounds/Makefile installs sounds using the "new"
+ language directory structure, but languageprefix needs to be set
+ = yes for sounds in subdirectories (digits/1, etc.) to play as
+ the correct language. Fix the generation of asterisk.conf to
+ include languageprefix=yes
+
+2008-08-26 14:09 +0000 [r140029] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: correct a file location in an error
+ message
+
+2008-08-25 21:47 +0000 [r139927] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: Fix a typo I made. Lesson learned, apply the
+ patch if one exists.
+
+2008-08-25 21:31 +0000 [r139909] Sean Bright <sean.bright@gmail.com>
+
+ * build_tools/get_moduleinfo, build_tools/get_makeopts: Some
+ versions of awk (nawk, for example) don't like empty regular
+ expressions so be slightly more verbose. (closes issue #13374)
+ Reported by: dougm Patches: 13374.diff uploaded by seanbright
+ (license 71) Tested by: dougm
+
+2008-08-25 20:46 +0000 [r139869] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: Make SIPADDHEADER() propagate indefinitely
+
+2008-08-25 15:52 +0000 [r139769] Mark Michelson <mmichelson@digium.com>
+
+ * main/config.c: Fix the logic in config_text_file_save so that if
+ an UpdateConfig manager action is issued and the file specified
+ in DstFileName does not yet exist, an error is not returned.
+ (closes issue #13341) Reported by: vadim Patches: 13341.patch
+ uploaded by putnopvut (license 60) (with small modification from
+ seanbright)
+
+2008-08-25 15:33 +0000 [r139764] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, res/res_features.c: This patch reverts the changes
+ made via 139347, and 139635, as users are seeing adverse
+ difference. I will un-close 13251. Back to the drawing board/
+ concept/ beginning/ whatever!
+
+2008-08-22 22:24 +0000 [r139635] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: I found some problems with the code I
+ committed earlier, when I merged them into trunk, so I'm coming
+ back to clean up. And, in the process, I found an error in the
+ code I added to trunk and 1.6.x, that I'll fix using this patch
+ also.
+
+2008-08-22 21:36 +0000 [r139621] Jeff Peeler <jpeeler@digium.com>
+
+ * main/manager.c: (closes issue #13359) Reported by: Laureano
+ Patches: originate_channel_check.patch uploaded by Laureano
+ (license 265)
+
+2008-08-22 19:45 +0000 [r139456-139553] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/threadstorage.h: Fix compilation when
+ DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported
+ by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy
+ (license 35)
+
+ * main/frame.c: Remove show_frame_stats_deprecated since it is not
+ used anywhere and causes build errors if building under dev-mode
+ with TRACE_FRAMES selected in menuselect. (closes issue #13362)
+ Reported by: snuffy
+
+ * channels/chan_iax2.c: Fix the build. Thanks, mvanbaak!
+
+ * channels/chan_iax2.c: Prevent a deadlock in chan_iax2 resulting
+ from incorrect locking order between iax2_pvt and ast_channel
+ structures. AST-13
+
+2008-08-21 23:39 +0000 [r139387] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Fixes loop that could possibly never exit
+ in the event of a channel never being able to be opened or
+ specify after a restart. (closes issue #11017)
+
+2008-08-21 23:03 +0000 [r139347] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c, res/res_features.c: (closes issue #13251) Reported
+ by: sergee Tested by: murf THis is a bold move for a static
+ release fix, but I wouldn't have made it if I didn't feel
+ confident (at least a *bit* confident) that it wouldn't mess
+ everyone up. The reasoning goes something like this: 1. We simply
+ cannot do anything with CDR's at the current point (in pbx.c,
+ after the __ast_pbx_run loop). It's way too late to have any
+ affect on the CDRs. The CDR is already posted and gone, and the
+ remnants have been cleared. 2. I was very much afraid that moving
+ the running of the 'h' extension down into the bridge code (where
+ it would be now practical to do it), would result in a lot more
+ calls to the 'h' exten, so I implemented it as another exten
+ under another name, but found, to my pleasant surprise, that
+ there was a 1:1 correspondence to the running of the 'h' exten in
+ the pbx_run loop, and the new spot at the end of the bridge. So,
+ I ifdef'd out the current 'h' loop, and moved it into the bridge
+ code. The only difference I can see is the stuff about the
+ AST_PBX_KEEPALIVE, and hopefully, if this is still an important
+ decision point, I can replicate it if there are complaints. To be
+ perfectly honest, the KEEPALIVE situation is not totally clear to
+ me, and how it relates to a post-bridge situation is less clear.
+ I suspect the users will point out everything in total clarity if
+ this steps on anyone's toes! 3. I temporarily swap the bridge_cdr
+ into the channel before running the 'h' exten, which makes it
+ possible for users to edit the cdr before it goes out the door.
+ And, of course, with the endbeforehexten config var set, the
+ users can also get at the billsec/duration vals. After the h
+ exten finishes, the cdr is swapped back and processing continues
+ as normal. Please, all who deal with CDR's, please test this
+ version of Asterisk, and file bug reports as appropriate!
+
+2008-08-21 10:11 +0000 [r139283] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c: Apply fix for issue #13310 to branch 1.4,
+ too.
+
+2008-08-20 22:14 +0000 [r139213] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: Fix a crash in the ChanSpy application. The
+ issue here is that if you call ChanSpy and specify a spy group,
+ and sit in the application long enough looping through the
+ channel list, you will eventually run out of stack space and the
+ application with exit with a seg fault. The backtrace was always
+ inside of a harmless snprintf() call, so it was tricky to track
+ down. However, it turned out that the call to snprintf() was just
+ the biggest stack consumer in this code path, so it would always
+ be the first one to hit the boundary. (closes issue #13338)
+ Reported by: ruddy
+
+2008-08-20 19:52 +0000 [r139151] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c: Fix bug where the samples were not accurate
+ when in G723 mode, which would cause the timestamp field of the
+ RTP header to be invalid.
+
+2008-08-20 19:35 +0000 [r139145] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Backport support
+ for Zaptel/DAHDI channel-level alarms from trunk/1.6, because not
+ doing so just makes it difficult for people with channels that
+ are in alarm when Asterisk starts up to get them going once the
+ alarm is cleared (closes issue #12160) Reported by: tzafrir
+ Patches: asterisk-chanalarms_14.patch uploaded by tzafrir
+ (license 46) Tested by: tzafrir
+
+2008-08-20 17:14 +0000 [r139074] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c: (closes issue #13263) Reported by: brainy Tested by:
+ murf The specialized reset routine is tromping on the flags field
+ of the CDR. I made a change to not reset the DISABLED bit. This
+ should get rid of this problem.
+
+2008-08-20 15:37 +0000 [r139015] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: sip_read should properly handle a NULL
+ return from sip_rtp_read. (closes issue #13257) Reported by:
+ travishein
+
+2008-08-19 23:22 +0000 [r138949] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/dahdi_compat.h: add DAHDI_POLICY_WHEN_FULL
+ compatability define for Zaptel
+
+2008-08-19 23:17 +0000 [r138942] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Reset agent_pvt variables back to the
+ values in agents.conf (from what the corresponding channel
+ variables were set to) when the agent logs out. (closes issue
+ #13098) Reported by: davidw Patches:
+ 20080731__issue13098_agent_ackcall_not_reset.diff uploaded by
+ bbryant (license 36) Tested by: davidw
+
+2008-08-19 22:56 +0000 [r138938] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Add configuration option to
+ chan_dahdi.conf to allow buffering policy and number of buffers
+ to be configured per channel. Syntax: buffers=<num of
+ buffers>,<policy> Where the number of buffers is some
+ non-negative integer and the policy is either "full", "half", or
+ "immediate".
+
+2008-08-19 18:50 +0000 [r138685-138886] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Add a lock and unlock prior to the
+ destruction of the chanspy_ds lock to ensure that no other
+ threads still have it locked. While this should not happen under
+ normal circumstances, it appears that if the spyer and spyee hang
+ up at nearly the same time, the following may occur. 1.
+ ast_channel_free is called on the spyee's channel. 2. The chanspy
+ datastore is removed from the spyee's channel in
+ ast_channel_free. 3. In the spyer's thread, the spyer attempts to
+ remove and destroy the datastore from the spyee channel, but the
+ datastore has already been removed in step 2, so the spyer
+ continues in the code. 4. The spyee's thread continues and calls
+ the datastore's destroy callback, chanspy_ds_destroy. This
+ involves locking the chanspy_ds. 5. Now the spyer attempts to
+ destroy the chanspy_ds lock. The problem is that in step 4, the
+ spyee has locked this lock, meaning that the spyer is attempting
+ to destroy a lock which is currently locked by another thread.
+ The backtrace provided in issue #12969 supports the idea that
+ this is possible (and has even occurred). This commit does not
+ close the issue, but should help in preventing one type of crash
+ associated with the use of app_chanspy.
+
+ * apps/app_queue.c: Change the inequalities used in app_queue with
+ regards to timeouts from being strict to non-strict for more
+ accuracy. (closes issue #13239) Reported by: atis Patches:
+ app_queue_timeouts_v2.patch uploaded by atis (license 242)
+
+2008-08-18 16:57 +0000 [r138663] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/codec_dahdi.c: look for transcoder in proper place based
+ on build against Zaptel or DAHDI
+
+2008-08-18 11:57 +0000 [r138569] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_dahdi.c: You know what's awesome? Code that
+ compiles... ;)
+
+2008-08-18 02:05 +0000 [r138516] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: fix compilation warnings
+
+2008-08-16 01:12 +0000 [r138309-138360] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: fixes use count to properly decrement if
+ an active dahdi channel is destroyed allowing module to be
+ unloaded
+
+ * channels/chan_dahdi.c: add forgotten locks around ss_thread_count
+ in ss_thread for dahdi restart
+
+2008-08-15 22:33 +0000 [r138258] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample: More fixes for
+ realtime peers. (closes issue #12921) Reported by: Nuitari
+ Patches: 20080804__bug12921.diff.txt uploaded by Corydon76
+ (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76
+ (license 14) Tested by: Corydon76
+
+2008-08-15 21:28 +0000 [r138119-138238] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: initialize condition variable
+ ss_thread_complete using ast_cond_init
+
+ * channels/chan_dahdi.c: declared static mutexes using
+ AST_MUTEX_DEFINE_STATIC macro
+
+ * channels/chan_dahdi.c: Fixes the dahdi restart functionality.
+ Dahdi restart allows one to restart all DAHDI channels, even if
+ they are currently in use. This is different from unloading and
+ then loading the module since unloading requires the use count to
+ be zero. Reloading the module is different in that the signalling
+ is not changed from what it was originally configured. Also, this
+ fixes not closing all the file descriptors for D-channels upon
+ module unload (which would prevent loading the module
+ afterwards). (closes issue #11017)
+
+2008-08-15 15:07 +0000 [r138027] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c: Ensure that when a hangup occurs in
+ autoservice, that a hangup frame gets properly deferred to be
+ read from the channel owner when it gets taken out of
+ autoservice. (closes issue #12874) Reported by: dimas Patches:
+ v1-12874.patch uploaded by dimas (license 88)
+
+2008-08-15 14:51 +0000 [r137847-138023] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c: Additional check for more string specifiers
+ than arguments. (closes issue #13299) Reported by: adomjan
+ Patches: 20080813__bug13299.diff.txt uploaded by Corydon76
+ (license 14) func_strings.c-sprintf.patch uploaded by adomjan
+ (license 487) Tested by: adomjan
+
+ * channels/chan_dahdi.c: Oops, wrong direction
+
+ * channels/chan_dahdi.c: When creating the secondary subchannel
+ name, it is necessary to compare to the existing channel name
+ without the "Zap/" or "DAHDI/" prefix, since our test string is
+ also without that prefix. (closes issue #13027) Reported by:
+ dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer
+ (license 525) (Slightly modified by me, to compensate for both
+ names)
+
+2008-08-14 14:05 +0000 [r137731] Russell Bryant <russell@digium.com>
+
+ * configs/sip.conf.sample: Comments in this config file were
+ aligned only if your tab size was set to 8. So, convert tabs to
+ spaces so that things should be aligned regardless of what tab
+ size you use in your editor.
+
+2008-08-14 02:03 +0000 [r137677-137679] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Zaptel-to-DAHDI.txt: forgot one module name that changed
+
+ * include/asterisk/dahdi_compat.h, channels/chan_dahdi.c,
+ build_tools/menuselect-deps.in, configure, configure.ac,
+ codecs/codec_dahdi.c: add support for Zaptel versions that
+ contain the new transcoder interface
+
+2008-08-13 21:35 +0000 [r137580] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Register DAHDISendKeypadFacility
+ application if dahdi_chan_mode is set to DAHDI + Zap. Mark
+ ZapSendKeypadFacility application as deprecated on usage.
+
+2008-08-13 20:46 +0000 [r137527-137530] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Zaptel-to-DAHDI.txt (added): add document describing what users
+ will need to be aware of when upgrading to this version and using
+ DAHDI
+
+ * apps/app_meetme.c: remove some more chan_zap references
+
+ * doc/asterisk-conf.txt, channels/chan_dahdi.c: document
+ dahdichanname option in doc/asterisk-conf.txt make chan_dahdi
+ read its configuration from zapata.conf if dahdichanname has been
+ set to 'no'
+
+2008-08-13 14:33 +0000 [r137348-137405] Sean Bright <sean.bright@gmail.com>
+
+ * doc/cdrdriver.txt: Update docs to reflect the change to cdr_tds
+
+ * cdr/cdr_tds.c: Bring cdr_tds in line with the other CDR backends
+ and have it try to store CDR(userfield) if it is set. The new
+ behavior is to check for the userfield column on module load, and
+ if it exists, we will store CDR(userfield) when CDRs are written.
+ A similar patch already went into trunk and 1.6.0. (closes issue
+ #13290) Reported by: falves11
+
+2008-08-11 13:33 +0000 [r137188] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_meetme.c: convert this module to be able to handle DAHDI
+ or Zaptel (reported on asterisk-users, don't know how this got
+ missed before)
+
+2008-08-11 00:20 +0000 [r137138] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: Deallocate database connection handle on
+ disconnect, as we allocate another one on connect. (closes issue
+ #13271) Reported by: dveiga
+
+2008-08-09 17:11 +0000 [r136999] Russell Bryant <russell@digium.com>
+
+ * configure, configure.ac: Ensure PBX_DAHDI_TRANSCODE will evaluate
+ to 0 if not found instead of empty. pointed out by tzafrir on
+ #asterisk-dev
+
+2008-08-09 15:25 +0000 [r136946] Tilghman Lesher <tlesher@digium.com>
+
+ * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged
+ revisions 136945 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008)
+ | 2 lines Regression fixes for Solaris ........
+
+2008-08-08 00:15 +0000 [r136726] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
+ pbx/ael/ael-test/ref.ael-vtest13,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/pbx_ael.c,
+ include/asterisk/ael_structs.h: (closes issue #13236) Reported
+ by: korihor Wow, this one was a challenge! I regrouped and ran a
+ new strategy for setting the ~~MACRO~~ value; I set it once per
+ extension, up near the top. It is only set if there is a switch
+ in the extension. So, I had to put in a chunk of code to detect a
+ switch in the pval tree. I moved the code to insert the set of
+ ~~exten~~ up to the beginning of the gen_prios routine, instead
+ of down in the switch code. I learned that I have to push the
+ detection of the switches down into the code, so everywhere I
+ create a new exten in gen_prios, I make sure to pass onto it the
+ values of the mother_exten first, and the exten next. I had to
+ add a couple fields to the exten struct to accomplish this, in
+ the ael_structs.h file. The checked field makes it so we don't
+ repeat the switch search if it's been done. I also updated the
+ regressions.
+
+2008-08-07 18:25 +0000 [r136560] Kevin P. Fleming <kpfleming@digium.com>
+
+ * build_tools/menuselect-deps.in, configure, configure.ac: change
+ the required dependency for codec_dahdi to only be satisfied by
+ DAHDI and not Zaptel, as the new transcoder interface is only in
+ DAHDI
+
+2008-08-07 18:14 +0000 [r136544] Shaun Ruffell <sruffell@digium.com>
+
+ * codecs/codec_dahdi.c: Updated codec_dahdi to use the new
+ transcoder interface in the first DAHDI release. Codec dahdi no
+ longer functions with the transcoder interface in zaptel at this
+ time (which the last zaptel release was 1.4.11). NOTE: Still
+ needs an update to the configure script to make sure that
+ codec_dahdi is only built if the new transcoder interface is
+ present in the drivers. (Issue: DAHDI-42)
+
+2008-08-07 16:50 +0000 [r136488] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: Update persistent state on all exit conditions.
+ (closes issue #12916) Reported by: sgenyuk Patches:
+ app_queue.patch.txt uploaded by neutrino88 (license 297) Tested
+ by: sgenyuk, aragon
+
+2008-08-07 16:30 +0000 [r136404-136484] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/doxyref.h: add a raw list of all libraries that
+ any part of Asterisk links directly to
+
+ * apps/app_voicemail.c: work around a bug in gcc-4.2.3 that
+ incorrectly ignores the casting away of 'const' for pointers when
+ the developer knows it is safe to do so
+
+ * Makefile: remove config.cache during distclean, in case the user
+ is using autoconf caching
+
+2008-08-07 01:31 +0000 [r136304-136348] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Also, parse
+ useincomingcalleridonzaptransfer (and add appropriate deprecation
+ warnings).
+
+ * channels/chan_dahdi.c: For backwards compatibility with previous
+ 1.4 versions which used "zapchan" in users.conf, ensure that we
+ still support it.
+
+2008-08-06 21:18 +0000 [r136241] Richard Mudgett <rmudgett@digium.com>
+
+ * channels/misdn_config.c, channels/chan_misdn.c,
+ configs/misdn.conf.sample: * The allowed_bearers setting in
+ misdn.conf misspelled one of its options: digital_restricted. *
+ Fixed some other spelling errors and typos.
+
+2008-08-06 20:42 +0000 [r136238] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: We only need to unregister the QueueStatus
+ manager command once on an unload
+
+2008-08-06 20:14 +0000 [r136190] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.redhat.asterisk: -C option takes a filename,
+ not a directory path. (closes issue #13007) Reported by:
+ klaus3000
+
+2008-08-06 18:58 +0000 [r136168] Russell Bryant <russell@digium.com>
+
+ * Makefile: Remove the use of --no-print-directory when compiling
+ subdirectories. This allows vim :make functionality to work
+ properly when errors have occurred in the build. Without printing
+ the directories, vim did not know how to find the file that the
+ error occurred in. If the extra bit of build noise annoys anyone,
+ just let me know, and I'll make this optional.
+
+2008-08-06 15:58 +0000 [r136062] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c, channels/chan_skinny.c: Since adding the
+ AST_CONTROL_SRCUPDATE frame type, there are places where
+ ast_rtp_new_source may be called where the tech_pvt of a channel
+ may not yet have an rtp structure allocated. This caused a crash
+ in chan_skinny, which was fixed earlier, but now the same crash
+ has been reported against chan_h323 as well. It seems that the
+ best solution is to modify ast_rtp_new_source to not attempt to
+ set the marker bit if the rtp structure passed in is NULL. This
+ change to ast_rtp_new_source also allows the removal of what is
+ now a redundant pointer check from chan_skinny. (closes issue
+ #13247) Reported by: pj
+
+2008-08-06 03:53 +0000 [r135899-135949] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c: Fix a longstanding bug in channel walking logic,
+ and fix the explanation to make sense. (Closes issue #13124)
+
+ * main/translate.c: Since powerof() can return an error condition,
+ it's foolhardy not to detect and deal with that condition.
+ (Related to issue #13240)
+
+ * include/asterisk/threadstorage.h, include/asterisk/utils.h: 1)
+ Bugfix for debugging code 2) Reduce compiler warnings for another
+ section of debugging code (Closes issue #13237)
+
+2008-08-06 00:29 +0000 [r135841-135850] Mark Michelson <mmichelson@digium.com>
+
+ * /: Remove properties that should not be here
+
+ * apps/app_skel.c: Revert inadvertent changes to app_skel that
+ occurred when I was testing for a memory leak
+
+ * include/asterisk/abstract_jb.h, main/channel.c, /,
+ apps/app_skel.c, main/abstract_jb.c, main/fixedjitterbuf.h:
+ Merging the issue11259 branch. The purpose of this branch was to
+ take into account "burps" which could cause jitterbuffers to
+ misbehave. One such example is if the L option to Dial() were
+ used to inject audio into a bridged conversation at regular
+ intervals. Since the audio here was not passed through the
+ jitterbuffer, it would cause a gap in the jitterbuffer's
+ timestamps which would cause a frames to be dropped for a brief
+ period. Now ast_generic_bridge will empty and reset the
+ jitterbuffer each time it is called. This causes injected audio
+ to be handled properly. ast_generic_bridge also will empty and
+ reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE
+ frame since the change in audio source could negatively affect
+ the jitterbuffer. All of this was made possible by adding a new
+ public API call to the abstract_jb called ast_jb_empty_and_reset.
+ (closes issue #11259) Reported by: plack Tested by: putnopvut
+
+2008-08-05 23:13 +0000 [r135799] Steve Murphy <murf@digium.com>
+
+ * apps/app_dial.c, main/cdr.c, main/channel.c, res/res_features.c,
+ include/asterisk/cdr.h: (closes issue #12982) Reported by: bcnit
+ Tested by: murf I discovered that also, in the previous bug fixes
+ and changes, the cdr.conf 'unanswered' option is not being
+ obeyed, so I fixed this. And, yes, there are two 'answer' times
+ involved in this scenario, and I would agree with you, that the
+ first answer time is the time that should appear in the CDR. (the
+ second 'answer' time is the time that the bridge was begun). I
+ made the necessary adjustments, recording the first answer time
+ into the peer cdr, and then using that to override the bridge
+ cdr's value. To get the 'unanswered' CDRs to appear, I purposely
+ output them, using the dial cmd to mark them as DIALED (with a
+ new flag), and outputting them if they bear that flag, and you
+ are in the right mode. I also corrected one small mention of the
+ Zap device to equally consider the dahdi device. I heavily tested
+ 10-sec-wait macros in dial, and without the macro call; I tested
+ hangups while the macro was running vs. letting the macro
+ complete and the bridge form. Looks OK. Removed all the
+ instrumentation and debug.
+
+2008-08-05 21:34 +0000 [r135747] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: In a conversion to use ast_strlen_zero, the
+ meaning of the flag IAX_HASCALLERID was perverted. This change
+ reverts IAX2 to the original meaning, which was, that the
+ callerid set on the client should be overridden on the server,
+ even if that means the resulting callerid is blank. In other
+ words, if you set "callerid=" in the IAX config, then the
+ callerid should be overridden to blank, even if set on the
+ client. Note that there's a distinction, even on realtime,
+ between the field not existing (NULL in databases) and the field
+ existing, but set to blank (override callerid to blank).
+
+2008-08-05 13:25 +0000 [r135597] Sean Bright <sean.bright@gmail.com>
+
+ * main/cli.c: Use PATH_MAX for filenames
+
+2008-08-04 20:15 +0000 [r135536] Russell Bryant <russell@digium.com>
+
+ * configs/chan_dahdi.conf.sample: fix a config sample typo
+
+2008-08-04 17:07 +0000 [r135479-135482] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.mandrake.asterisk: Define ASTSBINDIR for script
+
+ * apps/app_voicemail.c: Memory leak on unload (closes issue #13231)
+ Reported by: eliel Patches: app_voicemail.leak.patch uploaded by
+ eliel (license 64)
+
+2008-08-04 16:26 +0000 [r135473] Russell Bryant <russell@digium.com>
+
+ * configs/chan_dahdi.conf.sample: Add a minor clarification to the
+ documentation of mohinterpret and mohsuggest
+
+2008-08-01 11:43 +0000 [r135055-135058] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_ices.c: make app_ices compile on OpenBSD.
+
+ * channels/chan_skinny.c: fix some potential deadlocks in
+ chan_skinny (closes issue #13215) Reported by: qwell Patches:
+ 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7)
+ Tested by: mvanbaak
+
+2008-07-31 22:18 +0000 [r134983] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/http.c: accomodate users who seem to lack a sense of humor
+ :-)
+
+2008-07-31 21:53 +0000 [r134976] Tilghman Lesher <tlesher@digium.com>
+
+ * sample.call, main/manager.c, pbx/pbx_spool.c: Specify codecs in
+ callfiles and manager, to allow video calls to be set up from
+ callfiles and AMI. (closes issue #9531) Reported by: Geisj
+ Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76
+ (license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: Corydon76
+
+2008-07-31 19:37 +0000 [r134915] Russell Bryant <russell@digium.com>
+
+ * apps/app_ices.c: Get app_ices working again (closes issue #12981)
+ Reported by: dlogan Patches:
+ 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant
+ (license 36) 20080709__app_ices_v2_update_14.diff uploaded by
+ bbryant (license 36) Tested by: bbryant
+
+2008-07-31 19:23 +0000 [r134883] Steve Murphy <murf@digium.com>
+
+ * res/res_features.c: (closes issue #11849) Reported by: greyvoip
+ Tested by: murf OK, a few days of debugging, a bunch of
+ instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and
+ 5 solid notebook pages of notes later, I have made the small
+ tweek necc. to get the start time right on the second CDR when: A
+ Calls B B answ. A hits Xfer button on sip phone, A dials C and
+ hits the OK button, A hangs up C answers ringing phone B and C
+ converse B and/or C hangs up But does not harm the scenario
+ where: A Calls B B answ. B hits xfer button on sip phone, B dials
+ C and hits the OK button, B hangs up C answers ringing phone A
+ and C converse A and/or C hangs up The difference in start times
+ on the second CDR is because of a Masquerade on the B channel
+ when the xfer number is sent. It ends up replacing the CDR on the
+ B channel with a duplicate, which ends up getting tossed out. We
+ keep a pointer to the first CDR, and update *that* after the
+ bridge closes. But, only if the CDR has changed. I hope this
+ change is specific enough not to muck up any current CDR-based
+ apps. In my defence, I assert that the previous information was
+ wrong, and this change fixes it, and possibly other similar
+ scenarios. I wonder if I should be doing the same thing for the
+ channel, as I did for the peer, but I can't think of a scenario
+ this might affect. I leave it, then, as an exersize for the
+ users, to find the scenario where the chan's CDR changes and
+ loses the proper start time.
+
+2008-07-31 16:45 +0000 [r134814] Russell Bryant <russell@digium.com>
+
+ * channels/iax2-parser.c: In case we have some processing threads
+ that free more frames than they allocate, do not let the frame
+ cache grow forever. (closes issue #13160) Reported by: tavius
+ Tested by: tavius, russell
+
+2008-07-31 15:56 +0000 [r134758] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Add more timeout checks into app_queue,
+ specifically targeting areas where an unknown and potentially
+ long time has just elapsed. Also added a check to try_calling()
+ to return early if the timeout has elapsed instead of potentially
+ setting a negative timeout for the call (thus making it have *no*
+ timeout at all). (closes issue #13186) Reported by:
+ miquel_cabrespina Patches: 13186.diff uploaded by putnopvut
+ (license 60) Tested by: miquel_cabrespina
+
+2008-07-30 22:39 +0000 [r134704] Tilghman Lesher <tlesher@digium.com>
+
+ * main/sched.c, include/asterisk/sched.h: Oops, wrong define
+
+2008-07-30 22:02 +0000 [r134652] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_ael.c: (closes issue #13197) Reported by: pj (closes
+ issue #13051) Reported by: pj This patch substitutes commas in
+ the expr supplied to the if () statement, as in if ( expr ) ...
+ This solves both the bugs above, and makes the source symmetric
+ with switch statements, which were earlier reported to need this
+ sort of treatment. I tested this using the examples, both for the
+ compiler and at run time. Looks good.
+
+2008-07-30 21:38 +0000 [r134649] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Qwell pointed out, via IRC, that the
+ previous fix only worked when explicitly set. When nothing is
+ set, and the option is implied, it breaks, because configure sets
+ the prefix to 'NONE'. Fixing.
+
+2008-07-30 20:37 +0000 [r134540-134595] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_dundi.c: Reduce stack consumption by 12.5% of the max
+ stack size to fix a crash when compiled with LOW_MEMORY. (closes
+ issue #13154) Reported by: edantie
+
+ * funcs/func_curl.c: Fix a memory leak in func_curl. Every thread
+ that used this function leaked an allocation the size of a
+ pointer. (reported by jmls in #asterisk-dev)
+
+2008-07-30 19:47 +0000 [r134480-134536] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, configure.ac: Only override sysconfdir and mandir when
+ prefix=/usr (closes issue #13093) Reported by: pabelanger
+
+ * res/res_agi.c: launch_netscript sometimes returns -1, which fails
+ to set AGISTATUS. Map failure to -1, so that AGISTATUS is always
+ set. (closes issue #13199) Reported by: smw1218
+
+2008-07-30 18:31 +0000 [r134475] Mark Michelson <mmichelson@digium.com>
+
+ * main/app.c: Fix a spot where a function could return without
+ bringing a channel out of autoservice.
+
+2008-07-30 15:29 +0000 [r134254-134352] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: use the proper method for building version.h
+
+ * include/asterisk/dahdi_compat.h, apps/app_dahdibarge.c,
+ channels/chan_dahdi.c, apps/app_meetme.c, apps/app_flash.c,
+ apps/app_dahdiscan.c, apps/app_dahdiras.c, codecs/codec_dahdi.c:
+ build against the now-typedef-free dahdi/user.h
+
+2008-07-29 15:54 +0000 [r134223] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Merging the imap_consistency branch. The
+ main aim of this branch was to make the IMAP code function in the
+ same manner as the ODBC code does, eliminating the need for so
+ many IMAP-specific code chunks. The focal point of all of this
+ work was to make the various macros (e.g. RETRIEVE, DISPOSE)
+ functionally equivalent. While doing the above work, I also fixed
+ a few bugs that I came across in my testing. Among these were 1.
+ Fixed message forwarding. This was completely broken when using
+ IMAP. 2. Fixed the inability to save new messages as old and vice
+ versa. 3. Fixed the "delete" options in voicemail.conf when using
+ IMAP storage. Even though a few bugs were fixed and the code is a
+ lot more consistent, the one thing that was *not* improved in
+ this branch was performance. The merge of this to trunk may not
+ come immediately due to the amount of work it will probably
+ involve. (closes issue #12764) Reported by: balsamcn
+
+2008-07-28 21:50 +0000 [r134161] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Detect when sox fails to raise the volume,
+ because sox can't read the file. (closes issue #12939) Reported
+ by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: rickbradley
+
+2008-07-26 15:31 +0000 [r133980] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c, include/asterisk/doxyref.h: Add the licensing
+ section to the docs in 1.4, as well, so that we can work on
+ having an accurate list for each version of Asterisk that is
+ supported
+
+2008-07-25 18:00 +0000 [r133649-133709] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Remove unnecessary mmap flag (Closes issue
+ #13161)
+
+ * main/channel.c, channels/chan_agent.c, main/devicestate.c: Fix
+ some errant device states by making the devicestate API more
+ strict in terms of the device argument (only without the unique
+ identifier appended). (closes issue #12771) Reported by: davidw
+ Patches: 20080717__bug12771.diff.txt uploaded by Corydon76
+ (license 14) Tested by: davidw, jvandal, murf
+
+2008-07-25 15:00 +0000 [r133578] Russell Bryant <russell@digium.com>
+
+ * /, LICENSE: Merged revisions 133577 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008)
+ | 2 lines Fix the IAX2 URI for calling Digium ........
+
+2008-07-25 14:40 +0000 [r133572] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: We need to make sure to null-terminate the
+ "name" portion of SIP URI parameters so that there are no bogus
+ comparisons. Thanks to bbryant for pointing this out.
+
+2008-07-24 21:17 +0000 [r133361-133488] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Fix rtautoclear and rtcachefriends (Closes
+ issue #12707)
+
+ * /: Blocked revisions 133360 via svnmerge ........ r133360 |
+ tilghman | 2008-07-23 22:46:01 -0500 (Wed, 23 Jul 2008) | 2 lines
+ This part was not correctly patched for AST-2008-010. ........
+
+2008-07-23 21:49 +0000 [r133295] Jason Parker <jparker@digium.com>
+
+ * channels/chan_dahdi.c: inbandrelease is gone - it's now
+ inbanddisconnect
+
+2008-07-23 21:05 +0000 [r133226-133237] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/stringfields.h, main/utils.c: revert an
+ optimization that broke ABI... thanks russell!
+
+ * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
+ apps/app_dahdibarge.c, channels/chan_dahdi.c,
+ apps/app_dahdiras.c: make some more changes to the dahdi/zap
+ channel name support stuff to ensure allthe globals are 'const',
+ and clean up mmichelson's changes to app_chanspy to simplify the
+ code
+
+2008-07-23 19:39 +0000 [r132974-133169] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c,
+ channels/chan_dahdi.c: As suggested by seanbright, the
+ PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at
+ compile time, since dahdi_chan_name is determined at load time.
+ Also changed the next_unique_id_to_use to have the static
+ qualifier. Also added the dahdi_chan_name_len variable so that
+ strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for
+ the suggestion.
+
+ * apps/app_chanspy.c: Zap/pseudo is ten characters, but
+ DAHDI/pseudo is twelve. The strncmp call in next_channel should
+ account for this.
+
+ * apps/app_chanspy.c: Update the "last" channel in next_channel in
+ app_chanspy so that the same pseudo channel isn't constantly
+ returned. related to issue #13124
+
+ * channels/chan_dahdi.c: Small cleanup. Move the declaration of the
+ DAHDI_SPANINFO variable to the block where it is used. This
+ allows one less #ifdef HAVE_PRI to clutter things up. Thanks to
+ Tzafrir for pointing this out on #asterisk-dev
+
+ * channels/chan_dahdi.c: Fix building of chan_dahdi when HAVE_PRI
+ is not defined.
+
+2008-07-23 15:52 +0000 [r132872-132942] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: ensure that after a channel is created, if
+ it happened to be in 'channel alarm' state, when that alarm
+ clears we won't generate a spurious 'alarm cleared' message
+ (closes issue #12160) Reported by: tzafrir
+
+ * include/asterisk/stringfields.h, main/utils.c: minor optimization
+ for stringfields: when a field is being set to a larger value
+ than it currently contains and it happens to be the most recent
+ field allocated from the currentl pool, it is possible to 'grow'
+ it without having to waste the space it is currently using (or
+ potentially even allocate a new pool)
+
+2008-07-23 11:37 +0000 [r132826] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c: another Fix because of r119585, this
+ commit has broken high frequented BRI Ports, there was a
+ possibility that a channel, that was marked as in_use would be
+ reused later, the corresponding port could got stuck then. So it
+ is recommended to upgrade for chan_misdn users.
+
+2008-07-22 22:14 +0000 [r132790] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Allow Spiraled INVITEs to work correctly
+ within Asterisk. Prior to this change, a spiraled INVITE would
+ cause a 482 Loop Detected to be sent to the caller. With this
+ change, if a potential loop is detected, the Request-URI is
+ inspected to see if it has changed from what was originally
+ received. If pedantic mode is on, then this inspection is fully
+ RFC 3261 compliant. If pedantic mode is not on, then a string
+ comparison is used to test the equality of the two R-URIs. This
+ has been tested by using OpenSER to rewrite the R-URI and send
+ the INVITE back to Asterisk. (closes issue #7403) Reported by:
+ stephen_dredge
+
+2008-07-22 22:11 +0000 [r132784-132787] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c,
+ apps/app_dahdibarge.c, channels/chan_dahdi.c, apps/app_flash.c,
+ apps/app_dahdiras.c: fix up namespace pollution for
+ dahdi_chan_mode enum correct registration of AMI actions in
+ chan_dahdi; in zap-only mode, only register the Zap flavors of
+ the actions (and use Zap prefixes for headers and acks), but in
+ dahdi+zap mode, register both Zap and DAHDI flavors of actions
+
+ * Makefile.rules: add rules to create preprocessor output... useful
+ for debugging macros
+
+2008-07-22 21:19 +0000 [r132713] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged
+ revisions 132711 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008)
+ | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........
+
+2008-07-22 21:17 +0000 [r132704-132712] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: ensure that if any alarms exist at channel
+ creation time, they are handled identically to if they occurred
+ later, so that later alarm clearing will work properly and 'make
+ sense' (closes issue #12160) Reported by: tzafrir
+
+ * configure, configure.ac, acinclude.m4: make AST_C_COMPILE_CHECK
+ able to print a 'pretty' description of what it is doing
+
+2008-07-22 20:10 +0000 [r132645] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c, doc/sip-retransmit.txt (added): The most
+ common question on the #asterisk iRC channel and on mailing lists
+ seems to be in regards to an error message when retransmit fails.
+ This is frequently misunderstood as a failure of Asterisk, not a
+ failure of the network to reach the other party. This document
+ tries to assist the Asterisk user in sorting out these issues by
+ explaining the logic and pointing at some possible causes.
+ Hopefully, we will get other questions now :-)
+
+2008-07-22 19:57 +0000 [r132571-132642] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: correct wording in comment
+
+ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: use renamed
+ libpri API call for controlling this feature (was improperly
+ named before)
+
+ * channels/chan_dahdi.c: teach chan_dahdi how to find the D-channel
+ on BRI spans, and don't attempt to use channel 24 as a D-channel
+ on spans of unexpected sizes
+
+2008-07-21 20:51 +0000 [r132506-132507] Brett Bryant <bbryant@digium.com>
+
+ * apps/app_sendtext.c: Fix a bug where SENDTEXTSTATUS isn't set
+ properly when it isn't supported on a channel (yet _another_
+ useful patch by eliel). (issue #13081) Reported by: eliel
+ Patches: app_sendtext1.4.c uploaded by eliel (license 64) Tested
+ by: eliel
+
+ * channels/chan_iax2.c: Fix a bug in 1.4 branch with iax2 channels
+ not being removed when a call was rejected (from the calling box,
+ not the box that denied the registration). Related to revisions
+ 132466 in trunk, and 132467 in 1.6.0. Earlier I had accidently
+ tested 1.4 with a backport from those revisions, so I didn't see
+ this problem (oops).
+
+2008-07-19 16:45 +0000 [r132311] Kevin P. Fleming <kpfleming@digium.com>
+
+ * LICENSE: grant a license exception to allow distribution of
+ Asterisk binaries that use the UW IMAP Toolkit (which is licensed
+ under a non-GPL-compatible license)
+
+2008-07-18 19:06 +0000 [r131970-132112] Tilghman Lesher <tlesher@digium.com>
+
+ * main/say.c: Fix for Taiwanese number syntax (closes issue #12319)
+ Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch
+ uploaded by CharlesWang (license 444)
+
+ * main/config.c: Textual clarification (closes issue #13106)
+ Reported by: flefoll Patches:
+ config.c.br14.120173.patch-unknown-directive uploaded by flefoll
+ (license 244)
+
+ * include/asterisk/sched.h, channels/chan_iax2.c: Spinlock within
+ the destroy, to allow a scheduled job to continue, if it's
+ waiting on the mutex which the destroy thread has.
+
+ * main/sched.c: Oops
+
+ * main/sched.c, include/asterisk/sched.h: Preserve ABI
+ compatibility with last change
+
+ * main/sched.c, include/asterisk/sched.h, channels/chan_iax2.c:
+ Make the ast_assert call within ast_sched_del report something
+ useful.
+
+2008-07-18 16:15 +0000 [r131921] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/dlfcn.c (removed), main/loader.c, main/Makefile,
+ include/asterisk/dlfcn-compat.h (removed): remove the dlfcn
+ compatibility stuff, because no platforms that Asterisk currently
+ runs on it use it, and it doesn't build anyway
+
+2008-07-18 15:34 +0000 [r131915] Brett Bryant <bbryant@digium.com>
+
+ * res/res_features.c: Fix a bug in blind transfers where the
+ BLINDTRANSFER variable isn't always set to the other end of the
+ blind transfer. (closes issue #12586)
+
+2008-07-17 20:35 +0000 [r131790] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Revert part of issue #5620 (revision 6965)
+ as it appears that it was in error. This should fix talk call
+ progress on analog lines. (closes issue #12178) Reported by:
+ michael-fig Patches: 20080717__bug12178.diff.txt uploaded by
+ Corydon76 (license 14)
+
+2008-07-16 22:17 +0000 [r131491] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_iax2.c: Fix a bug in iax2 registration that allowed
+ peers to register with case-insensitive names (user_cmp_cb and
+ peer_cmp_cb are now both case-sensitive). (closes issue #13091)
+
+2008-07-16 21:46 +0000 [r131480] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Apparently, in certain cases, a callno is
+ already destroyed when iax2_destroy is called.
+
+2008-07-16 20:47 +0000 [r131421] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Always ensure that the channel's tech_pvt
+ reference is NULL after calling the destroy callback. (closes
+ issue #13060) Reported by: jpgrayson Patches:
+ chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license
+ 492)
+
+2008-07-16 20:23 +0000 [r131299-131369] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Move the init_queue call back to where it used
+ to be (changed Sept 12 last year). It was moved then to prevent a
+ memory leak. Since then, the same memory leak recurred and was
+ fixed in a better way. Now it has been found that the placement
+ of this init_queue call can cause problems if a realtime queue
+ has values changed to an empty string. The problem is that the
+ default value for that queue parameter would not be set. (closes
+ issue #13084) Reported by: elbriga
+
+ * apps/app_queue.c: Apparently, "thread safety" is important,
+ whatever that means. :P (Thanks Russell!)
+
+ * apps/app_queue.c: Make absolutely certain that the transfer
+ datastore is removed from the calling channel once the caller is
+ finished in the queue. This could have weird con- sequences when
+ dialing local queue members when multiple transfers occur on a
+ single call. Also fixed a memory leak that would occur when an
+ attended transfer occurred from a queue member. (closes issue
+ #13047) Reported by: festr
+
+2008-07-16 17:53 +0000 [r131242] Steve Murphy <murf@digium.com>
+
+ * pbx/pbx_ael.c: (closes issue #13090) Reported by: murf The
+ problem was that, esoteric as it is, because the hangerupper
+ context immediately preceded the std-priv-extent macro, that the
+ checking code accidentally would fall from traversing hangerupper
+ into the std-priv-exten macro, where it would hit the hangerupper
+ in the 'includes', and proceed into an infinite recursion. A
+ small fix to traverse into the statements of the context instead
+ of the context solves this issue. I also added some commented out
+ printfs for debug, which were pretty handy in the face of a dorky
+ gdb. This was a problem around since the package was first
+ written; but evidently pretty rare in turning up in the field.
+
+2008-07-15 17:47 +0000 [r131012] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/cdr.c: remove 4 lines of redundant code. (closes issue
+ #13080) Reported by: gknispel_proformatique Patches:
+ trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261)
+
+2008-07-15 17:19 +0000 [r130889-130959] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c, channels/chan_sip.c: astman_send_error does not
+ need a newline appended -- the API takes care of that for us.
+ (closes issue #13068) Reported by: gknispel_proformatique
+ Patches: asterisk_1_4_astman_send.patch uploaded by gknispel
+ (license 261) asterisk_trunk_astman_send.patch uploaded by
+ gknispel (license 261)
+
+ * channels/chan_iax2.c: Override the callerid in all cases when the
+ callerid is set in the user, not just when a remote callerid is
+ set. Also, if not set in the user, allow the remote CallerID to
+ pass through. (closes issue #12875) Reported by: dimas Patches:
+ 20080714__bug12875.diff.txt uploaded by Corydon76 (license 14)
+
+2008-07-14 17:50 +0000 [r130792] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_dial.c: Add a check to the CAN_EARLY_BRIDGE macro in
+ app_dial to be sure there are no audiohooks present on the
+ channels involved. This fixed a one-way audio situation I had in
+ my test setup. I couldn't find any open issues that suggested
+ one-way audio with regards to mixmonitor (or other audiohook)
+ usage, though.
+
+2008-07-14 17:10 +0000 [r130735] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/dnsmgr.c: notify the user that dnsmgr refresh wont work when
+ dnsmgr is not enabled. Previously this command would
+ automagically appear and disappear. This was confusing. (closes
+ issue #12796) Reported by: chappell Patches:
+ dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by:
+ russell, chappell, mvanbaak
+
+2008-07-14 10:38 +0000 [r130634] Russell Bryant <russell@digium.com>
+
+ * main/audiohook.c: Bump up the debug level for a message.
+
+2008-07-13 22:48 +0000 [r130573] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/manager.c: fix memory leak when originate from manager
+ cannot create a thread (closes issue #13069) Reported by:
+ gknispel_proformatique Patches:
+ asterisk_trunk_action_originate.patch uploaded by gknispel
+ (license 261) Tested by: gknispel_proformatique, mvanbaak
+
+2008-07-13 17:56 +0000 [r130514] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Reverting 2 changesets, as it breaks
+ incoming IAX2 calls (Related to issue #12963) Reported by:
+ mvanbaak
+
+2008-07-12 10:25 +0000 [r130373] Michiel van Baak <michiel@vanbaak.info>
+
+ * pbx/pbx_ael.c: in 1.4 the functions still have | as argument
+ seperator. This commit fixes the use of RAND in the ael random
+ function. (closes issue #13061) Reported by: danpwi
+
+2008-07-11 22:23 +0000 [r130298-130317] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile: forcibly remove the modules that are changing names
+
+ * include/asterisk/options.h, main/asterisk.c, cdr/cdr_csv.c,
+ Makefile, main/channel.c, apps/app_dahdibarge.c,
+ channels/chan_dahdi.c, doc/hardware.txt, apps/app_flash.c,
+ apps/app_dahdiras.c, main/file.c,
+ contrib/utils/zones2indications.c, include/asterisk/channel.h,
+ channels/chan_iax2.c: a whole pile of Zaptel/DAHDI compatibility
+ work, with lots more to come... this tree is not yet ready for
+ users to be easily upgrading or switching, but it needs to be :-)
+
+2008-07-11 20:03 +0000 [r130173-130236] Mark Michelson <mmichelson@digium.com>
+
+ * main/audiohook.c: Remove redundant logic
+
+ * main/audiohook.c: Fix a typo in audiohook_read_frame_both. While
+ this change has not been proven to fix any specific issue, it is
+ incorrect and could cause unforeseen problems.
+
+2008-07-11 18:51 +0000 [r130102-130169] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Ensure that a destination callno of 0 will
+ not match for frames that do not start a dialog (new, lagrq, and
+ ping). (closes issue #12963) Reported by: russellb Patches:
+ chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492)
+
+ * channels/chan_agent.c: Pass the devicestate from an underlying
+ channel up through the Agent channel. This should make the Agent
+ always report the correct device state, even when the underlying
+ channel is used for other purposes. (closes issue #12773)
+ Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded
+ by Corydon76 (license 14) Tested by: davidw
+
+2008-07-11 16:08 +0000 [r130039-130042] Kevin P. Fleming <kpfleming@digium.com>
+
+ * doc/configuration.txt, configs/extensions.conf.sample,
+ configs/sla.conf.sample, configs/zapata.conf.sample (removed),
+ contrib/scripts/autosupport, README,
+ configs/chan_dahdi.conf.sample (added), channels/chan_dahdi.c,
+ include/asterisk/doxyref.h, doc/sla.tex, doc/ael.txt,
+ configs/extensions.ael.sample, configs/smdi.conf.sample: new
+ installations should be using DAHDI instead of Zaptel, so the
+ sample config file is now chan_dahdi.conf instead of zapata.conf
+ also, convert remaining references to zapata.conf in various
+ places
+
+ * configs/zapata.conf.sample, channels/chan_dahdi.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: add support for a
+ configuration parameter for 'inband audio during RELEASE', which
+ is currently mandatory in libpri-1.4.4 but will become
+ configurable in libpri-1.4.5 later today (related to issue
+ #13042)
+
+2008-07-11 14:18 +0000 [r129970] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/astobj.h: add a simple ASTOBJ_TRYWRLOCK macro
+ ...
+
+2008-07-11 14:14 +0000 [r129907-129967] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/astmm.c: simplify calculation
+
+ * main/astmm.c: fix a flaw found while experimenting with structure
+ alignment and padding; low-fence checking would not work properly
+ on 64-bit platforms, because the compiler was putting 4 bytes of
+ padding between the fence field and the allocation memory block
+ added a very obvious runtime warning if this condition reoccurs,
+ so the developer who broke it can be chastised into fixing it :-)
+
+ * sounds/Makefile: don't attempt to set user/group ownership of
+ extracted sound files (reported on asterisk-users) (closes issue
+ #13059)
+
+2008-07-10 21:57 +0000 [r129741-129803] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Correctly deal with duplicate NEW frames
+ (due to retransmission). Also, fixup the destination call number
+ matching to be more strict and reliable. (closes issue #12963)
+ Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch
+ uploaded by jpgrayson (license 492) Tested by: jpgrayson,
+ Corydon76
+
+ * res/res_config_odbc.c: Oops
+
+2008-07-10 16:03 +0000 [r129567] Russell Bryant <russell@digium.com>
+
+ * sample.call: Note that pbx_spool.so is the module used for call
+ files (inspired by a question in #asterisk)
+
+2008-07-10 13:57 +0000 [r129505] Sean Bright <sean.bright@gmail.com>
+
+ * main/editline: Update svn:ignore
+
+2008-07-09 19:32 +0000 [r129436] Mark Michelson <mmichelson@digium.com>
+
+ * main/rtp.c: Fix a problem where inbound rfc2833 audio would be
+ sent to the core instead of being P2P bridged. When the core
+ regenerated the rfc2833 packet for the outbound leg, the SSRC
+ would be different than the RTP audio on the call leg causing
+ DTMF detection issues on the far end. (closes issue #12955)
+ Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by
+ tsearle (license 373) Tested by: tonyredstone
+
+2008-07-09 13:41 +0000 [r129343] Sean Bright <sean.bright@gmail.com>
+
+ * main/editline/makelist (removed), main/editline/makelist.in
+ (added), main/editline/configure, main/editline/Makefile.in,
+ main/editline/configure.in: Look for the system installed awk
+ instead of assuming it's at /usr/bin/awk. Pointed out by jmls via
+ #asterisk-dev.
+
+2008-07-08 21:31 +0000 [r129158-129208] Mark Michelson <mmichelson@digium.com>
+
+ * doc/imapstorage.txt: Update documentation to have the correct
+ option name
+
+ * apps/app_voicemail.c, doc/imapstorage.txt: Backport TCP-related
+ timeouts to IMAP voicemail in 1.4 since it should solve bugs
+ people are experiencing. Specifically, there are times where
+ communication with the IMAP server causes system calls to block
+ forever. If this should happen when querying the mailbox so that
+ chan_sip's do_monitor thread can send MWI to a phone, it means
+ that SIP calls cannot be processed any more. The timeout options
+ are outlined in doc/imapstorage.txt. Defaults for the timeouts
+ are sixty seconds. (closes issue #12987) Reported by: mthomasslo
+
+2008-07-08 20:27 +0000 [r129047-129149] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_dial.c, channels/chan_sip.c, include/asterisk/causes.h:
+ Cause SIP to return a 480 instead of a 404 when a sip peer
+ exists, but is not registered. (closes issue #12885) Reported by:
+ ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76
+ (license 14) Tested by: ibc
+
+ * channels/chan_iax2.c: Timestamp decoding for video mini-frames is
+ bogus, because the timestamp only includes 15 bits, unlike voice
+ frames, which contain a 16-bit timestamp. (closes issue #13013)
+ Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch
+ uploaded by jpgrayson (license 492)
+
+2008-07-08 09:52 +0000 [r128912-128950] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't hangup the call if we can't resolve
+ the Contact if there's a proxy route set for the call. ---- This
+ comment was added a while ago and today it hit me badly. /* OEJ:
+ Possible issue that may need a check: If we have a proxy route
+ between us and the device, should we care about resolving the
+ contact or should we just send it? */
+
+ * channels/chan_sip.c: Fix issues where repeated messages where
+ ignored, but retransmitted reliably instead of unreliably.
+ Reported by: johan Patches: 12746.txt uploaded by oej (license
+ 306) Tested by: johan (issue #12746)
+
+2008-07-08 00:01 +0000 [r128812-128856] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Check for non-NULL before stripping
+ characters. (closes issue #12954) Reported by: bfsworks Patches:
+ 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: deti
+
+ * apps/app_voicemail.c: Stop using deprecated method, as requested
+ by Kevin.
+
+2008-07-07 22:41 +0000 [r128795] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix handling of when a pvt disappears.
+ Properly return the pvt locked and don't hold the pvt lock while
+ destroying the ast_channel. (closes issue #13014) Reported by:
+ jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by
+ jpgrayson (license 492)
+
+2008-07-07 20:47 +0000 [r128737] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_iax2.c: Remove spurious trailing whitespace from
+ log messages and fix a spelling error in a log message. (closes
+ issue #13017) Reported by: jpgrayson Patches:
+ chan_iax2_space_after_newline.patch uploaded by jpgrayson
+ (license 492) chan_iax2_spelling.patch uploaded by jpgrayson
+ (license 492)
+
+2008-07-07 17:02 +0000 [r128639] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_iax2.c: By using the iaxdynamicthreadcount to
+ identify a thread, it was possible for thread identifiers to be
+ duplicated. By using a globally-unique monotonically- increasing
+ integer, this is now avoided. (closes issue #13009) Reported by:
+ jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by
+ jpgrayson (license 492)
+
+2008-07-07 16:51 +0000 [r128637] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configure, configure.ac: use tzafrir's patch to fix this problem
+ properly... i made the previous set of changes without thoroughly
+ testing them, doh! (closes issue #12911) Reported by: tzafrir
+ Patches: custum_dahdi_configure_2.diff uploaded by tzafrir
+ (license 46) Tested by: tzafrir
+
+2008-07-04 16:11 +0000 [r127973-128029] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_config.c: Move the free down one
+
+ * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c: Fix the
+ 'dialplan remove extension' logic, so that it a) works with
+ cidmatch, and b) completes contexts correctly when the extension
+ is ambiguous. (closes issue #12980) Reported by: licedey Patches:
+ 20080703__bug12980.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76
+
+2008-07-03 22:20 +0000 [r127754-127895] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/Makefile: remove this, it has been moved to the main
+ Makefile
+
+ * Makefile, main/editline/np/vis.c: a couple of small
+ Solaris-related fixes (closes issue #11885) Reported by: snuffy,
+ asgaroth
+
+ * configure, main/Makefile, configure.ac, acinclude.m4: ensure that
+ DAHDI_INCLUDE and ZAPTEL_INCLUDE are added in all the places
+ needed improve AST_EXT_LIB_CHECK to accept (and remember)
+ additional CFLAGS data like it does in trunk already (closes
+ issue #12911) Reported by: tzafrir
+
+2008-07-03 00:16 +0000 [r127663] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c,
+ channels/chan_sip.c, res/res_features.c, include/asterisk/cdr.h:
+ The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported
+ by: murf Tested by: murf, deeperror (closes issue #12907)
+ Reported by: falves11 Tested by: murf, falves11 (closes issue
+ #11849) Reported by: greyvoip As to 11849, I think these changes
+ fix the core problems brought up in that bug, but perhaps not the
+ more global problems created by the limitations of CDR's
+ themselves not being oriented around transfers. Reopen if necc,
+ but bug reports are not the best medium for enhancement
+ discussions. We need to start a second-generation CDR
+ standardization effort to cover transfers. (closes issue #11093)
+ Reported by: rossbeer Tested by: greyvoip, murf
+
+2008-07-02 20:47 +0000 [r127560] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Fix thread-safety of some of the
+ pbx_builtin_getvar_helper calls
+
+2008-07-02 19:47 +0000 [r127501] Tilghman Lesher <tlesher@digium.com>
+
+ * main/acl.c: Merged revisions 127466 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 |
+ tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines
+ Solaris fix (closes issue #12949) Reported by: snuffy Patches:
+ bug_12949.diff uploaded by snuffy (license 35) ........
+
+2008-07-01 23:36 +0000 [r127244] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Add error message to failed open(2) calls
+ inside the copy() function of app_voicemail. This idea came as
+ part of my work in helping to resolve issue #12764.
+
+2008-07-01 20:25 +0000 [r126999-127133] Tilghman Lesher <tlesher@digium.com>
+
+ * build_tools/cflags.xml, channels/chan_iax2.c: Disable the old,
+ slow search for matching callno in chan_iax2 (but allow it to be
+ reenabled for debugging)
+
+ * channels/chan_iax2.c: Oops
+
+ * channels/chan_iax2.c: Change around how we schedule pings and
+ lagrqs, and fix a reason why the jobs were not getting properly
+ cancelled. (closes issue #12903) Reported by: stevedavies
+ Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76
+ (license 14) Tested by: stevedavies
+
+ * channels/chan_iax2.c: Suppress annoying warning by finding the
+ remaining cases where the callno is not in the hash.
+
+2008-07-01 14:59 +0000 [r126735-126902] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Use domain part of SIP uri in register=
+ configuration as fromdomain. Reported by: one47 Patches:
+ sip-reg-fromdom2.dpatch uploaded by one47 (license 23) (closes
+ issue #12474)
+
+ * channels/chan_sip.c: Handle escaped URI's in call pickups. Patch
+ by oej and IgorG. Reported by: IgorG Patches:
+ bug12299-11062-v2.patch uploaded by IgorG (license 20) Tested by:
+ IgorG, oej (closes issue #12299)
+
+ * configs/sip.conf.sample: Clear up documentation on "domain="
+ setting in sip.conf Reported by: davidw (closes issue #12413)
+
+ * channels/chan_sip.c: Report 200 OK to all in-dialog OPTIONs
+ requests (to confirm that the dialog exist). Don't bother
+ checking the request URI. (closes issue #11264) Reported by: ibc
+
+ * channels/chan_sip.c: Fix bad XML for hold notification. Reported
+ by: gowen72 Patches: hold.patch uploaded by gowen72 (license 432)
+ (closes issue #12942)
+
+2008-06-30 23:11 +0000 [r126680] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_dahdi.c: Load the proper channel configuration file
+ based on which driver was detected.
+
+2008-06-30 22:30 +0000 [r126674] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/zapata.conf.sample: Add note about other names for
+ EuroISDN
+
+2008-06-30 16:05 +0000 [r126573] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/lock.h: Fix a typo in the non-DEBUG_THREADS
+ version of the recently added DEADLOCK_AVOIDANCE() macro. This
+ caused the lock to not actually be released, and as a result, not
+ avoid deadlocks at all. This resolves the issues reported in the
+ last while about Asterisk locking up all over the place (and most
+ commonly, in chan_iax2). (closes issue #12927) (closes issue
+ #12940) (closes issue #12925) (potentially closes others ...)
+
+2008-06-30 12:50 +0000 [r126516] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Send all responses to an INVITE reliably, so
+ that we retransmit if we don't get an ACK and also fail if we
+ don't get the very same precious ACK. Based on patch by tsearle,
+ with my own additions. (closes issue #12951) Reported by: tsearle
+ Patches: busy_retransmit.patch uploaded by tsearle (license 373)
+
+2008-06-29 18:05 +0000 [r126395] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx/Makefile: ignore warnings for prototypes in GTK headers
+
+2008-06-27 22:01 +0000 [r125740-126056] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: When we get a 408 Timeout, don't stop trying
+ to re-register. (closes issue #12863) Reported by: ricvil
+
+ * include/asterisk/tonezone_compat.h: Since HAVE_DAHDI is defined
+ to HAVE_ZAPTEL in dahdi_compat.h, we must first check for
+ HAVE_ZAPTEL. (closes issue #12938) Reported by: opticron Patches:
+ tonezone_compat.diff uploaded by opticron (license 267)
+
+ * main/utils.c, include/asterisk/lock.h: In this debugging
+ function, copy to a buffer instead of using potentially unsafe
+ pointers.
+
+ * channels/chan_local.c: Add proper deadlock avoidance. (closes
+ issue #12914) Reported by: ozan Patches:
+ 20080625__bug12914.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: ozan
+
+2008-06-26 23:03 +0000 [r125587] Jason Parker <jparker@digium.com>
+
+ * main/utils.c: Make sure to unlock the lock_info lock (huh?).
+ Possible deadlock?
+
+2008-06-26 22:52 +0000 [r125476-125585] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Add the interface of a queue member to the
+ output of the "queue show" command so that it can easily be
+ associated with a queue member's name. This helps so that the
+ appropriate queue member can be removed or paused since the
+ interface is required, not the member's name. (closes issue
+ #12783) Reported by: davevg Patches: app_queue.diff uploaded by
+ davevg (license 209) with small mod from me
+
+ * apps/app_queue.c: Backport of attended transfer queue_log patch
+ from trunk. This patch allows for attended transfers to be logged
+ in the queue_log the same way that blind transfers have always
+ been. It was decided by popular opinion on the asterisk-dev
+ mailing list that this should be backported to 1.4. Thanks to
+ everyone who gave an opinion.
+
+ * apps/app_queue.c: Prior to this patch, the "queue show" command
+ used cached information for realtime queues instead of giving
+ up-to-date info. Now realtime is queried for the latest and
+ greatest in queue info. (closes issue #12858) Reported by: bcnit
+ Patches: queue_show.patch uploaded by putnopvut (license 60)
+
+2008-06-26 16:32 +0000 [r125384] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Add support for peer realm based auth (a few
+ missing lines, the rest is well documented but never worked)
+
+2008-06-26 15:30 +0000 [r125327] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_dahdi.c: ensure that (whenever possible) if we
+ generate a log message because an ioctl() call to DAHDI/Zaptel
+ failed, that we include the reason it failed by including the
+ stringified error number (issue AST-80)
+
+2008-06-26 11:01 +0000 [r125218-125276] Tilghman Lesher <tlesher@digium.com>
+
+ * main/rtp.c: Check for rtcp structure before trying to delete
+ schedule. (closes issue #12872) Reported by: destiny6628 Patches:
+ 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: destiny6628
+
+ * configs/agents.conf.sample: Document ackcall=always. (closes
+ issue #12852) Reported by: davidw
+
+2008-06-25 22:21 +0000 [r125132] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_rpt.c, include/asterisk/dahdi_compat.h,
+ channels/chan_dahdi.c, configure,
+ include/asterisk/tonezone_compat.h (added), configure.ac: allow
+ tonezone to live in a different place than DAHDI/Zaptel, since
+ dahdi-tools and dahdi-linux are now separate packages and can be
+ installed in different places don't include tonezone.h in
+ dahdi_compat.h, because only a couple of modules need it get
+ app_rpt building again after the DAHDI changes (closes issue
+ #12911) Reported by: tzafrir
+
+2008-06-25 00:46 +0000 [r124908-124965] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_dahdi.c: Pvt deadlock causes some channels to get
+ stuck in Reserved status. (closes issue #12621) Reported by:
+ fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by
+ Corydon76 (license 14) Tested by: fabianoheringer
+
+ * apps/app_voicemail.c: Occasionally control characters find their
+ way into CallerID. These need to be stripped prior to placing
+ CallerID in the headers of an email. (closes issue #12759)
+ Reported by: RobH Patches: 20080602__bug12759__2.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: RobH
+
+ * channels/chan_sip.c: Don't access the pvt structure if unable to
+ acquire the lock. (closes issue #12162) Reported by: norman
+ Patches: 12162-lockfail.diff uploaded by qwell (license 4)
+
+2008-06-23 21:22 +0000 [r124743] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_iax2.c: emit a warning if the old IAX2 call
+ searching code finds a call when the new code did not... so that
+ we can get rid of the old code in 2-3 months
+
+2008-06-22 02:54 +0000 [r124540] Steve Murphy <murf@digium.com>
+
+ * apps/app_forkcdr.c: (closes issue #12910) Reported by: chris-mac
+ Sorry, my testing did not contain the simple case of forkCDR(v),
+ I am much embarrassed to admit. If I had, I would have more
+ solidly initialized the opts element for varset.
+
+2008-06-20 23:12 +0000 [r124395-124450] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_rpt.c: usleep with a value over 1,000,000 is
+ nonportable. Changing to use sleep() instead. (closes issue
+ #12814) Reported by: pputman Patches: app_rtp_sleep.patch
+ uploaded by pputman (license 81)
+
+ * main/app.c: If the last character in a string to be parsed is the
+ delimiter, then we should count that final empty string as an
+ additional argument.
+
+2008-06-20 21:14 +0000 [r124372] Jeff Gehlbach <jeffg@opennms.org>
+
+ * doc/asterisk-mib.txt, doc/digium-mib.txt: Fix issues in
+ digium-mib.txt and asterisk-mib.txt to placate smilint - bug
+ 12905
+
+2008-06-20 20:16 +0000 [r124182-124315] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: When using a Local channel, started by a
+ call file, with a destination of an AGI script, the AGI script
+ does not always get notified of a hangup if the underlying
+ channel hangs up early. (closes issue #11833) Reported by: IgorG
+ Patches: local_hangup-v1.diff uploaded by IgorG (license 20)
+
+ * channels/chan_dahdi.c: It's possible for a hangup to be received,
+ even just after the initial cid spill. (closes issue #12453)
+ Reported by: Alex728 Patches: 20080604__bug12453.diff.txt
+ uploaded by Corydon76 (license 14)
+
+2008-06-19 20:28 +0000 [r124112] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Fix IMAP forwarding so that messages are
+ sent to the proper mailbox. (closes issue #12897) Reported by:
+ jaroth Patches: destination_forward.patch uploaded by jaroth
+ (license 50)
+
+2008-06-19 19:55 +0000 [r124066] Brett Bryant <bbryant@digium.com>
+
+ * main/utils.c: Merge revision 124064 from trunk. Add errors that
+ report any locks held by threads when they are being closed.
+
+2008-06-19 16:58 +0000 [r123710-123930] Tilghman Lesher <tlesher@digium.com>
+
+ * main/channel.c: Change informative messages to use the _multiple
+ variant when multiple formats are possible. (Closes issue #12848)
+ Reported by klaus3000
+
+ * build_tools/strip_nonapi: Only process 40 arguments (20 files) at
+ once with xargs, because some older shells may force xargs to
+ separate on an odd boundary. (Closes issue #12883) Reported by
+ Nik Soggia
+
+ * configs/sip.conf.sample: Correct description of notifyringing
+ option. (Closes issue #12890) Reported by gminet
+
+ * main/asterisk.c: The RDTSC instruction was introduced on the
+ Pentium line of microprocessors, and is not compatible with
+ certain 586 clones, like Cyrix. Hence, asking for i386
+ compatibility was always incorrect. See
+ http://en.wikipedia.org/wiki/RDTSC (Closes issue #12886) Reported
+ by tecnoxarxa
+
+ * main/say.c, doc/lang (added), doc/lang/hebrew.ods (added): Add
+ support for saying numbers in Hebrew. (closes issue #11662)
+ Reported by: greenfieldtech Patches: say.c.patch-12042008
+ uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods
+ uploaded by greenfieldtech (with signficant changes to the
+ spreadsheet by me)
+
+ * pbx/pbx_spool.c: Set the variables top-down, so that if a script
+ sets a variable more than once, the last one will take
+ precedence. (closes issue #12673) Reported by: phber Patches:
+ 20080519__bug12673.diff.txt uploaded by Corydon76 (license 14)
+
+2008-06-17 20:26 +0000 [r123485] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Make chan_sip build under dev mode with
+ compilers >= GCC 4.2 Thanks to jpeeler for alerting me of this
+
+2008-06-17 18:56 +0000 [r123391] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_iax2.c: Fix 3 more places where not locking the
+ structure could cause the wrong lock to be unlocked. (Closes
+ issue #12795)
+
+2008-06-17 18:09 +0000 [r123274-123333] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Cisco BTS sends SIP responses with a tab
+ between the Cseq number and SIP request method in the Cseq:
+ header. Asterisk did not handle this properly, but with this
+ patch, all is well. (closes issue #12834) Reported by: tobias_e
+ Patches: 12834.patch uploaded by putnopvut (license 60) Tested
+ by: tobias_e
+
+ * apps/app_queue.c: davidw pointed out that the holdtime
+ calculation used by app_queue does not use "boxcar" filtering as
+ the comments say. The term "boxcar" means that the number of
+ samples used to calculate stays constant, with new samples
+ replacing the oldest ones. The queue holdtime calculation uses
+ all holdtime samples collected since the queue was loaded, so the
+ comment has been changed to be accurate. (closes issue #12781)
+ Reported by: davidw
+
+2008-06-17 15:48 +0000 [r123271] Russell Bryant <russell@digium.com>
+
+ * main/astobj2.c: Fix a memory leak in astobj2 that was pointed out
+ by seanbright. When a container got destroyed, the underlying
+ bucket list entry for each object that was in the container at
+ that time did not get free'd.
+
+2008-06-16 19:50 +0000 [r123110-123113] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_mgcp.c, channels/chan_dahdi.c,
+ channels/chan_skinny.c, channels/chan_h323.c,
+ channels/chan_iax2.c: Port "hasvoicemail" change from SIP to
+ other channel drivers
+
+ * channels/chan_sip.c: People expect that if "hasvoicemail" is set
+ in users.conf, even if "mailbox" isn't set, that SIP will detect
+ a mailbox. (closes issue #12855) Reported by: PLL Patches:
+ 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: PLL
+
+2008-06-16 12:31 +0000 [r122869-122919] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only compare the first 15 characters so that
+ even if the charset is specified we still accept it as SDP.
+ (closes issue #12803) Reported by: lanzaandrea Patches:
+ chan_sip.c.diff uploaded by lanzaandrea (license 496)
+
+ * channels/chan_sip.c: Don't send a BYE on a dialog that is already
+ gone during a REFER. (closes issue #12865) Reported by: flefoll
+ Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by
+ flefoll (license 244)
+
+2008-06-13 21:44 +0000 [r122713] Mark Michelson <mmichelson@digium.com>
+
+ * main/autoservice.c: Short circuit the loop in autoservice_run if
+ there are no channels to poll. If we continued, then the result
+ would be calling poll() with a NULL pollfd array. While this is
+ fine with POSIX's poll(2) system call, those who use Asterisk's
+ internal poll mechanism (Darwin systems) would have a failed
+ assertion occur when poll is called. (related to issue #10342)
+
+2008-06-13 18:57 +0000 [r122663] Jeff Peeler <jpeeler@digium.com>
+
+ * include/asterisk/dahdi_compat.h, res/res_musiconhold.c: fixed
+ dahdi compatability header from assuming either dahdi or zaptel
+ is installed (may not have either)
+
+2008-06-13 17:45 +0000 [r122617] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c: Remove extra option from previous solution
+ attempt
+
+2008-06-13 17:36 +0000 [r122613] Jeff Peeler <jpeeler@digium.com>
+
+ * configure, configure.ac: (closes issue #12846) Reported by:
+ Netview Tested by: jpeeler Use correct location to search for
+ tonezone.
+
+2008-06-13 16:29 +0000 [r122589] Terry Wilson <twilson@digium.com>
+
+ * apps/app_dial.c, res/res_features.c: This should fix the behavior
+ of the 'T' dial feature being passed incorrectly to the
+ transferee when builtin_atxfers are used. Also, doing a
+ builtin_atxfer to parking was broken and is fixed here as well.
+ (closes issue #11898) Reported by: sergee Tested by: otherwiseguy
+
+2008-06-12 19:08 +0000 [r122314] Jeff Peeler <jpeeler@digium.com>
+
+ * main/indications.c, include/asterisk/dahdi_compat.h (added),
+ main/loader.c, main/channel.c, channels/chan_dahdi.c (added),
+ configure, apps/app_zapscan.c (removed), apps/app_zapras.c
+ (removed), main/app.c, include/asterisk/options.h,
+ apps/app_rpt.c, channels/chan_mgcp.c, apps/app_read.c,
+ channels/chan_zap.c (removed), apps/app_page.c,
+ include/asterisk/indications.h, apps/app_dahdiras.c (added),
+ configure.ac, apps/app_disa.c, include/asterisk/channel.h,
+ apps/app_getcpeid.c, apps/app_queue.c, apps/app_zapbarge.c
+ (removed), channels/chan_misdn.c, apps/app_flash.c,
+ build_tools/menuselect-deps.in, funcs/func_channel.c,
+ main/file.c, res/snmp/agent.c, contrib/utils/zones2indications.c,
+ codecs/codec_dahdi.c (added), res/res_indications.c,
+ pbx/pbx_config.c, makeopts.in, apps/app_chanspy.c,
+ main/asterisk.c, apps/app_dahdibarge.c (added),
+ apps/app_meetme.c, include/asterisk/autoconfig.h.in,
+ apps/app_dahdiscan.c (added), acinclude.m4,
+ res/res_musiconhold.c, codecs/codec_zap.c (removed),
+ channels/chan_iax2.c: Adds DAHDI support alongside Zaptel. DAHDI
+ usage favored, but all Zap stuff should continue working. Release
+ announcement to follow.
+
+2008-06-12 18:50 +0000 [r122311] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Properly play a holdtime message if the
+ announce-holdtime option is set to "once." (closes issue #12842)
+ Reported by: ramonpeek Patches: patch001.diff uploaded by
+ ramonpeek (license 266)
+
+2008-06-12 18:22 +0000 [r122259] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix some race conditions that cause
+ ast_assert() to report that chan_iax2 tried to remove an entry
+ that wasn't in the scheduler
+
+2008-06-12 15:46 +0000 [r122208] Jeff Peeler <jpeeler@digium.com>
+
+ * apps/app_parkandannounce.c, res/res_features.c: (closes issue
+ #12193) Reported by: davidw Patch by: Corydon76, modified by me
+ to work properly with ParkAndAnnounce app
+
+2008-06-12 15:18 +0000 [r122130-122137] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_meetme.c: Flipflop the sections for two options, since
+ the section for 'X' (exit context) may otherwise absorb
+ keypresses meant for 's' (admin/user menu). (closes issue #12836)
+ Reported by: blitzrage Patches: 20080611__bug12836.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: blitzrage
+
+ * main/channel.c: Occasionally, the alertpipe loses its nonblocking
+ status, so detect and correct that situation before it causes a
+ deadlock. (Reported and tested by ctooley via #asterisk-dev)
+
+2008-06-12 14:51 +0000 [r122127] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, apps/app_forkcdr.c: Arkadia tried to warn me, but the
+ code added to ast_cdr_busy, _failed, and _noanswer was redundant.
+ Didn't spot it until I was resolving conflicts in trunk. Ugh.
+ Redundant code removed. It wasn't harmful. Just dumb.
+
+2008-06-12 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.21 released.
+
+2008-06-06 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.21-rc2 released.
+
+2008-06-05 18:03 +0000 [r120731-120735] Russell Bryant <russell@digium.com>
+
+ * UPGRADE-1.2.txt: fix filename
+
+ * UPGRADE-1.2.txt (added), UPGRADE.txt: Add the UPGRADE.txt file
+ from Asterisk 1.2, for handy reference.
+
+2008-06-05 16:56 +0000 [r120675] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Ignore appended resource when comparing JIDs.
+
+2008-06-05 16:38 +0000 [r120671] Russell Bryant <russell@digium.com>
+
+ * doc/smdi.txt, res/res_smdi.c: It turns out that searching on the
+ forwarding station isn't very useful for most people, so pull in
+ the changes that allow searching for SMDI messages based on other
+ components of the SMDI message. Also, update the SMDI
+ documentation.
+
+2008-06-04 22:05 +0000 [r120513] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Make sure that the string we set will survive
+ the unref of the queue member. Thanks to Russell, who pointed
+ this out.
+
+2008-06-04 18:35 +0000 [r120425] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_zap.c: If we fail to setup the PRI request channel,
+ don't continue, exit with an error. (closes issue #11989)
+ Reported by: Corydon76 Patches: 20080213__zap_memleak.diff.txt
+ uploaded by Corydon76 (license 14)
+
+2008-06-04 16:26 +0000 [r120371] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_config.c: Make the "dialplan remove include" CLI command
+ actually work. Also, tweak some formatting, and make the success
+ message a little bit more clear. (closes AST-52)
+
+2008-06-04 14:11 +0000 [r120285] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Tab completion when removing a member should
+ give the member's interface, not the name, since the interface is
+ what is expected for the command. (closes issue #12783) Reported
+ by: davevg
+
+2008-06-04 13:31 +0000 [r120282] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c, pbx/pbx_config.c: Fix a log message and add a message
+ for when the dialplan is done reloading. (closes issue #12716)
+ Reported by: chappell Patches: dialplan_reload_2.diff uploaded by
+ chappell (license 8)
+
+2008-06-03 22:41 +0000 [r120226] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_loopback.c: Due to incorrect use of the
+ AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform
+ any translation on the extension number before searching for it
+ in the target context. (closes issue #12473) Reported by:
+ chappell Patches: pbx_loopback.c.diff uploaded by chappell
+ (license 8)
+
+2008-06-03 22:15 +0000 [r120173] Jeff Peeler <jpeeler@digium.com>
+
+ * main/config.c: (closes issue #11594) Reported by: yem Tested by:
+ yem This change decreases the buffer size allocated on the stack
+ substantially in config_text_file_load when LOW_MEMORY is turned
+ on. This change combined with the fix from revision 117462
+ (making mkintf not copy the zt_chan_conf structure) was enough to
+ prevent the crash.
+
+2008-06-03 21:34 +0000 [r120168] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix another place where peer->callno could
+ change at a very bad time, and also fix a place where a peer was
+ used after the reference was released. (inspired by rev 120001)
+
+2008-06-03 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.21-rc1 released.
+
+2008-06-03 18:23 +0000 [r120001-120061] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: When listing the manager users, managers in
+ users.conf are not shown, even though they are allowed to
+ connect. (closes issue #12594) Reported by: bkruse Patches:
+ 12594-managerusers-2.diff uploaded by qwell (license 4) Tested
+ by: bkruse
+
+ * channels/chan_iax2.c: Save the callno when we're poking, because
+ our peer structure could change during deadlock avoidance (and
+ thus we unlock the wrong callno, causing a cascade failure).
+ (closes issue #12717) Reported by: gewfie Patches:
+ 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: gewfie
+
+2008-06-03 15:26 +0000 [r119929-119966] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18,
+ pbx/ael/ael-test/ref.ael-vtest13,
+ pbx/ael/ael-test/ref.ael-vtest17,
+ pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1,
+ pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5,
+ pbx/ael/ael-test/ref.ael-test15: Updated the regressions on AEL.
+ Hadn't updated this for the changes I made to preserve ${EXTEN}
+ in switches, which affected several tests because it adds extra
+ priorities, and at least one needed to be updated because of the
+ removal of the empty extension warning message.
+
+ * pbx/pbx_ael.c: as per
+ http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html,
+ which is a message from Philipp Kempgen, requesting that the
+ WARNING that an extension is empty be reduced to a NOTICE or
+ less, as empty extensions are syntactically possible, and no big
+ deal. With which I agree, and have removed that WARNING message
+ entirely. I think it is not necessary to see this message. It
+ didn't state that a NoOp() was inserted automatically on your
+ behalf, and really, as users, who cares? Why freak out dialplan
+ writers with unnecessary warnings? The details of the
+ machinations a compiler goes thru to produce working assembly
+ code is of little interest to most programmers-- we will follow
+ the unix principal of doing our work silently.
+
+2008-06-03 14:46 +0000 [r119926] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Treat ECONNREFUSED as an error that will
+ stop further retransmissions. (issue #AST-58, patch from
+ Switchvox)
+
+2008-06-02 20:08 +0000 [r119742-119838] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Revert a change made for issue #12479. This
+ change caused a regression such that a dial string such as
+ (IAX2/foo) did not automatically fall back to dialing the 's'
+ extension anymore. (closes issue #12770) Reported by: dagmoller
+
+ * main/manager.c: Improve CLI command blacklist checking for the
+ command manager action. Previously, it did not handle case or
+ whitespace properly. This made it possible for blacklisted
+ commands to get executed anyway. (closes issue #12765)
+
+2008-06-02 14:32 +0000 [r119740] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c, res/res_jabber.c: Do not link the guest
+ account with any configured XMPP client (in jabber.conf). The
+ actual connection is made when a call comes in Asterisk. Fix the
+ ast_aji_get_client function that was not able to retrieve an XMPP
+ client from its JID. (closes issue #12085) Reported by: junky
+ Tested by: phsultan
+
+2008-06-02 12:30 +0000 [r119687] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Even of the first PING or LAGRQ doesn't get
+ sent because it comes up too soon, make sure to reschedule so it
+ gets sent later.
+
+2008-06-02 09:29 +0000 [r119585-119636] Christian Richter <christian.richter@beronet.com>
+
+ * channels/misdn/isdn_lib.c: fixed compile issue when dev-mode is
+ enabled
+
+ * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h: Added
+ counter for unhandled_bmsg Print, this prevents the logs to be
+ flooded to fast and save CPU in this error scenario. Added
+ 'last_used' element to bc structure, when a bchannel changes from
+ used to free this exact time will be marked in last_used. When a
+ new channel is requested the find_free_chan function will check
+ if the new empty channel was used within the last second, if yes
+ it will search for the next channel, if no it will return this
+ channel. This simple mechanism has prooven to prevent race
+ conditions where the NT and TE tried to allocate the exact same
+ channel at the same time (RELEASE cause: 44).
+
+2008-06-02 01:06 +0000 [r119530-119533] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Change a debug message to an actual debug
+ message
+
+ * apps/app_dial.c: Fix another typo in documentation
+
+2008-06-01 20:47 +0000 [r119478] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_dial.c: small typo fix 'retires' => 'retries'
+
+2008-05-30 21:17 +0000 [r119404] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_queue.c: When joinempty=strict, it only failed on join
+ if there were busy members. If all members were logged out OR
+ paused, then it (incorrectly) let callers join the queue. (closes
+ issue #12451) Reported by: davidw
+
+2008-05-30 19:46 +0000 [r119354] Joshua Colp <jcolp@digium.com>
+
+ * main/autoservice.c: Fix a bug I found while testing for another
+ issue.
+
+2008-05-30 16:44 +0000 [r119301] Michiel van Baak <michiel@vanbaak.info>
+
+ * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
+ 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.slackware.asterisk: dont use a bashism way to
+ check the $VERSION variable. The rc/init.d scripts, and
+ safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2
+ (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski)
+ (closes issue #12687) Reported by: loloski Patches:
+ 20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by
+ mvanbaak (license 7) Tested by: loloski, mvanbaak
+
+2008-05-30 12:55 +0000 [r119076-119238] Russell Bryant <russell@digium.com>
+
+ * /, channels/chan_iax2.c: Merged revisions 119237 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30
+ May 2008) | 7 lines - Instead of only enforcing destination call
+ number checking on an ACK, check all full frames except for PING
+ and LAGRQ, which may be sent by older versions too quickly to
+ contain the destination call number. (As suggested by Tim Panton
+ on the asterisk-dev list) - Merge changes from
+ team/russell/iax2-frame-race, which prevents PING and LAGRQ from
+ being sent before the destination call number is known. ........
+
+ * main/autoservice.c: Fix a race condition in channel autoservice.
+ There was still a small window of opportunity for a DTMF frame,
+ or some other deferred frame type, to come in and get dropped.
+ (closes issue #12656) (closes issue #12656) Reported by: dimas
+ Patches: v3-12656.patch uploaded by dimas (license 88) -- with
+ some modifications by me
+
+ * include/asterisk/audiohook.h: Oddly enough, all of the contents
+ of audiohook.h were in there twice. I have removed the second
+ copy.
+
+2008-05-29 20:24 +0000 [r119071] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_zap.c: Call waiting tone occurs too often, because
+ it's getting serviced by both subchannels. (closes issue #11354)
+ Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded
+ by Corydon76 (license 14)
+
+2008-05-29 19:04 +0000 [r118956-119012] Russell Bryant <russell@digium.com>
+
+ * apps/app_milliwatt.c: - Fix a typo in the argument to Playtones -
+ use ast_safe_sleep() instead of calling the wait application
+ (thanks to tilghman for pointing these out!)
+
+ * /, channels/chan_iax2.c: Merged revisions 119008 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29
+ May 2008) | 7 lines Merge changes from
+ team/russell/iax2-another-fix-to-the-fix As described in the
+ following post to the asterisk-dev mailing list, only enforce
+ destination call numbers when processing an ACK.
+ http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html
+ (closes issue #12631) ........
+
+ * apps/app_milliwatt.c: - Mark app_milliwatt dependent on
+ res_indications (thanks to jsmith) - fix a typo in a log message
+ (thanks to qwell)
+
+ * apps/app_milliwatt.c: Change milliwatt to use the proper tone by
+ default (1004 Hz) instead of 1000 Hz. An option is there to use
+ 1000 Hz for anyone that might want it.
+
+2008-05-29 17:33 +0000 [r118953-118954] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/lock.h: Define also when not DEBUG_THREADS
+
+ * channels/chan_mgcp.c, channels/chan_zap.c, channels/chan_sip.c,
+ channels/chan_agent.c, channels/chan_alsa.c, main/utils.c,
+ include/asterisk/lock.h, channels/chan_iax2.c: Add some debugging
+ code that ensures that when we do deadlock avoidance, we don't
+ lose the information about how a lock was originally acquired.
+
+2008-05-29 00:25 +0000 [r118858] Steve Murphy <murf@digium.com>
+
+ * main/cdr.c, apps/app_forkcdr.c: (closes issue #10668) (closes
+ issue #11721) (closes issue #12726) Reported by: arkadia Tested
+ by: murf These changes: 1. revert the changes made via bug 10668;
+ I should have known that such changes, even tho they made sense
+ at the time, seemed like an omission, etc, were actually integral
+ to the CDR system via forkCDR. It makes sense to me now that
+ forkCDR didn't natively end any CDR's, but rather depended on
+ natively closing them all at hangup time via traversing and
+ closing them all, whether locked or not. I still don't completely
+ understand the benefits of setvar and answer operating on locked
+ cdrs, but I've seen enough to revert those changes also, and stop
+ messing up users who depended on that behavior. bug 12726 found
+ reverting the changes fixed his changes, and after a long review
+ and working on forkCDR, I can see why. 2. Apply the suggested
+ enhancements proposed in 10668, but in a completely compatible
+ way. ForkCDR will behave exactly as before, but now has new
+ options that will allow some actions to be taken that will
+ slightly modify the outcome and side-effects of forkCDR. Based on
+ conversations I've had with various people, these small tweaks
+ will allow some users to get the behavior they need. For
+ instance, users executing forkCDR in an AGI script will find the
+ answer time set, and DISPOSITION set, a situation not covered
+ when the routines were first written. 3. A small problem in the
+ cdr serializer would output answer and end times even when they
+ were not set. This is now fixed.
+
+2008-05-28 16:10 +0000 [r118716] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_iax2.c: merge revision 118702 from trunk to 1.4 --
+ Fixes a bug in chan_iax that uses send_command to poke a peer
+ while a channel is unlocked in some cases, and because it can
+ cause seemingly random failures could be related to some bugs in
+ the tracker...
+
+2008-05-28 14:23 +0000 [r118558-118646] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add an
+ option to use the source IP address of RTP as the destination IP
+ address of UDPTL when a specific option is enabled. If the remote
+ side is properly configured (ports forwarded) then UDPTL will
+ flow. (closes issue #10417) Reported by: cstadlmann
+
+ * channels/chan_sip.c: Fix an issue where codec preferences were
+ not set on dialogs that were not authenticated via a user or peer
+ and allow framing to work without rtpmap in the SDP. (closes
+ issue #12501) Reported by: slimey
+
+2008-05-27 19:15 +0000 [r118551] Tilghman Lesher <tlesher@digium.com>
+
+ * main/cli.c: When showing an error message for a command, don't
+ shorten the command output, as it tends to confuse the user (it's
+ fine for suggesting other commands, however). Reported by:
+ seanbright (on #asterisk-dev) Fixed by: me
+
+2008-05-27 19:07 +0000 [r118509] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Russell noted to me that in the case that
+ separate threads use their own addressing system, the fix I made
+ for issue 12376 does not guarantee uniqueness to the datastores'
+ uids. Though I know of no system that works this way, I am going
+ to change this right now to prevent trying to track down some
+ future bug that may occur and cause untold hours of debugging
+ time to track down. The change involves using a global counter
+ which increases with each new chanspy_ds which is created. This
+ guarantees uniqueness.
+
+2008-05-27 18:58 +0000 [r118465] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: NULL character should terminate only commands
+ back to the core, not log messages to the console. (closes issue
+ #12731) Reported by: seanbright Patches:
+ 20080527__bug12731.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: seanbright
+
+2008-05-27 17:17 +0000 [r118416] Michiel van Baak <michiel@vanbaak.info>
+
+ * apps/app_voicemail.c: small update to the g() option of
+ app_voicemail to note that gain changes only work on zap channels
+ right now. issue #12578 shows it's not clear right now.
+
+2008-05-27 16:38 +0000 [r118365] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Add a unique id to the datastore allocated in
+ app_chanspy since it is possible that multiple spies may be
+ listening to the same channel. (closes issue #12376) Reported by:
+ DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut
+ (license 60) Tested by: destiny6628 (closes issue #12243)
+ Reported by: atis
+
+2008-05-27 15:45 +0000 [r118358] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/queues.conf.sample: Add a note that pbx_config.so is
+ needed for Local channels. (Closes issue #12671)
+
+2008-05-25 16:02 +0000 [r118251] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Realtime flag affects construction in
+ multiple ways, so consulting whether rtcachefriends was set was
+ done too soon (needed to be done inside build_peer, not just as a
+ flag to build_peer). Also, fullcontact needed to be
+ reconstructed, because realtime separates the embedded ';' into
+ multiple fields. (closes issue #12722) Reported by: barthpbx
+ Patches: 20080525__bug12722.diff.txt uploaded by Corydon76
+ (license 14) Tested by: barthpbx (Much of the discussion happened
+ on #asterisk-dev for diagnosing this issue)
+
+2008-05-23 21:21 +0000 [r118163] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_zap.c: Fix a few things I missed to ensure
+ zt_chan_conf structure is not modified in mkintf
+
+2008-05-23 13:18 +0000 [r118052-118055] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/utils.h: Add format type checking for recently
+ de-inlined function
+
+ * doc/cli.txt (added), doc/00README.1st: Add information on using
+ the Asterisk console, including tab command line completion.
+ (Closes issue #12681)
+
+2008-05-23 12:30 +0000 [r118048] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/utils.h, main/utils.c: Don't declare a function
+ that takes variable arguments as inline, because it's not valid,
+ and on some compilers, will emit a warning.
+ http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes
+ issue #12289) Reported by: francesco_r Patches by Tilghman, final
+ patch by me
+
+2008-05-22 18:53 +0000 [r117809-117899] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Also remove preamble from asynchronous events
+ (reported by jsmith on #asterisk-dev)
+
+ * funcs/func_realtime.c: Take into account the length of delimiters
+ when calculating result string length. (closes issue #12696)
+ Reported by: adomjan Patches: func_realtime.c-longdelimiter.patch
+ uploaded by adomjan (license 487)
+
+2008-05-21 20:11 +0000 [r117582] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_zap.c: Ensure that passed in zt_chan_conf structure
+ is not modified in mkintf.
+
+2008-05-21 19:38 +0000 [r117574] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Apply the autoframing setting to dialogs
+ that do not get matched against a user or peer.
+
+2008-05-21 18:44 +0000 [r117519-117523] Tilghman Lesher <tlesher@digium.com>
+
+ * pbx/pbx_spool.c: Revert accidental commit of the last change
+
+ * main/asterisk.c, pbx/pbx_spool.c: Strip the preamble from the
+ output also when -rx is not being used (Related to issue #12702)
+
+2008-05-21 18:28 +0000 [r117479-117514] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Don't filter the magic character in the network
+ verboser. It gets filtered once it reaches the client. (related
+ to issue #12702, pointed out by tilghman)
+
+ * main/asterisk.c, pbx/pbx_gtkconsole.c: 1) Don't print the verbose
+ marker in front of every message from ast_verbose() being sent to
+ remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose
+ marker. (related to issue #12702)
+
+ * main/asterisk.c: Don't display the verbose marker for calls to
+ ast_verbose() that do not include a VERBOSE_PREFIX in front of
+ the message. (closes issue #12702) Reported by: johnlange Patched
+ by me
+
+2008-05-21 16:58 +0000 [r117462] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_zap.c: Pass a pointer for the conf parameter to the
+ function mkintf rather than the whole zt_chan_conf structure.
+
+2008-05-20 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.20 released.
+
+2008-05-14 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.20-rc3 released.
+
+2008-05-14 12:51 +0000 [r116230] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Accept text messages even with Content-Type:
+ text/plain;charset=Södermanländska
+
+2008-05-13 23:47 +0000 [r116088] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c, include/asterisk/lock.h: A change to the way
+ channel locks are handled when DEBUG_CHANNEL_LOCKS is defined.
+ After debugging a deadlock, it was noticed that when
+ DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin
+ of channel locks is obscured by the fact that all channel locks
+ appear to happen in the function ast_channel_lock(). This code
+ change redefines ast_channel_lock to be a macro which maps to
+ __ast_channel_lock(), which then relays the proper file name,
+ line number, and function name information to the core lock
+ functions so that this information will be displayed in the case
+ that there is some sort of locking error or core show locks is
+ issued.
+
+2008-05-13 21:17 +0000 [r115990-116038] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c: Fix a deadlock involving channel
+ autoservice and chan_local that was debugged and fixed by
+ mmichelson and me. We observed a system that had a bunch of
+ threads stuck in ast_autoservice_stop(). The reason these threads
+ were waiting around is because this function waits to ensure that
+ the channel list in the autoservice thread gets rebuilt before
+ the stop() function returns. However, the autoservice thread was
+ also locked, so the autoservice channel list was never getting
+ rebuilt. The autoservice thread was stuck waiting for the channel
+ lock on a local channel. However, the local channel was locked by
+ a thread that was stuck in the autoservice stop function. It
+ turned out that the issue came down to the local_queue_frame()
+ function in chan_local. This function assumed that one of the
+ channels passed in as an argument was locked when called.
+ However, that was not always the case. There were multiple cases
+ in which this channel was not locked when the function was
+ called. We fixed up chan_local to indicate to this function
+ whether this channel was locked or not. The previous assumption
+ had caused local_queue_frame() to improperly return with the
+ channel locked, where it would then never get unlocked. (closes
+ issue #12584) (related to issue #12603)
+
+ * main/autoservice.c: Fix an issue that I noticed in autoservice
+ while mmichelson and I were debugging a different problem. I
+ noticed that it was theoretically possible for two threads to
+ attempt to start the autoservice thread at the same time. This
+ change makes the process of starting the autoservice thread,
+ thread-safe.
+
+2008-05-13 20:28 +0000 [r115944] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_alsa.c: Use the right flag to open the audio in
+ non-blocking. (closes issue #12616) Reported by:
+ nicklewisdigiumuser
+
+2008-05-13 18:36 +0000 [r115884] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: If the socket dies (read returns 0=EOF), return
+ immediately. (Closes issue #12637)
+
+2008-05-12 17:51 +0000 [r115735] Mark Michelson <mmichelson@digium.com>
+
+ * main/utils.c: If a thread holds no locks, do not print any
+ information on the thread when issuing a core show locks command.
+ This will help to de-clutter output somewhat. Russell said it
+ would be fine to place this improvement in the 1.4 branch, so
+ that's why it's going here too.
+
+2008-05-09 16:34 +0000 [r115579] Joshua Colp <jcolp@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac:
+ Improve res_ninit and res_ndestroy autoconf logic on the Darwin
+ platform.
+
+2008-05-08 19:19 +0000 [r115545-115568] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Remove debug output.
+
+ * /, channels/chan_iax2.c: Merged revisions 115564 via svnmerge
+ from https://origsvn.digium.com/svn/asterisk/branches/1.2
+ ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08
+ May 2008) | 25 lines Fix a race condition that bbryant just found
+ while doing some IAX2 testing. He was running Asterisk trunk
+ running IAX2 calls through a few Asterisk boxes, however, the
+ audio was extremely choppy. We looked at a packet trace and saw a
+ storm of INVAL and VNAK frames being sent from one box to
+ another. It turned out that what had happened was that one box
+ tried to send a CONTROL frame before the 3 way handshake had
+ completed. So, that frame did not include the destination call
+ number, because it didn't have it yet. Part of our recent work
+ for security issues included an additional check to ensure that
+ frames that are supposed to include the destination call number
+ have the correct one. This caused the frame to be rejected with
+ an INVAL. The frame would get retransmitted for forever, rejected
+ every time ... This race condition exists in all versions that
+ got the security changes, in theory. However, it is really only
+ likely that this would cause a problem in Asterisk trunk. There
+ was a control frame being sent (SRCUPDATE) at the _very_
+ beginning of the call, which does not exist in 1.2 or 1.4.
+ However, I am fixing all versions that could potentially be
+ affected by the introduced race condition. These changes are what
+ bbryant and I came up with to fix the issue. Instead of simply
+ dropping control frames that get sent before the handshake is
+ complete, the code attempts to wait a little while, since in most
+ cases, the handshake will complete very quickly. If it doesn't
+ complete after yielding for a little while, then the frame gets
+ dropped. ........
+
+ * channels/chan_sip.c: Don't give up on attempting an outbound
+ registration if we receive a 408 Timeout. (closes issue #12323)
+
+ * contrib/scripts/postgres_cdr.sql (removed): remove
+ postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as
+ well (closes issue #9676)
+
+ * contrib/init.d/rc.debian.asterisk: Don't exit the script if
+ Asterisk is not running. (closes issue #12611)
+
+ * main/pbx.c: Don't use a channel before checking for channel
+ allocation failure. (closes issue #12609) Reported by: edantie
+
+ * contrib/init.d/rc.debian.asterisk: Use the same method for
+ executing Asterisk as the rest of the script. (closes issue
+ #12611) Reported by: b_plessis
+
+2008-05-07 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.20-rc2 released.
+
+2008-05-07 18:17 +0000 [r115512-115517] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Track peer references when stored in the
+ sip_pvt struct as the peer related to a qualify ping or a
+ subscription. This fixes some realtime related crashes. (closes
+ issue #12588) (closes issue #12555)
+
+2008-05-06 19:55 +0000 [r115418-115422] Jason Parker <jparker@digium.com>
+
+ * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115421
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) |
+ 7 lines read requires an argument on some non-bash shells (closes
+ issue #12593) Reported by: bkruse Patches:
+ getilbc.sh_12593_v1.diff uploaded by bkruse (license 132)
+ ........
+
+ * res/res_musiconhold.c: Switch to using ast_random() rather than
+ just rand(). This does not fix the bug reported, but I believe it
+ is correct. (from issue #12446) Patches: bug_12446.diff uploaded
+ by snuffy (license 35)
+
+2008-05-06 19:31 +0000 [r115415] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c: Don't print the terminating NUL. (Closes issue
+ #12589)
+
+2008-05-06 13:54 +0000 [r115341] Joshua Colp <jcolp@digium.com>
+
+ * configure, configure.ac: Add in missing argument.
+
+2008-05-05 22:50 +0000 [r115333] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, main/logger.c: Separate verbose output from CLI
+ output, by using a preamble. (closes issue #12402) Reported by:
+ Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt
+ uploaded by Corydon76 (license 14)
+ 20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by
+ Corydon76 (license 14)
+
+2008-05-05 22:10 +0000 [r115327] Joshua Colp <jcolp@digium.com>
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
+ configure.ac: Make sure that either the main speex library
+ contains preprocess functions or that speexdsp does. If both fail
+ then speex stuff can not be built.
+
+2008-05-05 21:41 +0000 [r115320] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Don't consider a caller "handled" until the
+ caller is bridged with a queue member. There was too much of an
+ opportunity for the member to hang up (either during a delay,
+ announcement, or overly long agi) between the time that he
+ answered the phone and the time when he actually was bridged with
+ the caller. The consequence of this was that if the member hung
+ up in that interval, then proper abandonment details would not be
+ noted in the queue log if the caller were to hang up at any point
+ after the member hangup. (closes issue #12561) Reported by:
+ ablackthorn
+
+2008-05-05 20:17 +0000 [r115308-115312] Tilghman Lesher <tlesher@digium.com>
+
+ * Makefile: Reverse order, such that user configs override default
+ selections
+
+ * include/asterisk/res_odbc.h: Err, the documentation on the return
+ value of ast_odbc_backslash_is_escape is exactly backwards.
+
+2008-05-05 19:49 +0000 [r115297-115304] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Avoid putting opaque="" in Digest
+ authentication. This patch came from switchvox. It fixes
+ authentication with Primus in Canada, and has been in use for a
+ very long time without causing problems with any other providers.
+ (closes issue AST-36)
+
+2008-05-05 03:22 +0000 [r115285] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk,
+ 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.slackware.asterisk: When starting Asterisk, bug
+ out if Asterisk is already running. (closes issue #12525)
+ Reported by: explidous Patches: 20080428__bug12525.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: mvanbaak
+
+2008-05-04 02:09 +0000 [r115276-115282] Joshua Colp <jcolp@digium.com>
+
+ * configure, acinclude.m4: Expand the test function for GCC
+ attributes so that more complex attributes are properly
+ recognized.
+
+ * include/asterisk/compiler.h: For my next trick I will make these
+ work with what our autoconf header file gives us.
+
+ * configure, acinclude.m4: Treat warnings as errors when checking
+ if a GCC attribute exists. We have to do this as GCC will just
+ ignore the attribute and pop up a warning, it won't actually fail
+ to compile.
+
+2008-05-02 20:25 +0000 [r115257] Brett Bryant <bbryant@digium.com>
+
+ * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in,
+ configure.ac, CHANGES: Add new "pri show version" command to show
+ the libpri version for support reasons.
+
+2008-05-02 14:28 +0000 [r115196] Mark Michelson <mmichelson@digium.com>
+
+ * include/asterisk/sched.h: Clarify a comment that was, well, just
+ wrong. It turns out that ignoring the way that macros expand.
+ Instead, I have clarified in the comment why the macro will work
+ even if the scheduler id for the task to be deleted changes
+ during the execution of the macro.
+
+2008-05-01 23:20 +0000 [r115017-115102] Tilghman Lesher <tlesher@digium.com>
+
+ * include/asterisk/res_odbc.h: Change the comment of deprecated to
+ an actual compiler deprecation
+
+ * main/utils.c: '#' is another reserved character for URIs that
+ also needs to be escaped. (closes issue #10543) Reported by:
+ blitzrage Patches: 20080418__bug10543.diff.txt uploaded by
+ Corydon76 (license 14)
+
+2008-05-01 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.20-rc1 released.
+
+2008-04-30 16:30 +0000 [r114891] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c:
+ Merge changes from team/russell/iax2_find_callno and
+ iax2_find_callno_1.4 These changes address a critical performance
+ issue introduced in the latest release. The fix for the latest
+ security issue included a change that made Asterisk randomly
+ choose call numbers to make them more difficult to guess by
+ attackers. However, due to some inefficient (this is by far, an
+ understatement) code, when Asterisk chose high call numbers,
+ chan_iax2 became unusable after just a small number of calls. On
+ a small embedded platform, it would not be able to handle a
+ single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run
+ more than about 16 IAX2 channels. Ouch. These changes address
+ some performance issues of the find_callno() function that have
+ bothered me for a very long time. On every incoming media frame,
+ it iterated through every possible call number trying to find a
+ matching active call. This involved a mutex lock and unlock for
+ each call number checked. So, if the random call number chosen
+ was 20000, then every media frame would cause 20000 locks and
+ unlocks. Previously, this problem was not as obvious since
+ Asterisk always chose the lowest call number it could. A second
+ container for IAX2 pvt structs has been added. It is an astobj2
+ hash table. When we know the remote side's call number, the pvt
+ goes into the hash table with a hash value of the remote side's
+ call number. Then, lookups for incoming media frames are a very
+ fast hash lookup instead of an absolutely insane array traversal.
+ In a quick test, I was able to get more than 3600% more IAX2
+ channels on my machine with these changes.
+
+2008-04-30 16:23 +0000 [r114890] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Don't crash on bad SIP replys. Fix created
+ in Huntsville together with Mark M (putnopvut) (closes issue
+ #12363) Reported by: jvandal Tested by: putnopvut, oej
+
+2008-04-30 14:46 +0000 [r114875-114880] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/iax2.h, channels/chan_iax2.c: use the ARRAY_LEN macro
+ for indexing through the iaxs/iaxsl arrays so that the size of
+ the arrays can be adjusted in one place, and change the size of
+ the arrays from 32768 calls to 2048 calls when LOW_MEMORY is
+ defined
+
+ * Makefile.rules: pay attention to *all* header files for
+ dependency tracking, not just the local ones (inspired by r578 of
+ asterisk-addons by tilghman)
+
+2008-04-29 19:40 +0000 [r114848] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Use the MACRO_CONTEXT and MACRO_EXTEN channel
+ variables instead of the channel's macrocontext and macroexten
+ fields. This is needed because if macros are daisy-chained, the
+ incorrect context and extension are placed on the new channel. I
+ also added locking to the channel prior to accessing these
+ variables as noted in trunk's janitor project file. (closes issue
+ #12549) Reported by: darren1713 Patches:
+ app_queue.c.macroextenpatch uploaded by darren1713 (license 116)
+ (with modifications from me) Tested by: putnopvut
+
+2008-04-29 17:08 +0000 [r114829] Jason Parker <jparker@digium.com>
+
+ * res/res_config_pgsql.c: Change warning message to debug, since
+ there are cases where 0 results is perfectly fine.
+
+2008-04-29 12:53 +0000 [r114823] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114822
+ via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr
+ 2008) | 2 lines stop script from appending source code if run
+ multiple times ........
+
+2008-04-28 04:47 +0000 [r114708] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c, channels/chan_gtalk.c: When modules are
+ embedded, they take on a different name, without the ".so"
+ extension. Specifically check for this name, when we're checking
+ if a module is loaded. (Closes issue #12534)
+
+2008-04-27 01:26 +0000 [r114695] Sean Bright <sean.bright@gmail.com>
+
+ * configure, configure.ac: When we don't explicitly pass a path to
+ the --with-tds configure option, we may end up finding tds.h in
+ /usr/local/include instead of /usr/include. If this happens, the
+ grep that looks for the version (from tdsver.h) will fail and
+ we'll have some problems during the build.
+
+2008-04-26 13:15 +0000 [r114689] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/vmail.cgi: Clicking forward without selecting a
+ message leaves an errant .lock file. (closes issue #12528)
+ Reported by: pukepail Patches: patch.diff uploaded by pukepail
+ (license 431)
+
+2008-04-25 21:54 +0000 [r114673] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Use consistent logic for checking to see if
+ a call number has been chosen yet. Also, remove some redundant
+ logic I recently added in a fix.
+
+2008-04-25 19:32 +0000 [r114662] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Move the unlock of the spyee channel to
+ outside the start_spying() function so that the channel is not
+ unlocked twice when using whisper mode.
+
+2008-04-25 15:53 +0000 [r114649] Tilghman Lesher <tlesher@digium.com>
+
+ * configs/zapata.conf.sample, configs/iax.conf.sample,
+ configs/iaxprov.conf.sample, configs/sip.conf.sample: Reference
+ documentation files that actually exist. (closes issue #12516)
+ Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded
+ by linuxmaniac (license 472)
+
+2008-04-24 21:35 +0000 [r114624-114632] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Re-invite RTP during a masquerade so that,
+ for instance, an AMI redirect of two channels which are natively
+ bridged will preserve audio on both channels. This prevents a
+ problem with Asterisk not re-inviting due to one of the channels
+ having being a zombie. (closes issue #12513) Reported by:
+ mneuhauser Patches:
+ asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by
+ mneuhauser (license 425)
+
+ * apps/app_queue.c: Output of channel variables when
+ eventwhencalled=vars was set was being truncated two characters.
+ This patch corrects the problem. (closes issue #12493) Reported
+ by: davidw
+
+ * channels/chan_local.c: Resolve a deadlock in chan_local by
+ releasing the channel lock temporarily. (closes issue #11712)
+ Reported by: callguy Patches: 11712.patch uploaded by putnopvut
+ (license 60) Tested by: acunningham
+
+2008-04-24 19:53 +0000 [r114621] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_local.c: Ensure that when we set the accountcode,
+ it actually shows up in the CDR. (Fix for AMI Originate) (Closes
+ issue #12007)
+
+2008-04-24 15:55 +0000 [r114608] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix a silly mistake in a change I made
+ yesterday that caused chan_iax2 to blow up very quickly. (issue
+ #12515)
+
+2008-04-24 14:55 +0000 [r114603] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Only have one max-forwards header in
+ outbound REFERs. Discovered in the Asterisk SIP Masterclass in
+ Orlando. Thanks Joe!
+
+2008-04-23 22:18 +0000 [r114597-114600] Russell Bryant <russell@digium.com>
+
+ * main/http.c: Improve some broken cookie parsing code. Previously,
+ manager login over HTTP would only work if the mansession_id
+ cookie was first. Now, the code builds a list of all of the
+ cookies in the Cookie header. This fixes a problem observed by
+ users of the Asterisk GUI. (closes AST-20)
+
+ * apps/app_chanspy.c, main/http.c: Fix an issue that caused getting
+ the correct next channel to not always work. Also, remove setting
+ the amount of time to wait for a digit from 5 seconds back down
+ to 1/10 of a second. I believe this was so the beep didn't get
+ played over and over really fast, but a while back I put in
+ another fix for that issue. (closes issue #12498) Reported by:
+ jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by
+ jsmith (license 15)
+
+2008-04-23 18:28 +0000 [r114594] Jason Parker <jparker@digium.com>
+
+ * res/res_musiconhold.c: Fix reload/unload for res_musiconhold
+ module. (closes issue #11575) Reported by: sunder Patches:
+ M11575_14_rev3.diff uploaded by junky (license 177)
+ bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176)
+
+2008-04-23 17:55 +0000 [r114587-114591] Russell Bryant <russell@digium.com>
+
+ * main/manager.c, include/asterisk/manager.h: Store the manager
+ session ID explicitly as 4 byte ID instead of a ulong. The
+ mansession_id cookie is coded to be limited to 8 characters of
+ hex, and this could break logins from 64-bit machines in some
+ cases. (inspired by AST-20)
+
+ * channels/chan_iax2.c: Fix find_callno_locked() to actually return
+ the callno locked in some more cases.
+
+2008-04-23 16:51 +0000 [r114584] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Add 502 support for both directions, not
+ only one... (see r114571)
+
+2008-04-23 14:54 +0000 [r114579] Joshua Colp <jcolp@digium.com>
+
+ * main/pbx.c: Instead of stopping dialplan execution when SayNumber
+ attempts to say a large number that it can not print out a
+ message informing the user and continue on. (closes issue #12502)
+ Reported by: bcnit
+
+2008-04-22 23:51 +0000 [r114571] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Treat a 502 just like a 503, when it comes
+ to processing a response code
+
+2008-04-22 22:15 +0000 [r114522-114558] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: When we receive a full frame that is
+ supposed to contain our call number, ensure that it has the
+ correct one. (closes issue #10078) (AST-2008-006)
+
+ * main/rtp.c, main/channel.c, formats/format_pcm.c, main/file.c: I
+ thought I was going to be able to leave 1.4 alone, but that was
+ not the case. I ran into some problems with G.722 in 1.4, so I
+ have merged in all of the fixes in this area that I have made in
+ trunk/1.6.0, and things are happy again.
+
+ * res/res_musiconhold.c: Trivial change to read the number of
+ samples from a frame before calling ast_write()
+
+ * res/res_features.c: After a parked call times out, allow the call
+ back to the parker to time out. (closes issue #10890)
+
+ * channels/chan_iax2.c: If the dial string passed to the call
+ channel callback does not indicate an extension, then consider
+ the extension on the channel before falling back to the default.
+ (closes issue #12479) Reported by: darren1713 Patches:
+ exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license
+ 116)
+
+ * channels/chan_sip.c, include/asterisk/sched.h: Merge changes from
+ team/russell/issue_9520 These changes make sure that the
+ reference count for sip_peer objects properly reflects the fact
+ that the peer is sitting in the scheduler for a scheduled
+ callback for qualifying peers or for expiring registrations.
+ Without this, it was possible for these callbacks to happen at
+ the same time that the peer was being destroyed. This was
+ especially likely to happen with realtime peers, and for people
+ making use of the realtime prune CLI command. (closes issue
+ #9520) Reported by: kryptolus Committed patch by me
+
+2008-04-21 14:39 +0000 [r114322] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only drop audio if we receive it without a
+ progress indication. We allow other frames through such as DTMF
+ because they may be needed to complete the call. (closes issue
+ #12440) Reported by: aragon
+
+2008-04-19 13:57 +0000 [r114297-114299] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_playback.c: Ensure that help text terminates with a
+ newline
+
+ * res/res_musiconhold.c: MOH usage information needs a terminating
+ newline, or else "asterisk -rx 'help moh reload'" will hang.
+ Reported via -dev list, fixed by me.
+
+2008-04-18 21:48 +0000 [r114275-114284] Russell Bryant <russell@digium.com>
+
+ * main/manager.c: Don't destroy a manager session if poll() returns
+ an error of EAGAIN.
+
+ * Makefile: ensure directories are created before we try to install
+ stuff into them
+
+ * Makefile: SUBDIRS_INSTALL is already listed as a subtarget for
+ bininstall
+
+2008-04-18 17:44 +0000 [r114257] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c, main/callerid.c: Clearing up error messages
+ so they make a bit more sense. Also removing a redundant error
+ message. Issue AST-15
+
+2008-04-18 15:24 +0000 [r114248] Russell Bryant <russell@digium.com>
+
+ * channels/chan_agent.c: Ensure that we don't ast_strdupa(NULL)
+ (closes issue #12476) Reported by: davidw Patch by me
+
+2008-04-18 13:33 +0000 [r114245] Sean Bright <sean.bright@gmail.com>
+
+ * channels/chan_sip.c: Only complete the SIP channel name once for
+ 'sip show channel <channel>'
+
+2008-04-18 06:49 +0000 [r114242] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_setcallerid.c: For consistency sake, ensure that the
+ values that ${CALLINGPRES} returns are valid as an input to
+ SetCallingPres. (Closes issue #12472)
+
+2008-04-17 22:15 +0000 [r114230] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c: Remove redundant safety net. The check for
+ the autoservice channel list state accomplishes the same goal in
+ a better way. (issue #12470) Reported By: atis
+
+2008-04-17 21:03 +0000 [r114207-114226] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Declaration of the peer channel in this scope
+ was making it so the peer variable defined in the outer scope was
+ never set properly, therefore making iterating through the
+ channel list always restart from the beginning. This bug would
+ have affected anyone who called chanspy without specifying a
+ first argument. (closes issue #12461) Reported by: stever28
+
+ * main/frame.c, include/asterisk/dsp.h: Add prototype for
+ ast_dsp_frame_freed. I'm not sure how this was compiling
+ before...
+
+ * main/dsp.c, main/frame.c, include/asterisk/frame.h: It was
+ possible for a reference to a frame which was part of a freed DSP
+ to still be referenced, leading to memory corruption and eventual
+ crashes. This code change ensures that the dsp is freed when we
+ are finished with the frame. This change is very similar to a
+ change Russell made with translators back a month or so ago.
+ (closes issue #11999) Reported by: destiny6628 Patches:
+ 11999.patch uploaded by putnopvut (license 60) Tested by:
+ destiny6628, victoryure
+
+2008-04-17 16:23 +0000 [r114204] Russell Bryant <russell@digium.com>
+
+ * Makefile: Fix the bininstall target to install from subdirs, as
+ well. (closes issue AST-8, patch from bmd at switchvox)
+
+2008-04-17 13:42 +0000 [r114198] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Use keepalives effectively in order diagnose
+ bug #12432.
+
+2008-04-17 12:56 +0000 [r114195] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_agi.c: Add special case for when the agi cannot be
+ executed, to comply with the documentation that we return failure
+ in that case. (closes issue #12462) Reported by: fmueller
+ Patches: 20080416__bug12462.diff.txt uploaded by Corydon76
+ (license 14) Tested by: fmueller
+
+2008-04-17 10:51 +0000 [r114191] Sean Bright <sean.bright@gmail.com>
+
+ * apps/app_chanspy.c: Make sure we have enough room for the
+ recording's filename.
+
+2008-04-16 20:46 +0000 [r114184] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: use the ZT_SET_DIALPARAMS ioctl properly by
+ initializing the structure to all zeroes in case it contains
+ fields that we don't write values into (which it does as of
+ Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian
+
+2008-04-16 19:59 +0000 [r114180] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_vpb.cc: Backport revisions for latest vpb drivers
+ to 1.4 (Closes issue #12457)
+
+2008-04-16 17:30 +0000 [r114173] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Fix "fallthrough" behavior here, so config
+ options in a previously configured user don't override settings
+ in general. (closes issue #12458) Reported by: tzafrir Patches:
+ chanzap_users_sections.diff uploaded by tzafrir (license 46)
+
+2008-04-16 14:10 +0000 [r114167] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Include the proper headers for using mkdir on
+ FreeBSD. (closes issue #12430) Reported by: ys Patches:
+ app_meetme.c.diff uploaded by ys (license 281)
+
+2008-04-15 20:26 +0000 [r114148] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Handle subscribe queues in all situations...
+ Thanks to festr_ on irc for telling me about this bug.
+
+2008-04-15 17:17 +0000 [r114120-114138] Jason Parker <jparker@digium.com>
+
+ * contrib/scripts/autosupport: Update Digium autosupport script,
+ for more useful information. (closes issue #12452) Reported by:
+ angler Patches: autosupport.diff uploaded by angler (license 106)
+
+ * apps/app_queue.c: Allow autofill to work in the general section
+ of queues.conf. Additionally, don't try to (re)set options when
+ they have empty values in realtime (all unset columns would have
+ an empty value). (closes issue #12445) Reported by: atis Patches:
+ 12445-autofill.diff uploaded by qwell (license 4)
+
+ * channels/chan_h323.c: The call_token on the pvt can occasionally
+ be NULL, causing a crash. If it is NULL, we can skip this
+ channel, since it can't the one we're looking for. (closes issue
+ #9299) Reported by: vazir
+
+2008-04-14 17:41 +0000 [r114106-114117] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Increase the retry count when attempting to show
+ channels. This apparently cleared an issue someone was seeing
+ when attempting to show channels when the load was high. (closes
+ issue #11667) Reported by: falves11 Patches: 11677.txt uploaded
+ by russell (license 2) Tested by: falves11
+
+ * apps/app_dial.c, apps/app_queue.c: If the datastore has been
+ moved to another channel due to a masquerade, then freeing the
+ datastore here causes an eventual double free when the new
+ channel hangs up. We should only free the datastore if we were
+ able to successfully remove it from the channel we are
+ referencing (i.e. the datastore was not moved). (closes issue
+ #12359) Reported by: pguido
+
+ * main/channel.c: Save a local copy of the generate callback prior
+ to unlocking the channel in case the generate callback goes NULL
+ on us after the channel is unlocked. Thanks to Russell for
+ pointing this need out to me.
+
+2008-04-14 14:52 +0000 [r114100-114103] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: It is possible for the remote side to say
+ they want T38 but not give any capabilities. (closes issue
+ #12414) Reported by: MVF
+
+ * main/rtp.c: Don't change the SSRC when a new source comes into
+ play, this might happen quite often and depending on the remote
+ side... they might not like this. (closes issue #12353) Reported
+ by: dimas
+
+2008-04-11 22:32 +0000 [r114083] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_iax2.c: Several places in the code called
+ find_callno() (which releases the lock on the pvt structure) and
+ then immediately locked the call and did things with it.
+ Unfortunately, the call can disappear between the find_callno and
+ the lock, causing Bad Stuff(tm) to happen. Added
+ find_callno_locked() function to return the callno withtout
+ unlocking for instances that it is needed. (issue #12400)
+ Reported by: ztel
+
+2008-04-11 21:35 +0000 [r114072] Jason Parker <jparker@digium.com>
+
+ * main/pbx.c: It's possible that a channel can have an async goto
+ on the successful execution of an application as well. Closes
+ issue #12172.
+
+2008-04-11 15:44 +0000 [r114045-114063] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_features.c: Fix a race condition that may happen between
+ a sip hangup and a "core show channel" command. This patch adds
+ locking to prevent the resulting crash. (closes issue #12155)
+ Reported by: tsearle Patches: show_channels_crash2.patch uploaded
+ by tsearle (license 373) Tested by: tsearle
+
+ * main/utils.c, include/asterisk/lock.h: Fix 1.4 build when
+ LOW_MEMORY is enabled.
+
+ * channels/chan_sip.c: Be sure that we're not about to set
+ bridgepvt NULL prior to dereferencing it. (closes issue #11775)
+ Reported by: fujin
+
+2008-04-10 17:26 +0000 [r114035] Jason Parker <jparker@digium.com>
+
+ * main/file.c: Only try to prefix language if we are not using an
+ absolute path (suffix it otherwise).
+ en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes
+ issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff
+ uploaded by qwell (license 4) Tested by: kuj, qwell
+
+2008-04-10 15:58 +0000 [r114021-114032] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Forgot the 1.4 branch for russian language
+ fix. (closes issue #12404) Reported by: IgorG Patches:
+ voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20)
+
+ * apps/app_meetme.c: Create the directory where name recordings
+ will go if it does not exist. (closes issue #12311) Reported by:
+ rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4)
+
+ * channels/chan_sip.c: Don't add custom URI options if they don't
+ exist OR they are empty. (closes issue #12407) Reported by:
+ homesick Patches: uri_options-1.4.diff uploaded by homesick
+ (license 91)
+
+2008-04-09 20:54 +0000 [r113927] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: We need to set the persistant_route [sic]
+ parameter for the sip_pvt during the initial INVITE, no matter if
+ we're building the route set from an INVITE request or response.
+ (closes issue #12391) Reported by: benjaminbohlmann Tested by:
+ benjaminbohlmann
+
+2008-04-09 18:57 +0000 [r113874] Tilghman Lesher <tlesher@digium.com>
+
+ * cdr/cdr_csv.c, configs/cdr.conf.sample: If the [csv] section does
+ not exist in cdr.conf, then an unload/load sequence is needed to
+ correct the problem. Track whether the load succeeded with a
+ variable, so we can fix this with a simple reload event, instead.
+
+2008-04-09 16:50 +0000 [r113784] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: If we receive an AUTHREQ from the remote
+ server and we are unable to reply (for example they have a secret
+ configured, but we do not) then queue a hangup frame on the
+ Asterisk channel. This will cause the channel to hangup and a
+ HANGUP to be sent via IAX2 to the remote side which is the proper
+ thing to do in this scenario. (closes issue #12385) Reported by:
+ viraptor
+
+2008-04-09 14:40 +0000 [r113681] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: If Asterisk receives a 488 on an INVITE (not
+ a reinvite), then we should not send a BYE. (closes issue #12392)
+ Reported by: fnordian Patches: chan_sip.patch uploaded by
+ fnordian (license 110) with small modification from me
+
+2008-04-09 01:34 +0000 [r113596] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_iax2.c: Initialize fr->cacheable to make valgrind
+ happy
+
+2008-04-08 19:07 +0000 [r113507] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_parkandannounce.c: Fix potential buffer overflow that
+ could happen if more than 100 announce files were specified when
+ calling ParkAndAnnounce. This overflow is not exploitable
+ remotely and so there is no need for a security advisory. (closes
+ issue #12386) Reported by: davidw
+
+2008-04-08 18:48 +0000 [r113402-113504] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c: Add a little more that is required for
+ previously added devices.
+
+ * channels/chan_skinny.c: Add support for several new(ish) devices
+ - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver
+ for providing me the required information.
+
+ * main/asterisk.c: Work around some silliness caused by
+ sys/capability.h - this should fix compile errors a number of
+ users have been experiencing.
+
+2008-04-08 16:51 +0000 [r113348-113399] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/astgenkey.8: Add security note on astgenkey's
+ manpage. (closes issue #12373) Reported by: lmamane Patches:
+ 20080406__bug12373.diff.txt uploaded by Corydon76 (license 14)
+
+ * channels/chan_sip.c: Move check for still-bridged channels out a
+ little further, to avoid possible deadlocks. (Closes issue
+ #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: callguy
+
+2008-04-08 15:03 +0000 [r113296] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/slinfactory.h, main/slinfactory.c,
+ main/audiohook.c: If audio suddenly gets fed into one side of a
+ channel after a lapse of frames flush the other factory so that
+ old audio does not remain in the factory causing the sync code to
+ not execute. (closes issue #12296) Reported by: jvandal
+
+2008-04-07 21:34 +0000 [r113240] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: (closes issue #12362) [redo of 113012] This
+ fixes a for loop (in realtime_peer) to check all the
+ ast_variables the loop was intending to test rather than just the
+ first one. The change exposed the problem of calling memcpy on a
+ NULL pointer, in this case the passed in sockaddr_in struct which
+ is now checked.
+
+2008-04-07 18:00 +0000 [r113118] Jason Parker <jparker@digium.com>
+
+ * channels/chan_skinny.c, configs/skinny.conf.sample: Allow
+ playback with noanswer (and add earlyrtp option). (closes issue
+ #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn
+ (license 30) Tested by: pj, qwell, DEA, wedhorn
+
+2008-04-07 17:51 +0000 [r113117] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_strings.c: Force ast_mktime() to check for DST, since
+ strptime(3) does not. (Closes issue #12374)
+
+2008-04-07 16:08 +0000 [r113065] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: This fix prevents a deadlock that was experienced
+ in chan_local. There was deadlock prevention in place in
+ chan_local, but it would not work in a specific case because the
+ channel was recursively locked. By unlocking the channel prior to
+ calling the generator's generate callback in
+ ast_read_generator_actions(), we prevent the recursive locking,
+ and therefore the deadlock. (closes issue #12307) Reported by:
+ callguy Patches: 12307.patch uploaded by putnopvut (license 60)
+ Tested by: callguy
+
+2008-04-07 15:16 +0000 [r113012] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: (closes issue #12362) (closes issue #12372)
+ Reported by: vinsik Tested by: tecnoxarxa This one line change
+ makes an if inside a for loop (in realtime_peer) check all the
+ ast_variables the loop was intending to test rather than just the
+ first one.
+
+2008-04-04 19:26 +0000 [r112766-112820] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c: Free newly allocated channel before
+ returning
+
+ * channels/chan_gtalk.c: Prevent call connections when codecs don't
+ match. (closes issue #10604) Reported by: keepitcool Patches:
+ branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested
+ by: phsultan
+
+2008-04-04 00:52 +0000 [r112709-112711] Joshua Colp <jcolp@digium.com>
+
+ * main/Makefile: Pass in the path to Zaptel for systems that
+ install Zaptel headers in a separate location.
+
+ * main/asterisk.c: One thing at a time... let's get 1.4 building.
+
+2008-04-03 23:57 +0000 [r112689] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * main/asterisk.c: add a Zaptel timer check to verify the timer is
+ responding when Zaptel support is compiled into Asterisk and
+ Zaptel drivers are loaded. This will help people not waste their
+ valuable time debugging side effects.
+
+2008-04-03 14:32 +0000 [r112393-112599] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c: Fix the testing of the "res" variable so
+ that it is more logically correct and makes the correct warning
+ and debug messages print. (closes issue #12361) Reported by:
+ one47 Patches: chan_zap_deferred_digit.patch uploaded by one47
+ (license 23)
+
+ * main/manager.c: Fix a race condition in the manager. It is
+ possible that a new manager event could be appended during a
+ brief time when the manager is not waiting for input. If an event
+ comes during this period, we need to set an indicator that there
+ is an event pending so that the manager doesn't attempt to wait
+ forever for an event that already happened. (closes issue #12354)
+ Reported by: bamby Patches: manager_race_condition.diff uploaded
+ by bamby (license 430) (comments added by me)
+
+ * apps/app_queue.c: Ensure that there is no timeout if none is
+ specified. (closes issue #12349) Reported by: johnlange
+
+2008-04-01 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.19 released.
+
+2008-03-28 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.19-rc4 released.
+
+2008-03-28 16:19 +0000 [r111658] Jason Parker <jparker@digium.com>
+
+ * formats/format_wav_gsm.c: The file size of WAV49 does not need to
+ be an even number. (closes issue #12128) Reported by: mdu113
+ Patches: 12128-noevenlength.diff uploaded by qwell (license 4)
+ Tested by: qwell, mdu113
+
+2008-03-28 14:35 +0000 [r111442-111605] Tilghman Lesher <tlesher@digium.com>
+
+ * doc/valgrind.txt: Update debugging text, since Valgrind
+ eliminated the --log-file-exactly option. (Closes issue #12320)
+
+ * main/acl.c: For FreeBSD, at least, the ifa_addr element could be
+ NULL. (closes issue #12300) Reported by: festr Patches:
+ acl.c.patch uploaded by festr (license 443)
+
+2008-03-27 13:03 +0000 [r111341-111391] Steve Murphy <murf@digium.com>
+
+ * apps/app_playback.c, main/pbx.c: These small documentation
+ updates made in response to a query in asterisk-users, where a
+ user was using Playback, but needed the features of Background,
+ and had no idea that Background existed, or that it might provide
+ the features he needed. I thought the best way to avert these
+ kinds of queries was to provide "See Also" references in all
+ three of "Background", "Playback", "WaitExten". Perhaps a project
+ to do this with all related apps is in order.
+
+ * pbx/pbx_ael.c, include/asterisk/ael_structs.h: (closes issue
+ #12302) Reported by: pj Tested by: murf These changes will set a
+ channel variable ~~EXTEN~~ just before generating code for a
+ switch, with the value of ${EXTEN}. The exten is marked as having
+ a switch, and ever after that, till the end of the exten, we
+ substitute any ${EXTEN} with ${~~EXTEN~~} instead in application
+ arguments; (and the ${EXTEN: also). The reason for this, is that
+ because switches are coded using separate extensions to provide
+ pattern matching, and jumping to/from these switch extensions
+ messes up the ${EXTEN} value, which blows the minds of users.
+
+2008-03-27 00:25 +0000 [r111245-111280] Jason Parker <jparker@digium.com>
+
+ * main/frame.c: Put this flag back so we don't change the API.
+
+ * main/frame.c: Remove excessive smoother optimization that was
+ causing audio glitches (small "pops") after (about 200ms later)
+ an "incorrectly" sized frame was received. While it would be very
+ nice to keep this as optimized as possible, it makes no sense for
+ the smoother to be dropping random bits of audio like this. Isn't
+ that the whole point of a smoother? Closes issue #12093.
+
+2008-03-26 19:55 +0000 [r111129] Joshua Colp <jcolp@digium.com>
+
+ * contrib/scripts/autosupport: Update autosupport script. (closes
+ issue #12310) Reported by: angler Patches: autosupport.diff
+ uploaded by angler (license 106)
+
+2008-03-26 19:51 +0000 [r111126] Kevin P. Fleming <kpfleming@digium.com>
+
+ * /, UPGRADE.txt: Merged revisions 111125 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar
+ 2008) | 2 lines update UPGRADE notes to document usage of the
+ script ........
+
+2008-03-26 19:37 +0000 [r111049-111121] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: This code change is made just for
+ clarification. It does exactly the same thing as before. It just
+ doesn't look as wrong.
+
+ * apps/app_voicemail.c: Add a lock to the vm_state structure and
+ use the lock around mail_open calls to prevent concurrent access
+ of the same mailstream. This, along with trunk's ability to
+ configure TCP timeouts for IMAP storage will help to prevent
+ crashes and hangs when using voicemail with IMAP storage. (closes
+ issue #10487) Reported by: ewilhelmsen
+
+2008-03-26 19:06 +0000 [r111024] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added):
+ Merged revisions 111019 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar
+ 2008) | 2 lines add a script to make getting the iLBC source code
+ simple for end users ........
+
+2008-03-26 19:04 +0000 [r111014-111020] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: If we are requested to authenticate a
+ reinvite make sure that it contains T38 SDP if need be. (closes
+ issue #11995) Reported by: fall
+
+ * channels/chan_iax2.c: Make sure that full video frames are sent
+ whenever the 15 bit timestamp rolls over. (closes issue #11923)
+ Reported by: mihai Patches: asterisk-fullvideo.patch uploaded by
+ mihai (license 94)
+
+2008-03-26 17:43 +0000 [r110880-110962] Kevin P. Fleming <kpfleming@digium.com>
+
+ * UPGRADE.txt: add note that the user will need to enable
+ codec_ilbc to get it to build
+
+ * codecs/ilbc/StateConstructW.h (removed),
+ codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/packing.h
+ (removed), codecs/ilbc/getCBvec.c (removed),
+ codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/enhancer.c
+ (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c
+ (removed), codecs/ilbc/getCBvec.h (removed),
+ codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h
+ (removed), codecs/ilbc/FrameClassify.c (removed),
+ codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed),
+ codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h
+ (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c
+ (removed), codecs/ilbc/anaFilter.c (removed),
+ codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c
+ (removed), codecs/ilbc/doCPLC.h (removed),
+ codecs/ilbc/anaFilter.h (removed), UPGRADE.txt,
+ codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/constants.c
+ (removed), codecs/ilbc/createCB.h (removed), CHANGES,
+ codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/constants.h
+ (removed), codecs/Makefile, codecs/ilbc/iCBSearch.c (removed),
+ codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed),
+ codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c
+ (removed), codecs/ilbc/iCBSearch.h (removed),
+ codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed),
+ codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c
+ (removed), codecs/ilbc/hpOutput.h (removed),
+ codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c,
+ codecs/ilbc/LPCencode.h (removed), codecs/ilbc/StateSearchW.h
+ (removed), codecs/ilbc/iCBConstruct.c (removed),
+ codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h
+ (removed), codecs/ilbc/syntFilter.h (removed),
+ codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.c
+ (removed): Merged revisions 110869 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar
+ 2008) | 2 lines due to licensing restrictions, we cannot
+ distribute the source code for iLBC encoding and decoding... so
+ remove it, and add instructions on how the user can obtain it
+ themselves ........
+
+2008-03-25 22:51 +0000 [r110779] Jason Parker <jparker@digium.com>
+
+ * cdr/cdr_custom.c: Make file access in cdr_custom similar to
+ cdr_csv. Fixes issue #12268. Patch borrowed from r82344
+
+2008-03-25 20:03 +0000 [r110727] Jeff Peeler <jpeeler@digium.com>
+
+ * channels/chan_sip.c: This one line change makes an if inside a
+ for loop (in realtime_peer) check all the ast_variables the loop
+ was intending to test rather than just the first one.
+
+2008-03-25 15:40 +0000 [r110635] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: When reverting a commit, I accidentally left
+ in this bit which was an experiment to see what would happen. It
+ passed the compile test, and I didn't notice I had left this
+ change in too. So this is a revert of a revert...sort of.
+
+2008-03-25 14:37 +0000 [r110628] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/options.h, main/asterisk.c, Makefile,
+ main/app.c: Add an option (transmit_silence) which transmits
+ silence during both Record() and DTMF generation. The reason this
+ is an option is that in order to transmit silence we have to
+ setup a translation path. This may not be needed/wanted in all
+ cases. (closes issue #10058) Reported by: tracinet
+
+2008-03-24 19:17 +0000 [r110618] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: This is a revert for revision 108288. The
+ reason is that that revision was not for an actual bug fix per
+ se, and so it really should not have been in 1.4 in the first
+ place. Plus, people who compile with DO_CRASH are more likely to
+ encounter a crash due to this change. While I think the usage of
+ DO_CRASH in ast_sched_del is a bit absurd, this sort of change is
+ beyond the scope of 1.4 and should be done instead in a developer
+ branch based on trunk so that all scheduler functions are fixed
+ at once. I also am reverting the change to trunk and 1.6 since
+ they also suffer from the DO_CRASH potential. (closes issue
+ #12272) Reported by: qq12345
+
+2008-03-24 17:34 +0000 [r110614] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Turn a NOTICE into a DEBUG message.
+
+2008-03-21 14:32 +0000 [r110474] Jason Parker <jparker@digium.com>
+
+ * codecs/gsm/Makefile: Don't attempt to do optimizations of gsm on
+ mips platforms either. (closes issue #12270) Reported by:
+ zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt
+ (license 33)
+
+2008-03-20 23:13 +0000 [r110163-110395] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c: Shorten the ast_waitfor() timeout from 500 ms
+ to 50 ms in the autoservice thread. This really should not make a
+ difference except in very rare cases. That case would be that all
+ of the channels in autoservice are not generating any frames. In
+ that case, this change reduces the potential amount of time that
+ a thread waits in ast_autoservice_stop() for the autoservice
+ thread to wrap back around to the beginning of its loop. (closes
+ issue #12266, reported by dimas)
+
+ * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions
+ 110335 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008)
+ | 6 lines Fix some very broken code that was introduced in 1.2.26
+ as a part of the security fix. The dnsmgr is not appropriate
+ here. The dnsmgr takes a pointer to an address structure that a
+ background thread continuously updates. However, in these cases,
+ a stack variable was passed. That means that the dnsmgr thread
+ would be continuously writing to bogus memory. ........
+
+ * apps/app_meetme.c: Fix a bug where when calls on the trunk side
+ hang up while on hold, the state is not properly reflected.
+ (closes issue #11990, reported by anakaoka, patched by me)
+
+2008-03-19 20:33 +0000 [r110083] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Add a missing unlock in the case that memory
+ allocation fails in app_chanspy. Thanks to Russell for confirming
+ that this was an issue.
+
+2008-03-19 19:11 +0000 [r110019-110035] Joshua Colp <jcolp@digium.com>
+
+ * res/res_musiconhold.c: Add sanity checking for position resuming.
+ We *have* to make sure that the position does not exceed the
+ total number of files present, and we have to make sure that the
+ position's filename is the same as previous. These values can
+ change if a music class is reloaded and give unpredictable
+ behavior. (closes issue #11663) Reported by: junky
+
+ * main/rtp.c: Make sure that the mark bit does not incorrectly
+ cause video frame timestamps to be calculated as if they are
+ audio frames. (closes issue #11429) Reported by: sperreault
+ Patches: 11429-frametype.diff uploaded by qwell (license 4)
+
+2008-03-19 17:12 +0000 [r109973] Jason Parker <jparker@digium.com>
+
+ * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml
+ (added): People report bugs about Asterisk crashing with DO_CRASH
+ enabled was getting a little silly... Now we only show certain
+ cflags when you run configure with --enable-dev-mode
+ (corresponding menuselect change to follow)
+
+2008-03-19 15:41 +0000 [r109908] Steve Murphy <murf@digium.com>
+
+ * main/config.c: (closes issue #11442) Reported by: tzafrir
+ Patches: 11442.patch uploaded by murf (license 17) Tested by:
+ murf I didn't give tzafrir very much time to test this, but if he
+ does still have remaining issues, he is welcome to re-open this
+ bug, and we'll do what is called for. I reproduced the problem,
+ and tested the fix, so I hope I am not jumping by just going
+ ahead and committing the fix. The problem was with what file_save
+ does with templates; firstly, it tended to print out multiple
+ options: [my_category](!)(templateref) instead of
+ [my_category](!,templateref) which is fixed by this patch.
+ Nextly, the code to suppress output of duplicate declarations
+ that would occur because the reader copies inherited declarations
+ down the hierarchy, was not working. Thus: [master-template](!)
+ mastervar = bar [template](!,master-template) tvar = value
+ [cat](template) catvar = val would be rewritten as: ;! ;!
+ Automatically generated configuration file ;! Filename:
+ experiment.conf (/etc/asterisk/experiment.conf) ;! Generator:
+ Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;!
+ [master-template](!) mastervar = bar
+ [template](!,master-template) mastervar = bar tvar = value
+ [cat](template) mastervar = bar tvar = value catvar = val This
+ has been fixed. Since the config reader 'explodes' inherited vars
+ into the category, users may, in certain circumstances, see
+ output different from what they originally entered, but it should
+ be both correct and equivalent.
+
+2008-03-19 04:06 +0000 [r109763-109838] Russell Bryant <russell@digium.com>
+
+ * main/utils.c: Tweak spacing in a recent change because I'm very
+ picky.
+
+ * apps/app_chanspy.c: Fix one place where the chanspy datastore
+ isn't removed from a channel. (issue #12243, reported by atis,
+ patch by me)
+
+2008-03-18 20:52 +0000 [r109713] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: This patch makes it so that all queue member
+ status changes are handled through device state code. This
+ removes several problems people were seeing where their queue
+ members would get into an "unknown" state. Huge props go to atis
+ on this one since he was the one who found the code section that
+ was causing the problem and proposed the solution. I just wrote
+ what he suggested :) (closes issue #12127) Reported by: atis
+ Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested
+ by: atis, jvandal
+
+2008-03-18 19:23 +0000 [r109648] Jason Parker <jparker@digium.com>
+
+ * codecs/log2comp.h: Allow codecs that use log2comp (g726) to
+ compile correctly on x86 with gcc4 optimizations. (closes issue
+ #12253) Reported by: fossil Patches: log2comp.patch uploaded by
+ fossil (license 140)
+
+2008-03-18 17:58 +0000 [r109575] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_agent.c: Make sure an agent doesn't try to send
+ dtmf to a NULL channel closes issue #12242 Reported by Yourname
+
+2008-03-18 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.19-rc3 released.
+
+2008-03-18 16:25 +0000 [r109482] Terry Wilson <twilson@digium.com>
+
+ * include/asterisk/astobj.h: Fix character string being treated ad
+ format string
+
+2008-03-18 15:10 +0000 [r109393] Jason Parker <jparker@digium.com>
+
+ * /, channels/chan_sip.c: Merged revisions 109391 via svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r109391 | qwell | 2008-03-18 10:08:41 -0500 (Tue, 18 Mar 2008) |
+ 3 lines Do not return with a successful authentication if the
+ From header ends up empty. (AST-2008-003) ........
+
+2008-03-18 14:58 +0000 [r109386] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c, channels/chan_sip.c: Put a maximum limit on the
+ number of payloads accepted, and also make sure a given payload
+ does not exceed our maximum value. (AST-2008-002)
+
+2008-03-18 06:37 +0000 [r109309] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael-test/ael-ntest23 (added),
+ pbx/ael/ael-test/ael-ntest23/t1/a.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t1/b.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t1/c.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t2/d.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t2/e.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t2/f.ael (added),
+ pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael_lex.c,
+ pbx/ael/ael-test/ael-ntest23/t3/g.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t3/h.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t3/i.ael (added), pbx/ael/ael.flex,
+ pbx/ael/ael-test/ael-ntest23/t3/j.ael (added),
+ pbx/ael/ael-test/ael-ntest23/qq.ael (added),
+ pbx/ael/ael-test/ael-ntest23/t1 (added),
+ pbx/ael/ael-test/ael-ntest23/t2 (added),
+ pbx/ael/ael-test/ael-ntest23/t3 (added),
+ pbx/ael/ael-test/ael-ntest23/extensions.ael (added): (closes
+ issue #11903) Reported by: atis Many thanks to atis for spotting
+ this problem and reporting it. The fix was to straighten out how
+ items are placed on and removed from the file stack. Regressions
+ as well as the provided test case helped to straighten out all
+ code paths. valgrind was used to make sure all memory allocated
+ was freed. Sorry for not solving this earlier. I got distracted.
+ Added the ntest23 regression test, which is mainly a copy of
+ ntest22, but with a few juicy errors thrown in, to replicate the
+ kind of error that atis spotted.
+
+2008-03-17 22:05 +0000 [r109226] Mark Michelson <mmichelson@digium.com>
+
+ * main/utils.c: Fix a logic flaw in the code that stores lock info
+ which is displayed via the "core show locks" command. The idea
+ behind this section of code was to remove the previous lock from
+ the list if it was a trylock that had failed. Unfortunately,
+ instead of checking the status of the previous lock, we were
+ referencing the index immediately following the previous lock in
+ the lock_info->locks array. The result of this problem, under the
+ right circumstances, was that the lock which we currently in the
+ process of attempting to acquire could "overwrite" the previous
+ lock which was acquired. While this does not in any way affect
+ typical operation, it *could* lead to misleading "core show
+ locks" output.
+
+2008-03-17 17:55 +0000 [r109171] Michiel van Baak <michiel@vanbaak.info>
+
+ * channels/chan_skinny.c: Update the directory of placed calls on
+ skinny phones when dialing a channel that does not provide
+ progress (analog ZAP lines) The phone does handle the double
+ update on calls to channels that do provide progress and wont
+ insert duplicate items (closes issue #12239) Reported by: DEA
+ Patches: chan_skinny-call-log.txt uploaded by DEA (license 3)
+
+2008-03-17 16:24 +0000 [r109107] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: 200 OKs in response to a reinvite need to be
+ sent reliably. If the remote side does not receive one the dialog
+ will be torn down. (closes issue #12208) Reported by: atrash
+
+2008-03-17 15:15 +0000 [r109057] Jason Parker <jparker@digium.com>
+
+ * main/file.c: Backport revision 106439 from trunk. I didn't
+ realize this was broken in 1.4 as well. Closes issue #12222.
+
+2008-03-17 14:18 +0000 [r109012] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_chanspy.c: Make sure that we release the lock on the
+ spyee channel if the spyee or spy has hung up (closes issue
+ #12232) Reported by: atis
+
+2008-03-16 21:47 +0000 [r108961] Michiel van Baak <michiel@vanbaak.info>
+
+ * main/dial.c: add missing break to case AST_CONTROL_SRCUPDATE
+ (closes issue #12228) Reported by: andrew Patches: SRC.patch
+ uploaded by andrew (license 240)
+
+2008-03-14 20:09 +0000 [r108792-108796] Russell Bryant <russell@digium.com>
+
+ * channels/chan_oss.c: Fix a channel name issue. chan_oss registers
+ the "Console" channel type, but it created channels with an "OSS"
+ prefix. (closes issue #12194, reported by davidw, patched by me)
+
+ * contrib/init.d/rc.suse.asterisk: Update the SuSE init script to
+ start networking before asterisk, as well. (closes issue #12200,
+ reported by and change suggested by reinerotto)
+
+2008-03-14 16:44 +0000 [r108737] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Fix a race condition in the SIP packet
+ scheduler which could cause a crash. chan_sip uses the scheduler
+ API in order to schedule retransmission of reliable packets (such
+ as INVITES). If a retransmission of a packet is occurring, then
+ the packet is removed from the scheduler and retrans_pkt is
+ called. Meanwhile, if a response is received from the packet as
+ previously transmitted, then when we ACK the response, we will
+ remove the packet from the scheduler and free the packet. The
+ problem is that both the ACK function and retrans_pkt attempt to
+ acquire the same lock at the beginning of the function call. This
+ means that if the ACK function acquires the lock first, then it
+ will free the packet which retrans_pkt is about to read from and
+ write to. The result is a crash. The solution: 1. If the ACK
+ function fails to remove the packet from the scheduler and the
+ retransmit id of the packet is not -1 (meaning that we have not
+ reached the maximum number of retransmissions) then release the
+ lock and yield so that retrans_pkt may acquire the lock and
+ operate. 2. Make absolutely certain that the ACK function does
+ not recursively lock the lock in question. If it does, then
+ releasing the lock will do no good, since retrans_pkt will still
+ be unable to acquire the lock. (closes issue #12098) Reported by:
+ wegbert (closes issue #12089) Reported by: PTorres Patches:
+ 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested
+ by: jvandal
+
+2008-03-14 14:29 +0000 [r108682] Jason Parker <jparker@digium.com>
+
+ * res/res_musiconhold.c: Fix a potential segfault if chan (or
+ chan->music_state) is NULL. Closes issue #12210, credit to
+ edantie for pointing this out.
+
+2008-03-13 21:38 +0000 [r108469-108583] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c, main/channel.c, include/asterisk/channel.h:
+ Fix another issue that was causing crashes in chanspy. This
+ introduces a new datastore callback, called chan_fixup(). The
+ concept is exactly like the fixup callback that is used in the
+ channel technology interface. This callback gets called when the
+ owning channel changes due to a masquerade. Before this was
+ introduced, if a masquerade happened on a channel being spyed on,
+ the channel pointer in the datastore became invalid. (closes
+ issue #12187) (reported by, and lots of testing from atis) (props
+ to file for the help with ideas)
+
+ * channels/chan_sip.c: Make a tweak that gets the LEDs on polycom
+ phones to blink when an extension that has been subscribed to
+ goes on hold. Otherwise, they just stay on like it does when an
+ extension is in use. (closes issue #11263) Reported by: russell
+ Patches: notify_hold.rev1.txt uploaded by russell (license 2)
+ Tested by: russell
+
+ * apps/app_followme.c: Fix a couple uses of sprintf. The second one
+ could actually cause an overflow of a stack buffer. It's not a
+ security issue though, it only depends on your configuration.
+
+2008-03-12 21:53 +0000 [r108227-108288] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: Change AST_SCHED_DEL use to ast_sched_del
+ for autocongestion in chan_sip. The scheduler callback will
+ always return 0. This means that this id is never rescheduled, so
+ it makes no sense to loop trying to delete the id from the
+ scheduler queue. If we fail to remove the item from the queue
+ once, it will fail every single time. (Yes I realize that in this
+ case, the macro would exit early because the id is set to -1 in
+ the callback, but it still makes no sense to use that macro in
+ favor of calling ast_sched_del once and being done with it) This
+ is the first of potentially several such fixes.
+
+ * include/asterisk/sched.h: Added a large comment before the
+ AST_SCHED_DEL macro to explain its purpose as well as when it is
+ appropriate and when it is not appropriate to use it. I also
+ removed the part of the debug message that mentions that this is
+ probably a bug because there are some perfectly legitimate places
+ where ast_sched_del may fail to delete an entry (e.g. when the
+ scheduler callback manually reschedules with a new id instead of
+ returning non-zero to tell the scheduler to reschedule with the
+ same idea). I also raised the debug level of the debug message in
+ AST_SCHED_DEL since it seems like it could come up quite
+ frequently since the macro is probably being used in several
+ places where it shouldn't be. Also removed the redundant line,
+ file, and function information since that is provided by ast_log.
+
+2008-03-12 19:57 +0000 [r108135] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c, main/channel.c: (closes issue #12187,
+ reported by atis, fixed by me after some brainstorming on the
+ issue with mmichelson) - Update copyright info on app_chanspy. -
+ Fix a race condition that caused app_chanspy to crash. The issue
+ was that the chanspy datastore magic that was used to ensure that
+ spyee channels did not disappear out from under the code did not
+ completely solve the problem. It was actually possible for
+ chanspy to acquire a channel reference out of its datastore to a
+ channel that was in the middle of being destroyed. That was
+ because datastore destruction in ast_channel_free() was done near
+ the end. So, this left the code in app_chanspy accessing a
+ channel that was partially, or completely invalid because it was
+ in the process of being free'd by another thread. The following
+ sort of shows the code path where the race occurred:
+ =============================================================================
+ Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy)
+ --------------------------------------||-------------------------------------
+ ast_channel_free() || - remove channel from channel list || -
+ lock/unlock the channel to ensure || that no references retrieved
+ from || the channel list exist. ||
+ --------------------------------------||-------------------------------------
+ || channel_spy() - destroy some channel data || - Lock chanspy
+ datastore || - Retrieve reference to channel || - lock channel ||
+ - Unlock chanspy datastore
+ --------------------------------------||-------------------------------------
+ - destroy channel datastores || - call chanspy datastore d'tor ||
+ which NULL's out the ds' || - Operate on the channel ...
+ reference to the channel || || - free the channel || || || -
+ unlock the channel
+ --------------------------------------||-------------------------------------
+ =============================================================================
+
+2008-03-12 19:16 +0000 [r108086] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_sip.c: if we receive an INVITE with a
+ Content-Length that is not a valid number, or is zero, then don't
+ process the rest of the message body looking for an SDP closes
+ issue #11475 Reported by: andrebarbosa
+
+2008-03-12 18:26 +0000 [r108083] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_mixmonitor.c, include/asterisk/audiohook.h,
+ main/audiohook.c: Add a trigger mode that triggers on both read
+ and write. The actual function that returns the combined audio
+ frame though will wait until both sides have fed in audio, or
+ until one side stops (such as the case when you call Wait).
+ (closes issue #11945) Reported by: xheliox
+
+2008-03-12 16:59 +0000 [r108031] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Destroy the channel lock after the channel
+ datastores. (inspired by issue #12187)
+
+2008-03-12 01:52 +0000 [r107877] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/scripts/iax-friends.sql, contrib/scripts/sip-friends.sql:
+ Document all of the possible realtime fields
+
+2008-03-11 23:37 +0000 [r107714-107826] Jason Parker <jparker@digium.com>
+
+ * doc/voicemail_odbc_postgresql.txt: Update documentation for pgsql
+ ODBC voicemail. (closes issue #12186) Reported by: jsmith
+ Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license
+ 15)
+
+ * channels/chan_gtalk.c: Copy voicemail dependency logic for
+ res_adsi to chan_gtalk (for jabber). (closes issue #12014)
+ Reported by: junky
+
+2008-03-11 20:48 +0000 [r107713] Kevin P. Fleming <kpfleming@digium.com>
+
+ * Makefile.rules, channels/Makefile: get chan_vpb to build properly
+ in dev mode
+
+2008-03-11 20:47 +0000 [r107712] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: Add a newline on a log
+
+2008-03-11 19:20 +0000 [r107582-107646] Joshua Colp <jcolp@digium.com>
+
+ * res/res_features.c: Make sure the visible indication is on the
+ right channel so when the masquerade happens the proper
+ indication is enacted. (closes issue #11707) Reported by: iam
+
+ * apps/app_meetme.c: Add an additional check for setting conference
+ parameter when using the marked user options. It was possible for
+ it to return to a no listen/no talk state if a masquerade
+ happened. (closes issue #12136) Reported by: aragon
+
+ * apps/app_exec.c: Fix a minor spelling error. (closes issue
+ #12183) Reported by: darrylc
+
+2008-03-11 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.19-rc2 released.
+
+2008-03-11 15:18 +0000 [r107352-107472] Kevin P. Fleming <kpfleming@digium.com>
+
+ * apps/app_rpt.c: backport a fix from trunk
+
+ * channels/misdn/isdn_lib.c, codecs/Makefile,
+ channels/chan_misdn.c: fix various other problems found by gcc
+ 4.3
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ apps/app_sms.c: stop checking for mktime() in the configure
+ script... we don't use it, and the test is buggy under gcc 4.3
+
+ * configure, main/Makefile, configure.ac, makeopts.in: check for
+ compiler support for -fno-strict-overflow before using it (tested
+ with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179)
+ Reported by: Netview
+
+ * configure, configure.ac: fix small bug in IMAP toolkit testing
+
+ * main/udptl.c, utils/Makefile, main/Makefile,
+ main/editline/readline.c, pbx/Makefile: fix up various compiler
+ warnings found with gcc-4.3: - the output of flex includes a
+ static function called 'input' that is not used, so for the
+ moment we'll stop having the compiler tell us about unused
+ variables in the flex source files (a better fix would be to
+ improve our flex post-processing to remove the unused function) -
+ main/stdtime/localtime.c makes assumptions about signed integer
+ overflow, and gcc-4.3's improved optimizer tries to take
+ advantage of handling potential overflow conditions at compile
+ time; for now, suppress these optimizations until we can fiure
+ out if the code needs improvement - main/udptl.c has some
+ references to uninitialized variables; in one case there was no
+ bug, but in the other it was certainly possibly for unexpected
+ behavior to occur - main/editline/readline.c had an unused
+ variable
+
+2008-03-11 00:59 +0000 [r107290] Terry Wilson <twilson@digium.com>
+
+ * channels/chan_sip.c: If we fail to alloc a channel, we should
+ re-lock the pvt structure before returning.
+
+2008-03-10 21:32 +0000 [r107230] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: Use non-global storage for eswitch
+
+2008-03-10 20:27 +0000 [r107173] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Make sure to reenable echo can after a
+ "failed" (canceled, etc) three-way call. (closes issue #11335)
+ Reported by: rebuild
+
+2008-03-10 20:17 +0000 [r107099-107161] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Fix another bug specifically related to asynchronous
+ call origination. Once the PBX is started on the channel using
+ ast_pbx_start(), then the ownership of the channel has been
+ passed on to another thread. We can no longer access it in this
+ code. If the channel gets hung up very quickly, it is possible
+ that we could access a channel that has been free'd. (inspired by
+ BE-386)
+
+ * main/pbx.c: Fix some bugs related to originating calls. If the
+ code failed to start a PBX on the channel (such as if you set a
+ call limit based on the system's load average), then there were
+ cases where a channel that has already been free'd using
+ ast_hangup() got accessed. This caused weird memory corruption
+ and crashes to occur. (fixes issue BE-386) (much debugging credit
+ goes to twilson, final patch written by me)
+
+ * main/channel.c: Resolve a compiler warning.
+
+ * main/channel.c: Fix a race condition where the generator can go
+ away (closes issue #12175, reported by edantie, patched by me)
+
+2008-03-10 14:33 +0000 [r107016] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_dial.c, main/cdr.c, include/asterisk/cdr.h: Move where
+ unanswered CDRs are dropped to the CDR core, not everything uses
+ app_dial. (closes issue #11516) Reported by: ys Patches:
+ branch_1.4_cdr.diff uploaded by ys (license 281) Tested by:
+ anest, jcapp, dartvader
+
+2008-03-08 15:59 +0000 [r106945] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: don't generate D-Channel "up" and "down"
+ messages unless the channel state is actually changing; also,
+ generate the "up" message when an implicit "up" occurs due to
+ reception of a normal event when we thought the channel was
+ "down"
+
+2008-03-07 22:51 +0000 [r106895] Russell Bryant <russell@digium.com>
+
+ * apps/app_meetme.c: Only start the SLA thread if SLA has actually
+ been configured.
+
+2008-03-07 22:14 +0000 [r106842] Jason Parker <jparker@digium.com>
+
+ * main/editline/Makefile.in: Fix hardcoded grep in editline, were
+ GNU grep is required. (closes issue #12124) Reported by: dmartin
+
+2008-03-07 19:32 +0000 [r106788] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: Ignore source update control frame. (closes issue
+ #12168) Reported by: plack
+
+2008-03-07 17:16 +0000 [r106704] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/sched.h: Change a warning message to a debug
+ message. This is happening quite frequently, and it is not worth
+ spamming users with these messages unless we are pretty confident
+ that it should never happen. As it stands today, it _will_ and
+ _does_ happen and until that gets cleaned up a reasonable amount
+ on the development side, let's not spam the logs of everyone
+ else. (closes issue #12154)
+
+2008-03-07 16:22 +0000 [r106552-106635] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Warn the user when a temporary greeting
+ exists (Closes issue #11409)
+
+ * main/rtp.c: Properly initialize rtp->schedid (Closes issue
+ #12154)
+
+ * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c,
+ apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c,
+ funcs/func_enum.c, channels/chan_misdn.c, main/frame.c,
+ main/manager.c: Safely use the strncat() function. (closes issue
+ #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt
+ uploaded by Corydon76 (license 14)
+
+2008-03-06 22:10 +0000 [r106437] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: Quell an annoying message that is likely to print
+ every single time that ast_pbx_outgoing_app is called. The reason
+ is that __ast_request_and_dial allocates the cdr for the channel,
+ so it should be expected that the channel will have a cdr on it.
+ Thanks to joetester on IRC for pointing this out
+
+2008-03-06 04:40 +0000 [r106328] Tilghman Lesher <tlesher@digium.com>
+
+ * sounds/Makefile: Upgrade to the next release of sounds
+
+2008-03-05 22:37 +0000 [r106237] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix a potential deadlock and a few
+ different potential crashes. (closes issue #12145, reported by
+ thiagarcia, patched by me)
+
+2008-03-05 22:32 +0000 [r106235] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_oss.c, main/rtp.c, channels/chan_mgcp.c,
+ apps/app_dial.c, main/channel.c, channels/chan_phone.c,
+ main/dial.c, channels/chan_zap.c, channels/chan_sip.c,
+ channels/chan_skinny.c, channels/chan_h323.c, main/file.c,
+ channels/chan_alsa.c, apps/app_followme.c,
+ include/asterisk/frame.h: Add a control frame to indicate the
+ source of media has changed. Depending on the underlying
+ technology it may need to change some things. (closes issue
+ #12148) Reported by: jcomellas
+
+2008-03-05 21:12 +0000 [r106178] Michiel van Baak <michiel@vanbaak.info>
+
+ * doc/realtime.txt: document var_metric so no bugreports will come
+ in when it's actually a configuration issue. (issue #12151)
+ Reported and patched by: caio1982 1.4 patch by me
+
+2008-03-05 15:32 +0000 [r106038] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: when a PRI call must be moved to a different
+ B channel at the request of the other endpoint, ensure that any
+ DSP active on the original channel is moved to the new one
+ (closes issue #11917) Reported by: mavetju Tested by: mavetju
+
+2008-03-05 15:17 +0000 [r106015] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, include/asterisk/sched.h: Correctly
+ initialize retransid in SIP, and ensure that the warning when
+ failing to delete a schedule entry can actually hit the log.
+ (closes issue #12140) Reported by: slavon Patches: sch2.patch
+ uploaded by slavon (license 288) (Patch slightly modified by me)
+
+2008-03-05 01:52 +0000 [r105932] Russell Bryant <russell@digium.com>
+
+ * main/rtp.c, main/translate.c, include/asterisk/frame.h: Fix a bug
+ that I just noticed in the RTP code. The calculation for setting
+ the len field in an ast_frame of audio was wrong when G.722 is in
+ use. The len field represents the number of ms of audio that the
+ frame contains. It would have set the value to be twice what it
+ should be.
+
+2008-03-04 18:10 +0000 [r105674-105676] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: In addition to setting the marker bit let's change
+ our ssrc so they know for sure it is a different source.
+
+ * main/rtp.c, channels/chan_sip.c, include/asterisk/rtp.h: When a
+ new source of audio comes in (such as music on hold) make sure
+ the marker bit gets set. (closes issue #10355) Reported by:
+ wdecarne Patches: 10355.diff uploaded by file (license 11)
+ (closes issue #11491) Reported by: kanderson
+
+2008-03-04 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.19-rc1 released.
+
+2008-03-04 04:31 +0000 [r105591] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Backport a minor bug fix from trunk that I found
+ while doing random code cleanup. Properly break out of the loop
+ when a context isn't found when verify that includes are valid.
+
+2008-03-03 18:06 +0000 [r105572] Jason Parker <jparker@digium.com>
+
+ * res/snmp/agent.c: Fix type for astNumChannels. (closes issue
+ #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg
+ (license 192)
+
+2008-03-03 17:16 +0000 [r105563-105570] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c: In the case of an ast_channel allocation
+ failure, take the local_pvt out of the pvt list before destroying
+ it.
+
+ * channels/chan_local.c: Fix a potential memory leak of the
+ local_pvt struct when ast_channel allocation fails. Also, in
+ passing, centralize the code necessary to destroy a local_pvt.
+
+ * main/autoservice.c: Update the copyright information for
+ autoservice. Most of the code in this file now is stuff that I
+ have written recently ...
+
+ * main/asterisk.c, main/channel.c, include/asterisk.h,
+ main/autoservice.c: Merge in some changes from
+ team/russell/autoservice-nochans-1.4 These changes fix up some
+ dubious code that I came across while auditing what happens in
+ the autoservice thread when there are no channels currently in
+ autoservice. 1) Change it so that autoservice thread doesn't keep
+ looping around calling ast_waitfor_n() on 0 channels twice a
+ second. Instead, use a thread condition so that the thread
+ properly goes to sleep and does not wake up until a channel is
+ put into autoservice. This actually fixes an interesting bug, as
+ well. If the autoservice thread is already running (almost always
+ is the case), then when the thread goes from having 0 channels to
+ have 1 channel to autoservice, that channel would have to wait
+ for up to 1/2 of a second to have the first frame read from it.
+ 2) Fix up the code in ast_waitfor_nandfds() for when it gets
+ called with no channels and no fds to poll() on, such as was the
+ case with the previous code for the autoservice thread. In this
+ case, the code would call alloca(0), and pass the result as the
+ first argument to poll(). In this case, the 2nd argument to
+ poll() specified that there were no fds, so this invalid pointer
+ shouldn't actually get dereferenced, but, this code makes it
+ explicit and ensures the pointers are NULL unless we have valid
+ data to put there. (related to issue #12116)
+
+2008-03-03 15:28 +0000 [r105557-105560] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: It is possible for no audio to pass between the
+ current digit and next digit so expand logic that clears
+ emulation to AST_FRAME_NULL. (closes issue #11911) Reported by:
+ edgreenberg Patches: v1-11911.patch uploaded by dimas (license
+ 88) Tested by: tbsky
+
+ * channels/chan_sip.c: Add a comment to describe some logic.
+ (closes issue #12120) Reported by: flefoll Patches:
+ chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license
+ 244)
+
+2008-02-29 23:34 +0000 [r105409] Russell Bryant <russell@digium.com>
+
+ * main/autoservice.c: Fix a major bug in autoservice. There was a
+ race condition in the handling of the list of channels in
+ autoservice. The problem was that it was possible for a channel
+ to get removed from autoservice and destroyed, while the
+ autoservice thread was still messing with the channel. This led
+ to memory corruption, and caused crashes. This explains multiple
+ backtraces I have seen that have references to autoservice, but
+ do to the nature of the issue (memory corruption), could cause
+ crashes in a number of areas. (fixes the crash in BE-386) (closes
+ issue #11694) (closes issue #11940) The following issues could be
+ related. If you are the reporter of one of these, please update
+ to include this fix and try again. (potentially fixes issue
+ #11189) (potentially fixes issue #12107) (potentially fixes issue
+ #11573) (potentially fixes issue #12008) (potentially fixes issue
+ #11189) (potentially fixes issue #11993) (potentially fixes issue
+ #11791)
+
+2008-02-29 14:47 +0000 [r105326] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Fix a potential memory leak
+
+2008-02-29 14:34 +0000 [r105296] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: If the message file does not exist, just
+ return harmlessly, instead of crashing. (Closes issue #12108)
+
+2008-02-29 13:48 +0000 [r105261] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_voicemail.c: Bump up the size of the uniqueid variable.
+ (closes issue #12107) Reported by: asgaroth
+
+2008-02-29 13:05 +0000 [r105209] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Automatically create new buddy upon reception
+ of a presence stanza of type subscribed. (closes issue #12066)
+ Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by
+ phsultan (license 73) trunk-12066-1.diff uploaded by phsultan
+ (license 73) Tested by: ffadaie, phsultan
+
+2008-02-28 22:23 +0000 [r105116] Russell Bryant <russell@digium.com>
+
+ * main/utils.c, include/asterisk/lock.h: Fix a bug in the lock
+ tracking code that was discovered by mmichelson. The issue is
+ that if the lock history array was full, then the functions to
+ mark a lock as acquired or not would adjust the stats for
+ whatever lock is at the end of the array, which may not be
+ itself. So, do a sanity check to make sure that we're updating
+ lock info for the proper lock. (This explains the bizarre stats
+ on lock #63 in BE-396, thanks Mark!)
+
+2008-02-28 21:56 +0000 [r105113] Tilghman Lesher <tlesher@digium.com>
+
+ * contrib/init.d/rc.debian.asterisk: Update init script for LSB
+ compat (closes issue #9843) Reported by: ibc Patches:
+ rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by:
+ paravoid
+
+2008-02-28 20:11 +0000 [r105059] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: When using autofill, members who are in use
+ should be counted towards the number of available members to call
+ if ringinuse is set to yes. Thanks to jmls who brought this issue
+ up on IRC
+
+2008-02-28 19:20 +0000 [r104920-105005] Jason Parker <jparker@digium.com>
+
+ * main/cdr.c, main/pbx.c: Make pbx_exec pass an empty string into
+ applications, if we get NULL. This protects against possible
+ segfaults in applications that may try to use data before
+ checking length (ast_strdupa'ing it, for example) (closes issue
+ #12100) Reported by: foxfire Patches: 12100-nullappargs.diff
+ uploaded by qwell (license 4)
+
+ * channels/chan_skinny.c: According to a video at www.cisco.com,
+ the 7921G supports 6 line appearances.
+
+2008-02-28 00:05 +0000 [r104868] Tilghman Lesher <tlesher@digium.com>
+
+ * main/Makefile, build_tools/strip_nonapi: Compatibility fix for
+ PPC64 (closes issue #12081) Reported by: jcollie Patches:
+ asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412)
+ Tested by: jcollie, Corydon76
+
+2008-02-27 21:49 +0000 [r104841] Mark Michelson <mmichelson@digium.com>
+
+ * main/dial.c: Two fixes: 1. Make the list of ast_dial_channels a
+ lockable list. This is because in some cases, the ast_dial may
+ exist in multiple threads due to asynchronous execution of its
+ application, and I found some cases where race conditions could
+ exist. 2. Check in ast_dial_join to be sure that the channel
+ still exists before attempting to lock it, since it could have
+ gotten hung up but the is_running_app flag on the
+ ast_dial_channel may not have been cleared yet. (closes issue
+ #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by
+ putnopvut (license 60) Tested by: jvandal
+
+2008-02-27 20:56 +0000 [r104787] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_chanspy.c: Don't loop around infinitely trying to spy on
+ our own channel, and don't forget to free/detach the datastore
+ upon hangup of the spy.
+
+2008-02-27 20:36 +0000 [r104783] Mark Michelson <mmichelson@digium.com>
+
+ * main/file.c: Bump a couple of more buffers up by 2 so that
+ annoying warnings aren't generated like crazy on every
+ fileexists_core call.
+
+2008-02-27 18:15 +0000 [r104704] Tilghman Lesher <tlesher@digium.com>
+
+ * main/manager.c: Ensure the session ID can't be 0.
+
+2008-02-27 17:41 +0000 [r104665] Joshua Colp <jcolp@digium.com>
+
+ * main/file.c: Bump up the buffer by 2.
+
+2008-02-27 17:33 +0000 [r104625] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: Fix a problem in ChanSpy where it could get
+ stuck in an infinite loop without being able to detect that the
+ calling channel hung up. (closes issue #12076, reported by junky,
+ patched by me)
+
+2008-02-27 17:26 +0000 [r104598] Jason Parker <jparker@digium.com>
+
+ * res/res_features.c: Inherit language from the transfering channel
+ on a blind transfer. (closes issue #11682) Reported by: caio1982
+ Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982
+ (license 22) Tested by: caio1982, victoryure
+
+2008-02-27 17:07 +0000 [r104596] Joshua Colp <jcolp@digium.com>
+
+ * main/loader.c: Use the lock (which already existed, it just
+ wasn't used) on the updaters list to protect the contents instead
+ of the overall module list lock. (closes issue #12080) Reported
+ by: ChaseVenters
+
+2008-02-27 16:53 +0000 [r104593] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/file.c: fallback to standard English prompts properly when
+ using new prompt directory layout (closes issue #11831) Reported
+ by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license
+ 20) (modified by me to improve code and conform rest of function
+ to coding guidelines)
+
+2008-02-27 16:45 +0000 [r104591] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: When we receive a known alarm, make sure
+ that the unknown alarm flag is not still set to make sure that
+ when we come back out of alarm, it gets reported in the log and
+ manager interface (after discussion with tzafrir on the -dev
+ list)
+
+2008-02-27 15:52 +0000 [r104536] Joshua Colp <jcolp@digium.com>
+
+ * res/res_smdi.c: Only stop the MWI monitor thread if it was
+ actually started. (closes issue #12086) Reported by: francesco_r
+
+2008-02-27 01:15 +0000 [r104332-104334] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: Avoid some recursion in the cleanup code for
+ the chanspy datastore (closes issue #12076, reported by junky,
+ patched by me)
+
+ * channels/chan_zap.c: Zaptel 1.4 now exposes FXO battery state as
+ an alarm. However, Asterisk 1.4 does not know what to do with
+ these alarms. Only Asterisk 1.6 cares about it. So, if we get an
+ unknown alarm in chan_zap, don't generate confusing log messages
+ about it.
+
+2008-02-26 18:26 +0000 [r104132-104141] Jason Parker <jparker@digium.com>
+
+ * Makefile: Add badshell to .PHONY target (thanks Kevin)
+
+ * Makefile: Since all shells aren't as awesome as bash, we have to
+ fail if somebody tries to use a literal "~" in DESTDIR.
+
+ * sounds/Makefile: Revert previous abspath change. ...abspath is
+ new in GNU make 3.81. I feel so...defeated. Must find new fix!
+
+ * sounds/Makefile: Fix a very bizarre issue we were seeing with our
+ buildbot when using a DESTDIR that wasn't an absolute path (such
+ as DESTDIR=~/asterisk-1.4). Apparently what was happening, was
+ that some of the targets were being expanded to the full path, so
+ $@ ended up being /root/asterisk-1.4/[...]/ rather than
+ ~/asterisk-1.4/[...]/ It appears that this may be a new "feature"
+ in GNU make. (*cough*
+ http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)
+
+2008-02-26 00:25 +0000 [r104119] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/smdi.h, apps/app_voicemail.c,
+ channels/chan_zap.c, res/res_smdi.c, configs/smdi.conf.sample:
+ Merge changes from team/russell/smdi-1.4 This commit brings in a
+ significant set of changes to the SMDI support in Asterisk. There
+ were a number of bugs in the current implementation, most notably
+ being that it was very likely on busy systems to pop off the
+ wrong message from the SMDI message queue. So, this set of
+ changes fixes the issues discovered as well as introducing some
+ new ways to use the SMDI support which are required to avoid the
+ bugs with grabbing the wrong message off of the queue. This code
+ introduces a new interface to SMDI, with two dialplan functions.
+ First, you get an SMDI message in the dialplan using
+ SMDI_MSG_RETRIEVE() and then you access details in the message
+ using the SMDI_MSG() function. A side benefit of this is that it
+ now supports more than just chan_zap. For example, with this
+ implementation, you can have some FXO lines being terminated on a
+ SIP gateway, but the SMDI link in Asterisk. Another issue with
+ the current implementation is that it is quite common that the
+ station ID that comes in on the SMDI link is not necessarily the
+ same as the Asterisk voicemail box. There are now additional
+ directives in the smdi.conf configuration file which let you map
+ SMDI station IDs to Asterisk voicemail boxes. Yet another issue
+ with the current SMDI support was related to MWI reporting over
+ the SMDI link. The current code could only report a MWI change
+ when the change was made by someone calling into voicemail. If
+ the change was made by some other entity (such as with IMAP
+ storage, or with a web interface of some kind), then the MWI
+ change would never be sent. The SMDI module can now poll for MWI
+ changes if configured to do so. This work was inspired by and
+ primarily done for the University of Pennsylvania. (also related
+ to issue #9260)
+
+2008-02-26 00:03 +0000 [r104111] Jason Parker <jparker@digium.com>
+
+ * channels/chan_h323.c: IPTOS_MINCOST is not defined on Solaris.
+ (closes issue #12050) Reported by: asgaroth Patches: 12050.patch
+ uploaded by putnopvut (license 60)
+
+2008-02-25 23:42 +0000 [r104102-104106] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: This patch fixes some pretty significant
+ problems with how app_chanspy handles pointers to channels that
+ are being spied upon. It was very likely that a crash would occur
+ if the channel being spied upon hung up. This was because the
+ current ast_channel handling _requires_ that the object is locked
+ or else it could disappear at any time (except in the owning
+ channel thread). So, this patch uses some channel datastore magic
+ on the spied upon channel to be able to detect if and when the
+ channel goes away. (closes issue #11877) (patch written by me,
+ but thanks to kpfleming for the idea, and to file for review)
+
+ * main/utils.c: Improve the lock tracking code a bit so that a
+ bunch of old locks that threads failed to lock don't sit around
+ in the history. When a lock is first locked, this checks to see
+ if the last lock in the list was one that was failed to be
+ locked. If it is, then that was a lock that we're no longer
+ sitting in a trylock loop trying to lock, so just remove it.
+ (inspired by issue #11712)
+
+2008-02-25 21:37 +0000 [r104095] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Make it so a users.conf user creates both a
+ SIP peer and a SIP user. The user will be used for inbound
+ authentication for the device, and peer will be used for placing
+ calls to the device. (closes issue #9044) Reported by: queuetue
+ Patches: sip-gui-friend.diff uploaded by qwell (license 4)
+
+2008-02-25 21:31 +0000 [r104094] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: If the destination folder is full, don't
+ delete a message when exiting. (closes issue #12065) Reported by:
+ selsky Patch by: (myself)
+
+2008-02-25 20:49 +0000 [r104092] Jason Parker <jparker@digium.com>
+
+ * main/config.c: Allow the use of #include and #exec in situations
+ where the max include depth was only 1. Specifically, this fixes
+ using #include and #exec in extconfig.conf. This was basically
+ caused because the config file itself raises the include level to
+ 1. I opted not to raise the include limit, because recursion here
+ could cause very bizarre behavior. Pointed out, and tested by
+ jmls (closes issue #12064)
+
+2008-02-25 18:38 +0000 [r104086] Russell Bryant <russell@digium.com>
+
+ * channels/chan_agent.c: Ensure that the channel doesn't disappear
+ in agent_logoff(). If it does, it could cause a crash. (fixes the
+ crash reported in BE-396)
+
+2008-02-25 16:16 +0000 [r104082-104084] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: If a resubscription comes in for a dialog we
+ no longer know about tell the remote side that the dialog does
+ not exist so they subscribe again using a new dialog. (closes
+ issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff
+ uploaded by file (license 11)
+
+ * channels/chan_sip.c: Due to recent changes tag will no longer be
+ NULL if not present so we have to use ast_strlen_zero to see if
+ it's actually blank. (closes issue #12061) Reported by: flefoll
+ Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by
+ flefoll (license 244)
+
+2008-02-22 22:45 +0000 [r104037] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Backwards debug message. (closes issue
+ #12052) Reported by: flefoll Patches:
+ chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license
+ 244)
+
+2008-02-21 21:05 +0000 [r104026-104027] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c: And as a followup to revision 104026,
+ completely remove event-related calls from a section of code
+ where we know there was no event to handle or get.
+
+ * channels/chan_zap.c: Remove an incorrect debug message. It
+ reported that it had received a specific event and tried to
+ report which event was received. What actually was happening was
+ that it was reporting the number of bytes returned from a call to
+ read(). Thanks to Jared Smith for bringing the issue up on IRC
+
+2008-02-21 14:33 +0000 [r104015] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/manager.c: reduce the likelihood that HTTP Manager session
+ ids will consist of primarily '1' bits
+
+2008-02-20 22:32 +0000 [r103956] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Clear up confusion when viewing the
+ QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the
+ user's perspective, the queue does exist, we shouldn't tell them
+ we couldn't find the queue. Instead since it is a dead queue,
+ report a 0 waiting count This issue was brought up on IRC by jmls
+
+2008-02-20 22:06 +0000 [r103953] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_zap.c: Don't wait for additional digits when
+ overlap dialing is enabled if the setup message contains the
+ sending_complete information element. (closes issue #11785)
+ Reported by: klaus3000 Patches:
+ sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by
+ klaus3000 (license 65)
+
+2008-02-20 21:40 +0000 [r103904] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: Fix a crash if the channel becomes NULL
+ while attempting to lock it. (closes issue #12039) Reported by:
+ danpwi
+
+2008-02-20 17:53 +0000 [r103845] Tilghman Lesher <tlesher@digium.com>
+
+ * main/stdtime/localtime.c: Compat fix for Solaris (closes issue
+ #12022) Reported by: asgaroth Patches:
+ 20080219__bug12022.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: asgaroth
+
+2008-02-19 20:28 +0000 [r103823] Joshua Colp <jcolp@digium.com>
+
+ * channels/h323/ast_h323.cxx: Send CallerID Name in setup message.
+ (closes issue #11241) Reported by: tusar Patches:
+ h323id_as_callerid_name.patch uploaded by tusar (license 344)
+
+2008-02-19 20:02 +0000 [r103821] Russell Bryant <russell@digium.com>
+
+ * channels/chan_local.c: Account for the fact that the "other"
+ channel can disappear while the local pvt is not locked. (fixes a
+ problem introduced in rev 100581) (closes issue #12012) Reported
+ by: stevedavies Patch by me
+
+2008-02-19 17:31 +0000 [r103807-103812] Joshua Colp <jcolp@digium.com>
+
+ * configure, configure.ac: Don't look for launchd when cross
+ compiling. (closes issue #12029) Reported by: ovi
+
+ * channels/chan_sip.c: Fix building of chan_sip.
+
+2008-02-19 10:27 +0000 [r103806] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Make sure we send error replies correctly by
+ checking the via header.
+
+2008-02-18 23:56 +0000 [r103801] Joshua Colp <jcolp@digium.com>
+
+ * main/channel.c: Ensure that emulated DTMFs do not get interrupted
+ by another begin frame. (closes issue #11740) Reported by: gserra
+ Patches: v1-11740.patch uploaded by dimas (license 88) (closes
+ issue #11955) Reported by: tsearle (closes issue #10530) Reported
+ by: xmarksthespot
+
+2008-02-18 22:28 +0000 [r103790-103795] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Fix previous commit so that we actually
+ disable echocanbridged if echocancel is off.
+
+ * channels/chan_zap.c: Correct a message when echocancelwhenbridged
+ is on, but echocancel is not. Issue #12019
+
+2008-02-18 20:52 +0000 [r103786] Mark Michelson <mmichelson@digium.com>
+
+ * main/app.c: There was an invalid assumption when calculating the
+ duration of a file that the filestream in question was created
+ properly. Unfortunately this led to a segfault in the situation
+ where an unknown format was specified in voicemail.conf and a
+ voicemail was recorded. Now, we first check to be sure that the
+ stream was written correctly or else assume a zero duration.
+ (closes issue #12021) Reported by: jakep Tested by: putnopvut
+
+2008-02-18 17:31 +0000 [r103780] Tilghman Lesher <tlesher@digium.com>
+
+ * main/rtp.c, channels/chan_sip.c: When a SIP channel is being
+ auto-destroyed, it's possible for it to still be in bridge code.
+ When that happens, we crash. Delay the RTP destruction until the
+ bridge is ended. (closes issue #11960) Reported by: norman
+ Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76
+ (license 14) Tested by: norman
+
+2008-02-18 16:37 +0000 [r103770] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_zap.c: Fix a linked list corruption that under the
+ right circumstances could lead to a looped list, meaning it will
+ traverse forever. (closes issue #11818) Reported by: michael-fig
+ Patches: 11818.patch uploaded by putnopvut (license 60) Tested
+ by: michael-fig
+
+2008-02-18 16:11 +0000 [r103763-103768] Joshua Colp <jcolp@digium.com>
+
+ * main/asterisk.c: Backport fix from issue #9325. (closes issue
+ #11980) Reported by: rbrunka
+
+ * channels/chan_sip.c: Don't care if the extension given doesn't
+ exist for subscription based MWI.
+
+2008-02-15 23:31 +0000 [r103726-103741] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Fix a crash in chan_iax2 due to a race
+ condition (closes issue #11780) Reported by: guillecabeza
+ Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license
+ 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license
+ 380)
+
+ * main/loader.c: In the case that you try to directly reload a
+ module has returned AST_MODULE_LOAD_DECLINE, log a message
+ indicating that the module is not fully initialized and must be
+ initialized using "module load".
+
+ * main/loader.c: Don't attempt to execute the reload callback for a
+ module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash
+ that was reported against chan_console in trunk. (closes issue
+ #11953, reported by junky, fixed by me)
+
+2008-02-15 17:26 +0000 [r103688-103722] Mark Michelson <mmichelson@digium.com>
+
+ * doc/imapstorage.txt, configure, configure.ac: Final round of
+ changes for configure script logic for IMAP Now if a directory is
+ specified, then we will search that directory for a source
+ installation of the IMAP toolkit. If none is found, then we will
+ use that directory as the basis for detecting a package
+ installation of the IMAP c-client. If that check fails, then
+ configure will fail.
+
+ * configure, configure.ac: Fix a bit of wrong logic in the
+ configure script that caused problems when trying to configure
+ without IMAP. Patch suggestion from phsultan, but I modified it
+ slightly. (closes issue #12003) Reported by: pj Tested by:
+ putnopvut
+
+ * doc/imapstorage.txt, configure, configure.ac: I apparently
+ misunderstood one of the requirements of this configure change.
+ Now, if a source directory is specified with the --with-imap
+ option, and a valid source installation is not detected there,
+ then configure will fail and will not check for a package
+ installation.
+
+ * doc/imapstorage.txt: Make a small clarification in the
+ documentation
+
+ * doc/imapstorage.txt: Update documentation regarding configuration
+ of IMAP
+
+ * apps/app_voicemail.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Change to the
+ configure logic regarding IMAP. Prior to this commit, if you
+ wished to configure Asterisk with IMAP support, you would use the
+ --with-imap configure switch in one of the following two ways:
+ --with-imap=/some/directory would look in the directory specified
+ for a UW IMAP source installation --with-imap would assume that
+ you had imap-2004g installed in .. relative to the Asterisk
+ source With this set of changes the two above options still work
+ the same, but there are two new behaviors, too.
+ --with-imap=system will assume that you have -libc-client.so
+ where you store your shared objects and will attempt to find
+ c-client headers in your include path either in the imap or
+ c-client directory. If either of the two original methods of
+ specifying the imap option should fail, then the check for
+ --with-imap =system will be performed in addition. It is only
+ after this "system" check that failure can happen.
+
+ * apps/app_voicemail.c: Fix build for non-IMAP builds
+
+ * apps/app_voicemail.c: Fix the new message count if delete=yes
+ when using IMAP storage. (closes issue #11406) Reported by:
+ jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license
+ 50) Tested by: jaroth
+
+2008-02-14 19:51 +0000 [r103683-103684] Jason Parker <jparker@digium.com>
+
+ * funcs/func_cdr.c: swap location for this..
+
+ * funcs/func_cdr.c: Document the 'l' option to the CDR() function.
+ (Thanks voipgate for pointing out the option, and Leif for
+ providing text for it.) Closes issue #11695.
+
+2008-02-13 06:25 +0000 [r103556-103607] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_agent.c: We aren't talking to ourselves; we're
+ talking to someone else. (closes issue #11771) Reported by:
+ msetim Patches: ami_agent_talkingto-1.4.diff uploaded by caio1982
+ (license 22) Tested by: caio1982, msetim
+
+ * apps/app_voicemail.c: Refuse to load app_voicemail if res_adsi is
+ not loaded (which is a symbol dependency) (closes issue #11760)
+ Reported by: non-poster Patches: 20080114__bug11760.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: Corydon76,
+ non-poster, jamesgolovich
+
+2008-02-12 22:24 +0000 [r103503-103504] Jason Parker <jparker@digium.com>
+
+ * main/asterisk.c: revert accidental change from last commit. oops
+
+ * contrib/scripts/safe_asterisk, main/asterisk.c: Remove condition
+ that was impossible.
+
+2008-02-12 15:09 +0000 [r103324-103385] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Even if no CallerID name or number has been
+ provided by the remote party still use the configured sip.conf
+ ones. (closes issue #11977) Reported by: pj
+
+ * apps/app_meetme.c: If entering a conference with the 'w' option
+ ensure that we can't listen or speak until the marked user
+ appears. (closes issue #11835) Reported by: alanmcmillan
+
+2008-02-11 17:05 +0000 [r103315] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/zapata.conf.sample: improve 2BCT documentation a bit
+ (thanks Jared)
+
+2008-02-09 06:23 +0000 [r103197] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Commit fix for being unable to send
+ voicemail from VoiceMailMain Reported by: William F Acker (via
+ the -users mailing list) Patch by: Corydon76 (license 14)
+
+2008-02-08 18:48 +0000 [r103070-103120] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Prevent a potential three-thread deadlock. Also
+ added a comment block to explicitly state the locking order
+ necessary inside app_queue. (closes issue #11862) Reported by:
+ flujan Patches: 11862.patch uploaded by putnopvut (license 60)
+ Tested by: flujan
+
+ * channels/chan_iax2.c: Yield the thread and return -1 if the ioctl
+ fails for Zaptel timing device. (closes issue #11891) Reported
+ by: tzafrir
+
+2008-02-08 15:08 +0000 [r102968] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_iax2.c: Make sure the presence of dbsecret is
+ factored into user scoring. (closes issue #11952) Reported by:
+ bbhoss
+
+2008-02-07 19:53 +0000 [r102858] Jason Parker <jparker@digium.com>
+
+ * res/res_features.c: Specify which digit string was matched in
+ debug message. (closes issue #11949) Reported by: dimas Patches:
+ v1-feature-debug.patch uploaded by dimas (license 88)
+
+2008-02-07 16:41 +0000 [r102807] Kevin P. Fleming <kpfleming@digium.com>
+
+ * configs/zapata.conf.sample: document usage of 'transfer'
+ configuration option for ISDN PRI switch-side transfers
+
+2008-02-06 17:59 +0000 [r102653-102725] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Only consider a T.38-only INVITE compatible
+ if we have both a joint capability between us and them and if
+ they provided T.38.
+
+ * main/global_datastores.c: Add missing header file and
+ ASTERISK_FILE_VERSION usage. (closes issue #11936) Reported by:
+ snuffy
+
+2008-02-06 15:19 +0000 [r102651] Russell Bryant <russell@digium.com>
+
+ * configs/features.conf.sample: Clarify setting DYNAMIC_FEATURES so
+ that it gets inherited by outbound channels. (due to a discussion
+ between me and a user via email)
+
+2008-02-06 11:48 +0000 [r102627] Kevin P. Fleming <kpfleming@digium.com>
+
+ * pbx/Makefile, res/Makefile: ensure that all remaining
+ multi-object modules are built using their proper CFLAGS and
+ include directory paths
+
+2008-02-06 00:26 +0000 [r102576] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Move around some defines to unbreak ODBC
+ storage. (closes issue #11932) Reported by: snuffy
+
+2008-02-05 20:02 +0000 [r102453] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_mgcp.c: Clear the DTMF buffer on hangup. (closes
+ issue #11919) Reported by: eferro Patches:
+ mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337)
+ Tested by: eferro
+
+2008-02-05 19:52 +0000 [r102450] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: If a REGISTER attempt comes in that is a
+ retransmission of a previous REGISTER do not create a new nonce
+ value. (issue #BE-381)
+
+2008-02-05 17:15 +0000 [r102425] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/Makefile: ensure that components of chan_misdn.so are
+ built using any special build options that the configure script
+ generated (reported by Philipp Kempgen on asterisk-dev)
+
+2008-02-05 15:09 +0000 [r102378] Joshua Colp <jcolp@digium.com>
+
+ * res/res_clioriginate.c: Perform dialing asynchronously when using
+ the originate CLI command so the CLI does not appear to block.
+ (closes issue #11927) Reported by: bbhoss
+
+2008-02-04 21:06 +0000 [r102214-102323] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, utils/muted.c, configure,
+ include/asterisk/autoconfig.h.in, configure.ac: Cross-platform
+ fix: OS X now deprecates the use of the daemon(3) API. (closes
+ issue #11908) Reported by: oej Patches:
+ 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76
+
+ * funcs/func_strings.c: Missing braces. (closes issue #11912)
+ Reported by: dimas Patches: sprintf.patch uploaded by dimas
+ (license 88)
+
+2008-02-03 16:38 +0000 [r102090-102142] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Use the same CSEQ on CANCEL as on INVITE
+ (according to RFC 3261) (closes issue #9492) Reported by:
+ kryptolus Patches: bug9492.txt uploaded by oej (license 306)
+ Tested by: oej
+
+ * channels/chan_sip.c: Handle ACK and CANCEL in an invite
+ transaction - even if we get INFO transactions during the actual
+ call setup. (closes issue #10567) Reported by: jacksch Tested by:
+ oej Patch by: oej inspired by suggestions from neutrino88 in the
+ bug tracker
+
+2008-02-01 23:06 +0000 [r101989] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Change the SDP_SAMPLE_RATE macro. It turns
+ out that even though G.722 is 16 kHz, it is supposed to specified
+ as 8 kHz in the RTP, and RTP timestamps are supposed to be
+ calculated based on 8 kHz. (Apparently this is due to a bug in a
+ spec, but people follow it anyway, because it's the spec ...)
+
+2008-02-01 21:54 +0000 [r101894-101942] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: Fix the VM_DUR variable for forwarded
+ voicemail, and fixed several other bugs while I'm in the area.
+ (closes issue #11615) Reported by: jamessan Patches:
+ 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76, jamessan
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ acinclude.m4: Change detection of getifaddrs to use
+ AST_C_COMPILE_CHECK, backported from trunk (as suggested by
+ kpfleming)
+
+2008-02-01 17:41 +0000 [r101822] Jason Parker <jparker@digium.com>
+
+ * apps/app_authenticate.c: Remove a needless (and incorrect) call
+ to feof() after fgets(). This would have exited the loop early if
+ you had an authentication file with no newline at the end.
+
+2008-02-01 17:27 +0000 [r101818-101820] Russell Bryant <russell@digium.com>
+
+ * apps/app_authenticate.c: off by one error
+
+ * apps/app_authenticate.c: Don't overwrite the last character of a
+ line if it's not a newline. This would happen if the last line in
+ the file doesn't have a newline. (pointed out by Qwell)
+
+2008-02-01 15:55 +0000 [r101772] Tilghman Lesher <tlesher@digium.com>
+
+ * configure, include/asterisk/autoconfig.h.in, configure.ac,
+ main/acl.c: Compatibility fix for OpenWRT (reported by Brian
+ Capouch via the mailing list)
+
+2008-02-01 00:32 +0000 [r101693] Russell Bryant <russell@digium.com>
+
+ * channels/chan_iax2.c: Add some more sanity checking on IAX2 dial
+ strings for the case that no peer or hostname was provided, which
+ is the one part of the dial string that is absolutely required.
+ If it's not there, bail out. (closes issue #11897) Reported by
+ sokhapkin Patch by me
+
+2008-02-01 00:06 +0000 [r101649] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_amd.c: From bugtracker: "fix totalAnalysisTime to handle
+ periods of no channel activity" (closes issue #9256) Reported by:
+ cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt
+ uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81,
+ rjain
+
+2008-01-31 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.18 released.
+
+2008-01-31 23:10 +0000 [r101601] Russell Bryant <russell@digium.com>
+
+ * main/translate.c, main/file.c: Fix a couple of places where
+ ast_frfree() was not called on a frame that came from a
+ translator. This showed itself by g729 decoders not getting
+ released. Since the flag inside the translator frame never got
+ unset by freeing the frame to indicate it was no longer in use,
+ the translators never got destroyed, and thus the g729 licenses
+ were not released. (closes issue #11892) Reported by: xrg
+ Patches: 11892.diff uploaded by russell (license 2) Tested by:
+ xrg, russell
+
+2008-01-31 21:00 +0000 [r101531] Mark Michelson <mmichelson@digium.com>
+
+ * res/res_monitor.c: 1. Prevent the addition of an extra '/' to the
+ beginning of an absolute pathname. 2. If ast_monitor_change_fname
+ is called and the new filename is the same as the old, then exit
+ early and don't set the filename_changed field in the monitor
+ structure. Setting it in this case was causing ast_monitor_stop
+ to erroneously delete them. (closes issue #11741) Reported by:
+ garlew Tested by: putnopvut
+
+2008-01-31 19:52 +0000 [r101482] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c, channels/chan_iax2.c: Solaris compat fixes
+ for struct in_addr funkiness. Issue #11885, patch by snuffy.
+
+2008-01-31 19:30 +0000 [r101480] Steve Murphy <murf@digium.com>
+
+ * main/pbx.c: closes issue #11845; that's the one where there's a
+ 1004 byte cdr leak with every AMI Redirect to a zap channel
+
+2008-01-31 19:17 +0000 [r101413-101433] Russell Bryant <russell@digium.com>
+
+ * channels/chan_agent.c: Add more missing locking of the agents
+ list ...
+
+ * channels/chan_agent.c: Move the locking from find_agent() into
+ the agent dialplan function handler to ensure that the agent
+ doesn't disappear while we're looking at it.
+
+ * channels/chan_agent.c: Add missing locking to the find_agent()
+ function.
+
+2008-01-30 15:41 +0000 [r101222] Joshua Colp <jcolp@digium.com>
+
+ * main/slinfactory.c: Fix an issue where if a frame of higher
+ sample size preceeded a frame of lower sample size and
+ ast_slinfactory_read was called with a sample size of the
+ combined values or higher a crash would happen. (closes issue
+ #11878) Reported by: stuarth
+
+2008-01-30 15:34 +0000 [r101219] Jason Parker <jparker@digium.com>
+
+ * configs/extensions.conf.sample: Change default config to use
+ descending channel order of groups, rather than ascending. Fixes
+ a potential source of confusion in glare-type situations. Issue
+ 11875, reported by JimVanM.
+
+2008-01-30 15:23 +0000 [r101216] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fix a logic error with regards to autofill.
+ Prior to this change, it was possible for a caller to go out of
+ turn if autofill were enabled and callers ahead in the queue were
+ attempting to call a member. This change fixes this.
+
+2008-01-30 11:20 +0000 [r101152] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Stop musiconhold on attended transfer.
+ (closes issue #11872) Reported by: gareth Patches:
+ svn-101018.patch uploaded by gareth (license 208)
+
+2008-01-29 23:50 +0000 [r101080] Dwayne M. Hubbard <dhubbard@digium.com>
+
+ * build_tools/make_version: updated build_tools to handle the
+ autotag directory structure changes; changes related to BE-353.
+ Patch by The Russell and reviewed by The Me.
+
+2008-01-29 23:02 +0000 [r100973-101035] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Remove a memory leak from updating realtime
+ queues
+
+ * apps/app_queue.c: Fixing an erroneous return value returned when
+ attempting to pause or unpause a queue member fails. Fixes
+ BE-366, thanks to John Bigelow for writing the patch.
+
+2008-01-29 17:57 +0000 [r100934] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_mixmonitor.c: Don't forget to record the channel so we
+ know whether it is bridged or not later. (closes issue #11811)
+ Reported by: slavon
+
+2008-01-29 17:43 +0000 [r100932] Russell Bryant <russell@digium.com>
+
+ * main/Makefile: Fix the last couple of issues related to building
+ from a path that contains spaces. (closes issue #11834)
+
+2008-01-29 17:41 +0000 [r100930] Jason Parker <jparker@digium.com>
+
+ * channels/misdn_config.c: Initialize an array to 0s if config
+ option not specified. (closes issue #11860) Patches:
+ misdn_get_config.v1.diff uploaded by IgorG (license 20)
+
+2008-01-29 17:21 +0000 [r100882-100922] Russell Bryant <russell@digium.com>
+
+ * Makefile: Use GNU make magic instead of shell magic to escape
+ spaces in the working directory. (related to issue #11834)
+
+ * Makefile: Fix building Asterisk when the working path has spaces
+ in it. (closes issue #11834) Reported by: spendergrass Patched
+ by: me
+
+2008-01-29 16:10 +0000 [r100835] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Allow zap groups above 30 to work properly.
+ (closes issue #11590) Reported by: tbsky
+
+2008-01-29 10:36 +0000 [r100793] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: fixed potential segfault in misdn show
+ channels CLI command
+
+2008-01-29 08:26 +0000 [r100740] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: (closes issue #11736) Reported by: MVF
+ Patches: bug11736-2.diff uploaded by oej (license 306) Tested by:
+ oej, MVF, revolution (russellb: This was the showstopper for the
+ release.)
+
+2008-01-28 21:02 +0000 [r100675] Tilghman Lesher <tlesher@digium.com>
+
+ * main/pbx.c: WaitExten didn't handle AbsoluteTimeout properly
+ (went to 't' instead of 'T')
+
+2008-01-28 20:55 +0000 [r100673] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_vpb.cc, UPGRADE.txt: Undoing the deprecation of
+ chan_vpb. It is alive and well.
+
+2008-01-28 20:42 +0000 [r100672] Jason Parker <jparker@digium.com>
+
+ * apps/app_voicemail.c: When using ODBC_STORAGE, make sure we put
+ greeting files into the database like we do with the others.
+ Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded
+ by dimas (license 88)
+
+2008-01-28 18:34 +0000 [r100626-100629] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: For some reason, the use of this strdupa()
+ is leading to memory corruption on freebsd sparc64. This trivial
+ workaround fixes it. (closes issue #10300, closes issue #11857,
+ reported by mattias04 and Home-of-the-Brave)
+
+ * res/res_features.c: Fix a crash in ast_masq_park_call() (issue
+ #11342) Reported by: DEA Patches: res_features-park.txt uploaded
+ by DEA (license 3)
+
+2008-01-28 18:23 +0000 [r100624] Jason Parker <jparker@digium.com>
+
+ * channels/chan_zap.c: Correct a comment which made little/no
+ sense.
+
+2008-01-28 17:15 +0000 [r100581] Russell Bryant <russell@digium.com>
+
+ * main/channel.c, channels/chan_local.c,
+ include/asterisk/channel.h: Make some deadlock related fixes.
+ These bugs were discovered and reported internally at Digium by
+ Steve Pitts. - Fix up chan_local to ensure that the channel lock
+ is held before the local pvt lock. - Don't hold the channel lock
+ when executing the timing function, as it can cause a deadlock
+ when using chan_local. This actually changes the code back to be
+ how it was before the change for issue #10765. But, I added some
+ other locking that I think will prevent the problem reported
+ there, as well.
+
+2008-01-27 21:59 +0000 [r100465] Tilghman Lesher <tlesher@digium.com>
+
+ * main/rtp.c, channels/chan_mgcp.c, main/cdr.c,
+ channels/chan_misdn.c, main/dnsmgr.c, channels/chan_sip.c,
+ channels/chan_h323.c, include/asterisk/sched.h, main/file.c,
+ pbx/pbx_dundi.c, channels/chan_iax2.c: When deleting a task from
+ the scheduler, ignoring the return value could possibly cause
+ memory to be accessed after it is freed, which causes all sorts
+ of random memory corruption. Instead, if a deletion fails, wait a
+ bit and try again (noting that another thread could change our
+ taskid value). (closes issue #11386) Reported by: flujan Patches:
+ 20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
+ Tested by: Corydon76, flujan, stuarth`
+
+2008-01-25 22:32 +0000 [r100418] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_vpb.cc, UPGRADE.txt: Deprecating chan_vpb. It is
+ now preferred that users of Voicetronix products use chan_zap in
+ combination with their zaptel drivers.
+
+2008-01-25 21:24 +0000 [r100378] Jason Parker <jparker@digium.com>
+
+ * channels/chan_sip.c: This would have never been true, since we're
+ passing (sizeof(req.data) - 1) as the len to recvfrom().
+
+2008-01-24 21:57 +0000 [r100264] Kevin P. Fleming <kpfleming@digium.com>
+
+ * include/asterisk/app.h: make these macros not assume that the
+ only other field in the structure is 'argc'... this is true when
+ someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable
+ to define your own structure as long as it has the right fields
+
+2008-01-24 17:22 +0000 [r100164] Russell Bryant <russell@digium.com>
+
+ * main/asterisk.c: Update main Asterisk copyright info to 2008
+
+2008-01-24 16:41 +0000 [r100138] Jason Parker <jparker@digium.com>
+
+ * main/acl.c: Fix compilation on Solaris. (closes issue #11832)
+ Patches: bug-11832.diff uploaded by snuffy (license 35)
+
+2008-01-23 21:07 +0000 [r99977-99978] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Second attempt. Don't change invitestate
+ when receiving 18x messages in CANCEL state. (issue #11736)
+ Reported by: MVF Patch by oej.
+
+ * channels/chan_sip.c: Make sure we don't cancel destruction on
+ calls in CANCEL state, even if we get 183 while waiting for
+ answer on our CANCEL. (issue #11736) Reported by: MVF Patches:
+ bug11736.txt uploaded by oej (license 306) Tested by: MVF
+
+2008-01-23 20:25 +0000 [r99975] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_externalivr.c: Fixing a typo.
+
+2008-01-23 17:46 +0000 [r99923] Russell Bryant <russell@digium.com>
+
+ * apps/app_chanspy.c: ChanSpy issues a beep when it starts at the
+ beginning of a list of channels to potentially spy on. However,
+ if there were no matching channels, it would beep at you over and
+ over, which is pretty annoying. Now, it will only beep once in
+ the case that there are no channels to spy on, but it will still
+ beep again once it reaches the beginning of the channel list
+ again. (closes issue #11738, patched by me)
+
+2008-01-23 16:18 +0000 [r99878] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_sip.c: These flag tests were illogical. They were
+ testing sip_peer flags on a sip_pvt. Thanks to Russell for
+ helping to get this odd problem figured out.
+
+2008-01-23 04:31 +0000 [r99718-99777] Tilghman Lesher <tlesher@digium.com>
+
+ * apps/app_voicemail.c: When we reset the password via an external
+ command, we should also reset the password stored in the
+ in-memory list, too (otherwise it doesn't really take effect).
+ (closes issue #11809) Reported by: davetroy Patches:
+ fix_externpass.diff uploaded by davetroy (license 384)
+
+ * res/res_odbc.c: Oops, should have checked for a NULL obj, here,
+ too
+
+ * main/acl.c: Just confirmed that all current platforms need this
+ header file
+
+2008-01-22 20:56 +0000 [r99652] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Thanks to Russell's education I realize that
+ BUFSIZ has changed since I learned the C language over 20 years
+ ago... Resetting chan_sip to the size of BUFSIZ that I expected
+ in my old head to avoid to heavy memory allocations on some
+ systems.
+
+2008-01-22 20:34 +0000 [r99643] Tilghman Lesher <tlesher@digium.com>
+
+ * main/acl.c: Fix the defines for OS X (and Solaris, too)
+
+2008-01-22 17:41 +0000 [r99592-99594] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_local.c, res/res_features.c, channels/chan_agent.c,
+ apps/app_followme.c: Add more dependencies on chan_local and add
+ a note to the description of chan_local so that people don't
+ disable it in menuselect just to clean up.
+
+ * apps/app_dial.c: Add dependency on chan_local to app_dial. Dial
+ still runs without chan_local, but will be missing forwarding
+ functionality.
+
+2008-01-22 16:54 +0000 [r99540] Tilghman Lesher <tlesher@digium.com>
+
+ * main/acl.c: Ensure that we can get an address even when we don't
+ have a default route. (closes issue #9225) Reported by: junky
+ Patches: 20080122__bug9225.diff.txt uploaded by Corydon76
+ (license 14) Tested by: oej, loloski, sergee
+
+2008-01-22 15:08 +0000 [r99501] Olle Johansson <oej@edvina.net>
+
+ * channels/chan_sip.c: Cleaning up some documentation that led to
+ confusion in a bug report
+
+2008-01-21 23:55 +0000 [r99426] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: Fixing an issue wherein monitoring local
+ channels was not possible. During a channel masquerade, the
+ monitors on the two channels involved are swapped. In 99% of the
+ cases this results in the desired effect. However, if monitoring
+ a local channel, this caused the monitor which was on the local
+ channel to get moved onto a channel which is immediately hung up
+ after the masquerade has completed. By swapping the monitors
+ prior to the masquerade, we avoid the problem by tricking the
+ masquerade into placing the monitor back onto the channel where
+ we want it. During the investigation of the issue, the channel's
+ monitor was the only thing that was swapped in such a manner
+ which did not make sense to have done. All other variable
+ swapping made sense.
+
+2008-01-21 18:11 +0000 [r99341] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c, configs/res_odbc.conf.sample,
+ include/asterisk/res_odbc.h: Permit the user to specify number of
+ seconds that a connection may remain idle, which fixes a crash on
+ reconnect with the MyODBC driver. (closes issue #11798) Reported
+ by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt
+ uploaded by Corydon76 (license 14) Tested by: mvanbaak
+
+2008-01-21 16:01 +0000 [r99301] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Bump the buffer size for Via headers up to
+ 512. There are some exceptionally large Via headers out there.
+ (closes issue #11783) Reported by: ofirroval
+
+2008-01-19 10:05 +0000 [r99187] Russell Bryant <russell@digium.com>
+
+ * main/slinfactory.c: Fix a couple of memory leaks with frame
+ handling. Specifically, ast_frame_free() needed to be called on
+ the frame that came from the translator to signed linear.
+
+2008-01-18 22:57 +0000 [r99127] Joshua Colp <jcolp@digium.com>
+
+ * include/asterisk/channel.h: Remove the __ in front of the unused
+ variable. This causes some compilers to freak out.
+
+2008-01-18 21:37 +0000 [r99079-99081] Russell Bryant <russell@digium.com>
+
+ * include/asterisk/translate.h, main/frame.c: Revert adding the
+ packed attribute, as it really doesn't make sense why that would
+ do any good. Fix the real bug, which is to do the check to see if
+ the frame came from a translator at the beginning of
+ ast_frame_free(), instead of at the end. This ensures that it
+ always gets checked, even if none of the parts of the frame are
+ malloc'd, and also ensures that we aren't looking at free'd
+ memory in the case that it is a malloc'd frame. (closes issue
+ #11792, reported by explidous, patched by me)
+
+ * include/asterisk/translate.h: Since we're relying on the offset
+ between the frame and the beginning of the translator pvt struct,
+ set the packed attribute to make sure we get to the right place.
+ (potential fix for issue #11792)
+
+2008-01-18 17:13 +0000 [r99032] Terry Wilson <twilson@digium.com>
+
+ * res/res_features.c: This should at least temporarily fix a
+ problem where the 't' Dial option is incorrectly passed to the
+ transferee when built-in attended transfers are used. There is
+ still a problem with 'T', but better to fix some problems than no
+ problems while we work on it. (closes issue #7904) Reported by:
+ k-egg Patches: transfer-fix-b14-r97657.diff uploaded by sergee
+ (license 138) Tested by: sergee, otherwiseguy
+
+2008-01-17 23:42 +0000 [r99007-99014] Pari Nannapaneni <paripurnachand@digium.com>
+
+ * configs/cdr.conf.sample: doh! revert a revert of a revert
+ (changed by mistake in 99010)
+
+ * main/manager.c, configs/cdr.conf.sample: missed that one while
+ reverting
+
+ * main/manager.c: reverting 99001 - We need the Max-Age for
+ extending the life of cookie mansession_id
+
+2008-01-17 22:37 +0000 [r99004] Russell Bryant <russell@digium.com>
+
+ * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h:
+ Have IAX2 optimize the codec translation path just like chan_sip
+ does it. If the caller's codec is in our codec list, move it to
+ the top to avoid transcoding. (closes issue #10500) Reported by:
+ stevedavies Patches: iax-prefer-current-codec.patch uploaded by
+ stevedavies (license 184) iax-prefer-current-codec.1.4.patch
+ uploaded by stevedavies (license 184) Tested by: stevedavies, pj,
+ sheldonh
+
+2008-01-17 21:31 +0000 [r99001] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/manager.c: we should only send the Set-Cookie header to the
+ browser on the first response after creating a manager session,
+ not on every response (doing so causes the browser to clear any
+ local cookies it may have associated with the session)
+
+2008-01-17 16:19 +0000 [r98991] Jason Parker <jparker@digium.com>
+
+ * configs/zapata.conf.sample: Add a clarification about the
+ immediate= option of zapata.conf Issue 11784, patch by klaus3000.
+
+2008-01-16 22:36 +0000 [r98982] Russell Bryant <russell@digium.com>
+
+ * .cleancount, include/asterisk/channel.h: Add an unused pointer to
+ the ast_channel struct. This makes the ast_channel structure
+ retain the same size as it had in previous 1.4 releases. Also,
+ all of the offsets for members in the structure are still the
+ same (except for the two pointers that got replaced for the new
+ spy/whisper architecture.)
+
+2008-01-16 20:34 +0000 [r98966-98973] Joshua Colp <jcolp@digium.com>
+
+ * .cleancount: Bump up cleancount due to previous commit that
+ changed the channel structure.
+
+ * apps/app_chanspy.c, apps/app_mixmonitor.c, main/rtp.c,
+ main/channel.c, apps/app_meetme.c, include/asterisk/audiohook.h
+ (added), main/Makefile, include/asterisk/chanspy.h (removed),
+ include/asterisk/channel.h, main/audiohook.c (added): Replace
+ current spy architecture with backport of audiohooks. This should
+ take care of current known spy issues.
+
+ * channels/chan_iax2.c: Add missing NULLs at end of two
+ ast_load_realtimes. (closes issue #11769) Reported by: tequ
+ Patches: chaniax.patch uploaded by dimas (license 88)
+
+2008-01-16 17:20 +0000 [r98964] Mark Michelson <mmichelson@digium.com>
+
+ * channels/chan_local.c: Fix a deadlock in chan_local in
+ local_hangup. There was contention because the local_pvt was held
+ and it was attempting to lock a channel, which is the incorrect
+ locking order. (closes issue #11730) Reported by: UDI-Doug
+ Patches: 11730.patch uploaded by putnopvut (license 60) Tested
+ by: UDI-Doug
+
+2008-01-16 15:08 +0000 [r98951-98960] Joshua Colp <jcolp@digium.com>
+
+ * main/dial.c: Introduce a lock into the dialing API that protects
+ it when destroying the structure. (closes issue #11687) Reported
+ by: callguy Patches: 11687.diff uploaded by file (license 11)
+
+ * main/rtp.c: Add two more SDP names for ulaw and alaw. (closes
+ issue #11777) Reported by: tootai
+
+ * channels/chan_sip.c: Don't drop the old record route information
+ when dealing with packets related to a reinvite. (closes issue
+ #11545) Reported by: kebl0155 Patches: reinvite-patch.txt
+ uploaded by kebl0155 (license 356)
+
+ * build_tools/menuselect-deps.in, configure,
+ include/asterisk/autoconfig.h.in, codecs/codec_speex.c,
+ configure.ac, makeopts.in: Add autoconf logic for speexdsp. Later
+ versions use a separate library for some things so we need to use
+ it if present in codec_speex. (closes issue #11693) Reported by:
+ yzg
+
+2008-01-15 23:50 +0000 [r98943-98946] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Change a buffer in check_auth() to be a
+ thread local dynamically allocated buffer, instead of a massive
+ buffer on the stack. This fixes a crash reported by Qwell due to
+ running out of stack space when building with LOW_MEMORY defined.
+ On a very related note, the usage of BUFSIZ in various places in
+ chan_sip is arbitrary and careless. BUFSIZ is a system specific
+ define. On my machine, it is 8192, but by definition (according
+ to google) could be as small as 256. So, this buffer in
+ check_auth was 16 kB. We don't even support SIP messages larger
+ than 4 kB! Further usage of this define should be avoided, unless
+ it is used in the proper context.
+
+ * main/rtp.c, include/asterisk/translate.h, main/frame.c,
+ main/translate.c, main/abstract_jb.c, channels/chan_iax2.c,
+ codecs/codec_zap.c, include/asterisk/frame.h: Commit a fix for
+ some memory access errors pointed out by the valgrind2.txt output
+ on issue #11698. The issue here is that it is possible for an
+ instance of a translator to get destroyed while the frame
+ allocated as a part of the translator is still being processed.
+ Specifically, this is possible anywhere between a call to
+ ast_read() and ast_frame_free(), which is _a lot_ of places in
+ the code. The reason this happens is that the channel might get
+ masqueraded during this time. During a masquerade, existing
+ translation paths get destroyed. So, this patch fixes the issue
+ in an API and ABI compatible way. (This one is for you,
+ paravoid!) It changes an int in ast_frame to be used as flag
+ bits. The 1 bit is still used to indicate that the frame contains
+ timing information. Also, a second flag has been added to
+ indicate that the frame came from a translator. When a frame with
+ this flag gets released and has this flag, a function is called
+ in translate.c to let it know that this frame is doing being
+ processed. At this point, the flag gets cleared. Also, if the
+ translator was requested to be destroyed while its internal frame
+ still had this flag set, its destruction has been deffered until
+ it finds out that the frame is no longer being processed.
+ Admittedly, this feels like a hack. But, it does fix the issue,
+ and I was not able to think of a better solution ...
+
+2008-01-15 20:08 +0000 [r98894-98934] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Based on the boundary found move over the
+ correct amount. (closes issue #11750) Reported by: tasker
+
+ * channels/chan_sip.c: Accept "; boundary=" not just ";boundary="
+ in the multipart mixed content type. (closes issue #11750)
+ Reported by: tasker
+
+2008-01-14 20:59 +0000 [r98849] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Adding in appropriate unlocks for the locks
+ I added. Thanks to joetester on IRC for pointing this out.
+
+2008-01-14 17:38 +0000 [r98774] Russell Bryant <russell@digium.com>
+
+ * main/translate.c: Revert a change that introduces an unacceptable
+ performance hit and is causing memory leaks ... (from rev 97973)
+
+2008-01-14 16:35 +0000 [r98733-98737] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Fixing another compilation error. I'm a bit off
+ today :(
+
+ * apps/app_queue.c: Oops. Last commit had compilation error.
+
+ * apps/app_queue.c: Adding explicit defaults for missing options to
+ init_queue. This is necessary because if a user either removes or
+ comments one of these options and reloads their queues, the
+ option will not reset to its default, instead maintaining the
+ value from prior to the reload. Thanks to John Bigelow for
+ pointing this error out to me.
+
+2008-01-12 00:05 +0000 [r98467] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_odbc.c: Add a connection timeout attribute, as that was
+ what was intended with the login timeout, but ODBC divides it up
+ into 2 different timeouts. (Closes issue #11745)
+
+2008-01-11 22:46 +0000 [r98390] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_dundi.c: Fix up setting the EID on BSD based systems.
+ (closes issue #11646) Reported by: caio1982 Patches:
+ dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22)
+ dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested
+ by: caio1982, mvanbaak
+
+2008-01-11 21:28 +0000 [r98372] Pari Nannapaneni <paripurnachand@digium.com>
+
+ * main/http.c: Comment explaining how to force browser to always
+ read some html files from server.
+
+2008-01-11 19:51 +0000 [r98317-98325] Joshua Colp <jcolp@digium.com>
+
+ * main/rtp.c: If the incoming RTP stream changes codec force the
+ bridge to break if the other side does not support it. (closes
+ issue #11729) Reported by: tsearle Patches:
+ new_codec_patch_udiff.patch uploaded by tsearle (license 373)
+
+ * res/res_agi.c: If the channel is hungup during RECORD FILE send a
+ result code of -1 to be uniform with everything else. (closes
+ issue #11743) Reported by: davevg Patches: res_agi.diff uploaded
+ by davevg (license 209)
+
+2008-01-11 19:10 +0000 [r98315] Mark Michelson <mmichelson@digium.com>
+
+ * main/channel.c: Properly report the hangup cause as no answer
+ when someone does not answer (closes issue #10574, reported by
+ boch, patched by moy)
+
+2008-01-11 18:25 +0000 [r98266] Tilghman Lesher <tlesher@digium.com>
+
+ * codecs/gsm/Makefile: Add another exception (which doesn't work)
+ for -march optimization flag. Reported by: thomasmebes Patch by:
+ tilghman (Closes issue #11563)
+
+2008-01-11 18:25 +0000 [r98265] Russell Bryant <russell@digium.com>
+
+ * doc/security.txt, main/asterisk.c, configure,
+ include/asterisk/autoconfig.h.in, main/Makefile, configure.ac,
+ makeopts.in: Backport the ability to set the ToS bits on Linux
+ when not running as root. Normally, we would not backport
+ features into 1.4, but, I was convinced by the justification
+ supplied by the supplier of this patch. He pointed out that this
+ patch removes a requirement for running as root, thus reducing
+ the potential impacts of security issues. (closes issue #11742)
+ Reported by: paravoid Patches: libcap.diff uploaded by paravoid
+ (license 200)
+
+2008-01-11 17:22 +0000 [r98219] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_followme.c: Ensure the return value of ast_bridge_call
+ is passed back up as the application return value. This is needed
+ for transfers to function so the PBX core knows to continue
+ execution. (closes issue #10327) Reported by: kkiely
+
+2008-01-11 15:52 +0000 [r98164] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: Back out changes from revision 97077, since
+ it wasn't perfect
+
+2008-01-11 03:39 +0000 [r97976-98082] Russell Bryant <russell@digium.com>
+
+ * main/frame.c: Fix samples vs. length calculations for g722
+
+ * main/translate.c: Simplify this code with a suggestion from Luigi
+ on the asterisk-dev list. Instead of using is16kHz(), implement a
+ format_rate() function.
+
+ * main/translate.c: Fix various timing calculations that made
+ assumptions that the audio being processed was at a sample rate
+ of 8 kHz.
+
+2008-01-10 23:08 +0000 [r97973] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c, main/translate.c: 1) When we get a
+ translated frame out, clone it, because if the translator pvt is
+ freed before we use the frame, bad things happen. 2) Getting a
+ failure from ast_sched_delete means that the schedule ID is
+ currently running. Don't just ignore it. (Closes issue #11698)
+
+2008-01-10 21:57 +0000 [r97925] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Let us leave a voicemail for ourself if we
+ have logged into VoiceMailMain and chosen to leave a message.
+ (closes issue #11735, reported and patched by jamessan)
+
+2008-01-10 21:37 +0000 [r97849-97889] Steve Murphy <murf@digium.com>
+
+ * pbx/ael/ael_lex.c, pbx/Makefile, pbx/ael/ael.flex: Applied the
+ same fixes for ael.flex as was done in 97849 for ast_expr2.fl;
+ overrode the normally generate yyfree func with our own version
+ that checks the pointer for non-null before passing to free().
+ Also takes care of a little problem with 2.5.33 and the use of
+ the __STDC_VERSION__ macro.
+
+ * main/ast_expr2.fl, main/Makefile, main/ast_expr2f.c: This is a
+ fix for 2 things: a problem Terry was having in OSX with null
+ pointers, which was my fault, as I probably forgot to run the sed
+ script last time I made mods. So, I moved the fix into the flex
+ input itself. Then, I found when I used flex 2.5.33, that it was
+ using __STDC_VERSION__, and that's not real good; so I added back
+ in a DIFFERENT sed script to fix that little mess. Tested
+ everything, a couple different ways. Hope I did no harm, at the
+ least.
+
+2008-01-10 20:12 +0000 [r97847] Jason Parker <jparker@digium.com>
+
+ * include/asterisk/frame.h: Fix a comment that is no longer true.
+
+2008-01-10 16:19 +0000 [r97734-97753] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_kdeconsole.h (removed), configs/modules.conf.sample,
+ pbx/kdeconsole_main.cc (removed): Remove other remnants of
+ pbx_kdeconsole
+
+ * pbx/pbx_kdeconsole.cc (removed), build_tools/menuselect-deps.in,
+ configure, include/asterisk/autoconfig.h.in, configure.ac,
+ makeopts.in: Remove pbx_kdeconsole from the tree. It hasn't
+ worked in ages, and nobody has complained. (closes issue #11706,
+ reported by caio1982)
+
+2008-01-10 15:07 +0000 [r97697] Joshua Colp <jcolp@digium.com>
+
+ * funcs/func_groupcount.c: Don't try to copy the category from the
+ group if no category exists. (closes issue #11724) Reported by:
+ IgorG Patches: group_count.v1.patch uploaded by IgorG (license
+ 20)
+
+2008-01-09 23:01 +0000 [r97640-97645] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_gtkconsole.c: Strip terminal sequences from the verbose
+ messages
+
+ * pbx/pbx_gtkconsole.c: Make pbx_gtkconsole build ... but doesn't
+ actually load on my system still (related to issue #11706)
+
+2008-01-09 20:28 +0000 [r97618-97622] Jason Parker <jparker@digium.com>
+
+ * main/cli.c: Correctly display a message if a command could not be
+ found. Also fix a comment which may have led to this happening.
+ Issue 11718, reported by kshumard.
+
+ * main/cli.c: Fix some locking and return value funkiness. We
+ really shouldn't be unlocking this lock inside of a function,
+ unless we locked it there too.
+
+2008-01-09 18:48 +0000 [r97575] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: Part 2 of app_queue doxygen improvements. Some
+ smaller functions this time
+
+2008-01-09 18:02 +0000 [r97529] Russell Bryant <russell@digium.com>
+
+ * res/res_features.c: Fix saying the parking space number to the
+ caller doing the parking ...
+
+2008-01-09 17:21 +0000 [r97491] Kevin P. Fleming <kpfleming@digium.com>
+
+ * codecs/codec_zap.c: report the same message whether Zaptel does
+ not have transcoder support loaded or no transcoders were found
+
+2008-01-09 16:44 +0000 [r97489] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * channels/chan_gtalk.c: Set the caller id within the gtalk_alloc
+ function. As underlined in issue #10437 by Josh, we need to
+ prevent a possible memory leak. We only set the name part of the
+ caller id, the number part is not relevant when dealing with
+ JIDs. Closes issue #11549.
+
+2008-01-09 16:11 +0000 [r97450] Joshua Colp <jcolp@digium.com>
+
+ * apps/app_meetme.c: Don't do conferencing totally in Zaptel if
+ Monitor is running on the channel. (closes issue #11709) Reported
+ by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy
+ (license 371)
+
+2008-01-09 15:43 +0000 [r97410-97448] Kevin P. Fleming <kpfleming@digium.com>
+
+ * channels/chan_zap.c: pass the right variable to get an error
+ string... oops
+
+ * channels/chan_zap.c: add error number output to ioctl failure
+ messages to help with debugging
+
+2008-01-09 00:44 +0000 [r97350] Tilghman Lesher <tlesher@digium.com>
+
+ * main/cli.c, main/editline/readline.c: Allow filename completion
+ on zero-length modules, remove a memory leak, remove a file
+ descriptor leak, and make filename completion thread-safe.
+ Patched and tested by tilghman. (Closes issue #11681)
+
+2008-01-09 00:17 +0000 [r97206-97308] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: use the \retval doxygen command properly
+
+ * apps/app_queue.c: Part 1 of N of adding doxygen comments to
+ app_queue. I picked some of the most common functions used (which
+ also happen to be some the biggest/ugliest functions too) to
+ document first. I'm pretty new to doxygen so criticism is
+ welcome.
+
+ * apps/app_queue.c: Some coding guidelines-related cleanup
+
+2008-01-08 20:48 +0000 [r97195] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_mgcp.c: Fix various DTMF issues in chan_mgcp.
+ (closes issue #11443) Reported by: eferro Patches:
+ dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license
+ 337)
+
+2008-01-08 20:47 +0000 [r97194] Tilghman Lesher <tlesher@digium.com>
+
+ * main/autoservice.c, main/utils.c: Increase constants to where
+ we're less likely to hit them while debugging. (Closes issue
+ #11694)
+
+2008-01-08 20:42 +0000 [r97192] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_voicemail.c: Making some changes designed to not allow
+ for a corrupted mailstream for a vm_state. 1. Add locking to the
+ vm_state retrieval functions so that no linked list corruption
+ occurs. 2. Make sure to always grab the persistent vm_state when
+ mailstream access is necessary. 3. Correct an incorrect return
+ value in the init_mailstream function. (closes issue #11304,
+ reported by dwhite)
+
+2008-01-08 19:53 +0000 [r97093-97152] Joshua Colp <jcolp@digium.com>
+
+ * funcs/func_groupcount.c: If no group has been provided to the
+ GROUP_COUNT dialplan function then use the first one specific to
+ the channel. (closes issue #11077) Reported by: m4him
+
+ * apps/app_queue.c: Make app_queue calls work with directed pickup.
+ (closes issue #11700) Reported by: jbauer
+
+2008-01-08 18:02 +0000 [r97077] Tilghman Lesher <tlesher@digium.com>
+
+ * main/asterisk.c, channels/chan_sip.c: Apply multiple crash fixes,
+ found in issue #11386, but not completely closing that issue.
+
+2008-01-07 20:47 +0000 [r96884-96932] Russell Bryant <russell@digium.com>
+
+ * configs/extensions.conf.sample, /: Merged revisions 96931 via
+ svnmerge from
+ https://origsvn.digium.com/svn/asterisk/branches/1.2 ........
+ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) |
+ 2 lines Change misery.digium.com to pbx.digium.com ........
+
+ * res/res_smdi.c: Don't crash if something happens when setting up
+ an SMDI interface and it gets destroyed before the SMDI port
+ handling thread gets created.
+
+2008-01-07 14:34 +0000 [r96797-96815] Philippe Sultan <philippe.sultan@gmail.com>
+
+ * res/res_jabber.c: Indentation fix, makes the code easier to read
+
+ * res/res_jabber.c: Compute the base64 value over the
+ [authzid]\0authcid\0password string, thus excluding the trailing
+ NULL byte. This change has already been committed to trunk, see
+ #11644.
+
+2008-01-05 02:09 +0000 [r96644] Russell Bryant <russell@digium.com>
+
+ * main/devicestate.c: Don't pass an empty string as the device
+ name.
+
+2008-01-04 23:03 +0000 [r96575] Tilghman Lesher <tlesher@digium.com>
+
+ * main/devicestate.c: Fix the problem of notification of a device
+ state change to a device with a '-' in the name. Could probably
+ do with a better fix in trunk, but this bug has been open way too
+ long without a better solution. Reported by: stevedavies Patch
+ by: tilghman (Closes issue #9668)
+
+2008-01-04 22:55 +0000 [r96573] Jason Parker <jparker@digium.com>
+
+ * res/res_features.c: Properly continue in the dialplan if using
+ PARKINGEXTEN and the slot is full. Issue 11237, patch by me.
+
+2008-01-04 19:27 +0000 [r96525] Tilghman Lesher <tlesher@digium.com>
+
+ * channels/chan_sip.c: If you change the bindaddr in sip.conf to a
+ non-bound address and reload, sip goes kablooie. Reported and
+ patched by: one47 (Closes issue #11535)
+
+2008-01-04 16:19 +0000 [r96394-96449] Russell Bryant <russell@digium.com>
+
+ * channels/chan_zap.c: Make use of the temporary channel pointer
+ while the pvt is unlocked. (closes issue #11675) Reported by:
+ flefoll Patches: chan_zap.c.patch-store-owner-before-unlock
+ uploaded by flefoll (license 244)
+
+ * channels/chan_iax2.c: Don't crash if the iax2 pvt structure has
+ been destroyed before we get to this point (closes issue #11672,
+ reported by snuffy, patched by me)
+
+2008-01-03 21:37 +0000 [r96318] Tilghman Lesher <tlesher@digium.com>
+
+ * res/res_config_pgsql.c: Missed initialization caused crash.
+ Reported and fixed by: tiziano (Closes issue #11671)
+
+2008-01-03 12:12 +0000 [r96198-96199] Christian Richter <christian.richter@beronet.com>
+
+ * channels/chan_misdn.c: make sure frame is completely clean,
+ before we send it to asterisk as DTMF. If we don't make it clean,
+ it happens that one way audio occurs..
+
+ * channels/chan_misdn.c: when overlapdial was used and no number
+ was dialed, the call was dropped, now we just jump into the s
+ extension, which makes a lot more sense.
+
+2008-01-02 23:46 +0000 [r96102] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: We need to reset the membername to NULL on each
+ iteration of this loop, otherwise the result is that multiple
+ members can have the same name, since the variable was not reset
+ on each iteration of the loop.
+
+2008-01-02 22:14 +0000 [r96020-96024] Russell Bryant <russell@digium.com>
+
+ * pbx/pbx_config.c: Convert locks of the contexts list in
+ pbx_config to the appropriate rdlock or wrlock
+
+ * pbx/pbx_dundi.c: pbx_dundi only needs a rdlock on the contexts
+ list.
+
+ * apps/app_macro.c: app_macro only needs a rdlock on the contexts
+ list.
+
+2008-01-02 Russell Bryant <russell@digium.com>
+
+ * Asterisk 1.4.17 released.
+
+2008-01-02 20:24 +0000 [r95946] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_sip.c: Allocate a SIP refer structure when
+ performing a transfer using BYE with Also so that the transfer
+ information is properly stored. (AST-2008-028) (closes issue
+ #11637) Reported by: greyvoip
+
+2008-01-02 17:51 +0000 [r95890] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: A change to improve the accuracy of queue
+ logging in the case where a member does not answer during the
+ specified timeout period. Prior to this change, there was a small
+ chance that the member name recorded in this case would be blank.
+ Also prior to this change, if using the ringall strategy, if no
+ one answered the call during the specified timeout, the member
+ name listed in the queue log would randomly be one of the members
+ that was rung. (closes issue #11498, reported and tested by
+ hloubser, patched by me)
+
+2007-12-31 23:43 +0000 [r95577] Mark Michelson <mmichelson@digium.com>
+
+ * main/pbx.c: Avoiding a potentially bad locking situation.
+ ast_merge_contexts_and_delete writelocks the conlock, then calls
+ ast_hint_extension, which attempts to readlock the same lock.
+ Recursion with read-write locks is dangerous, so the inner lock
+ needs to be removed. I did this by copying the "guts" of
+ ast_hint_extension into ast_merge_contexts_and_delete (sans the
+ extra lock). (this change is inspired by the locking problems
+ seen in issue #11080, but I have no idea if this is the
+ problematic area experienced by the reporters of that issue)
+
+2007-12-31 20:27 +0000 [r95470] Tilghman Lesher <tlesher@digium.com>
+
+ * funcs/func_env.c: Allow the default "0" to be returned if the
+ STAT fails (Closes issue #11659)
+
+2007-12-28 18:24 +0000 [r95191] Russell Bryant <russell@digium.com>
+
+ * channels/chan_sip.c: Remove duplicate increment of the header
+ count in the add_header() function. (closes issue #11648)
+ Reported by: makoto Patch provided by sergee, committed patch by
+ me, inspired by comments from putnopvut
+
+2007-12-28 00:16 +0000 [r95095] Mark Michelson <mmichelson@digium.com>
+
+ * apps/app_queue.c: I found a bug while browsing the queue code and
+ managed to reproduce it in a small setup. If a queue uses the
+ ringall strategy, it was possible through unfortunate coincidence
+ for a single member at a given penalty level to make app_queue
+ think that all members at that penalty level were unavailable and
+ cause the members at the next penalty level to be rung. With this
+ patch, we will only move to the next penalty level if ALL the
+ members at a given penalty level are unreachable.
+
+2007-12-27 21:40 +0000 [r95024] Russell Bryant <russell@digium.com>
+
+ * main/channel.c: Don't report a syntax error when an empty string
+ is passed to ast_get_group. Just return 0. (closes issue #11540)
+ Reported by: tzafrir Patches: group_empty.diff uploaded by
+ tzafrir (license 46) -- slightly changed by me
+
+2007-12-27 20:09 +0000 [r94977] Mark Michelson <mmichelson@digium.com>
+
+ * main/io.c: Fixing a typo in a comment.
+
+2007-12-27 17:32 +0000 [r94905-94924] Joshua Colp <jcolp@digium.com>
+
+ * channels/chan_h323.c: Include types.h in chan_h323 as without it
+ it can not be compiled on some operating systems like FreeBSD to
+ name one. (closes issue #11585) Reported by: sobomax Patches:
+ chan_h323.c.diff uploaded by sobomax (license 359)
+
+ * channels/chan_sip.c: Use ast_strlen_zero to see if our_contact is
+ set or not on the dialog. It is possible for it to be a pointer
+ to NULL. (closes issue #11557) Reported by: FuriousGeorge
+
+2007-12-27 15:16 +0000 [r94828-94831] Russell Bryant <russell@digium.com>
+
+ * main/pbx.c: Now that the contexts lock is a read/write lock, it
+ should not be locked here in ast_hint_state_changed(). This makes
+ it get locked recursively which now causes a deadlock. (closes
+ issue #11080, thanks to callguy for the access to a deadlocked
+ machine)
+
+ * include/asterisk/translate.h, main/translate.c: Use the constant
+ that I really meant to use here ...
+
+ * main/translate.c: Change ast_translator_best_choice() to only pay
+ attention to audio formats. This fixes a problem where Asterisk
+ claims that a translation path can not be found for channels
+ involving video. (closes issue #11638) Reported by: cwhuang
+ Tested by: cwhuang Patch suggested by cwhuang, with some
+ additional changes by me.
+
+2007-12-27 01:01 +0000 [r94824] Kevin P. Fleming <kpfleming@digium.com>
+
+ * main/manager.c: make this comment explain the situation in an
+ even more explicit fashion
+
+2007-12-26 20:43 +0000 [r94808] Tilghman Lesher <tlesher@digium.com>
+