From: Asterisk Autobuilder Date: Mon, 10 Dec 2012 01:58:49 +0000 (+0000) Subject: Importing files for 11.2.0-rc1 release. X-Git-Tag: 11.2.0-rc1~2 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=95d4d363ca6d4a434841d79aed2debcac344f578;p=thirdparty%2Fasterisk.git Importing files for 11.2.0-rc1 release. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/11.2.0-rc1@377526 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/.lastclean b/.lastclean new file mode 100644 index 0000000000..425151f3a4 --- /dev/null +++ b/.lastclean @@ -0,0 +1 @@ +40 diff --git a/.version b/.version new file mode 100644 index 0000000000..94ba9ae2bd --- /dev/null +++ b/.version @@ -0,0 +1 @@ +11.2.0-rc1 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 0000000000..8c03a7aa25 --- /dev/null +++ b/ChangeLog @@ -0,0 +1,22369 @@ +2012-12-10 Asterisk Development Team + + * Asterisk 11.2.0-rc1 Released. + +2012-12-10 01:41 +0000 [r377505-377511] Tilghman Lesher + + * main/xmldoc.c, /: Improve documentation by making all of the + colors used readable, no matter what the background color is. + Dark blue on a black background is unreadable, as is yellow on a + light background. This patch turns on the bright attribute for + colors when on a dark background and turns *off* the bright + attribute when the -W command line option is used (indicating a + _light_ background). This ensures that text is readable in both + cases. Patch by: tilghman Review: + https://reviewboard.asterisk.org/r/2224 ........ Merged revisions + 377509 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 377510 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, addons/cdr_mysql.c: Remove some dead code and additionally + handle a case that wasn't handled. ........ Merged revisions + 377487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 377504 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-12-09 01:22 +0000 [r377462] Joshua Colp + + * channels/chan_motif.c: Add missing support for "who hung up" to + chan_motif. (closes issue ASTERISK-20671) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/2208/ + +2012-12-08 00:29 +0000 [r377401-377433] Richard Mudgett + + * contrib/realtime/mysql/sippeers.sql, /: Fix order of SIP + allow/disallow in MySQL contrib script. Using the contrib + sippeers.sql script to create the sippeers MySQL table would + result in being unable to place calls if you set the disallow + value to all. (closes issue ASTERISK-20756) Reported by: Andre + Luis Patches: sippeers.patch patch uploaded by Andre Luis + ........ Merged revisions 377431 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 377432 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/astmm.c: MALLOC_DEBUG: Only wait if we want atexit + allocation dumps. ........ Merged revisions 377398 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 377399 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-12-07 22:02 +0000 [r377383] Kinsey Moore + + * /, codecs/codec_dahdi.c: codec_dahdi: Fix output of "transcoder + show" CLI command. In r306010 "Asterisk media architecture + conversion - no more format bitfields", the logic for + incrementing encoders and decoders when opening transcoder + channels was changed without making the corresponding change when + decrementing encoder / decoder channels. The result being that + when a channel was destroyed, codec_dahdi couldn't properly tell + if it was an encoder or decoder, and the default case is to + assume it was a decoder. This could result in negative numbers + for decoders in use like in: VOIP6*CLI> transcoder show 2/-2 + encoders/decoders of 92 channels are in use. (closes issue + ASTERISK-19921) Patch-by: Shaun Ruffell ........ Merged revisions + 377382 from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-12-06 23:58 +0000 [r377355] Richard Mudgett + + * apps/confbridge/conf_config_parser.c, /, apps/app_confbridge.c: + confbridge: Fix some resource leaks on conference teardown. * + Made destroy_conference_bridge() destroy a missed ast_mutex_t and + ast_cond_t. * Made join_conference_bridge() init the + ast_mutex_t's and ast_cond_t so destroy_conference_bridge() can + destroy them unconditionally. * Made join_conference_bridge() + abort if the new conference could not be added to the conferences + container. * Made leave_conference() discard any post-join + actions if join_conference_bridge() had to abort early. * Made + the join_conference_bridge() diagnostic messages better describe + what happened. * Renamed leave_conference_bridge() to + leave_conference() and made it only take a conference user + pointer. The conference pointer was redundant. * Made + conf_bridge_profile_copy() use struct copy instead of memcpy(). * + No need to lock the conference in start_conf_record_thread() + since all of the callers already have it locked. ........ Merged + revisions 377354 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-12-06 17:28 +0000 [r377340] Russell Bryant + + * main/named_acl.c: Add CLI tab completion to 'acl show'. The 'acl + show' CLI command allows you to show the details about a specific + named ACL in acl.conf. This patch adds tab completion to the + command. Review: https://reviewboard.asterisk.org/r/2230/ + +2012-12-06 14:11 +0000 [r377319] Matthew Jordan + + * main/manager.c: Fix memory leak in 'manager show event' when + command entered incorrectly When the CLI command 'manager show + event' was run incorrectly and its usage instructions returned, a + reference to the event container was leaked. This would prevent + the container from being reclaimed when Asterisk exits. We now + properly decrement the count on the ao2 object using the nifty + RAII_VAR macro. Thanks to Russell for helping me stumble on this, + and Terry for writing that ridiculously helpful macro. + +2012-12-05 17:08 +0000 [r377262] Jonathan Rose + + * res/res_srtp.c, /: res_srtp: Fix a crash caused by srtp_dealloc + on an already dealloced session When srtp_create fails, the + session may be dealloced or just not alloced. At the same time + though, the session pointer might not be set to NULL in this + process and attempting to srtp_dealloc it again will cause a + segfault. This patch checks for failure of srtp_create and sets + the session pointer to NULL if it fails. (closes issue + ASTERISK-20499) Reported by: tootai Review: + https://reviewboard.asterisk.org/r/2228/ ........ Merged + revisions 377256 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 377261 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-12-05 16:50 +0000 [r377259] Joshua Colp + + * /, channels/chan_sip.c: Fix a SIP request memory leak with TLS + connections. During the TLS re-work in chan_sip some TLS specific + code was moved into a separate function. This function operates + on a copy of the incoming SIP request. This copy was never + deinitialized causing a memory leak for each request processed. + This function is now given a SIP request structure which it can + use to copy the incoming request into. This reduces the amount of + memory allocations done since the internal allocated components + are reused between packets and also ensures the SIP request + structure is deinitialized when the TLS connection is torn down. + (closes issue ASTERISK-20763) Reported by: deti ........ Merged + revisions 377257 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 377258 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-12-05 02:19 +0000 [r377213-377244] Richard Mudgett + + * main/format.c, /: Fix registering core show codecs/codec CLI + commands twice. ........ Merged revisions 377241 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/confbridge/conf_config_parser.c, /: confbridge: Fix several + small issues. * Made func_confbridge_helper() allow an empty + value when setting options. You previously could not + Set(CONFBRIDGE(user,pin)=) and clear the configured pin from the + dialplan. * Made func_confbridge_helper() handle its datastore + better if multiple threads attempt to set the first CONFBRIDGE + option value on the channel. * Made the func_confbridge_helper() + only output one diagnostic message concerning the option. * Made + the bridge video_mode able to repeatedly change in the config + file and CONFBRIDGE dialplan function. The video_mode option + values are an enum and not independent of each other. * Made + handle_cli_confbridge_show_bridge_profile() better handle the + video_mode option. * Simplified datastore handling code in + conf_find_user_profile() and conf_find_bridge_profile(). (closes + issue ASTERISK-20655) Reported by: Birger "WIMPy" Harzenetter + ........ Merged revisions 377227 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_confbridge.c: confbridge: Update online XML + documentation. ........ Merged revisions 377212 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-12-04 12:59 +0000 [r377195] Russell Bryant + + * contrib/scripts/install_prereq: Add libuuid to install_prereq for + Fedora. I ran this script and my build failed. pjproject requires + this. + +2012-12-03 22:58 +0000 [r377039-377167] Richard Mudgett + + * main/asterisk.c, /: Cleanup ast_run_atexits() atexits list. * + Convert atexits list to a mutex instead of a rd/wr lock. The lock + is only write locked. * Move CLI verbose Asterisk ending message + to where AMI message is output in really_quit() to avoid further + surprises about using stuff already shutdown. (issue + ASTERISK-20649) Reported by: Corey Farrell ........ Merged + revisions 377165 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 377166 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/asterisk.c, /, include/asterisk/_private.h, + main/stdtime/localtime.c: Cleanup core main on exit. * Cleanup + time zones on exit. * Make exit clean/unclean report consistent + for AMI and CLI in really_quit(). (issue ASTERISK-20649) Reported + by: Corey Farrell Patches: core-cleanup-1_8-10.patch (license + #5909) patch uploaded by Corey Farrell + core-cleanup-11-trunk.patch (license #5909) patch uploaded by + Corey Farrell Modified ........ Merged revisions 377135 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 377136 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/config.c, /: Cleanup config cache on exit. (issue + ASTERISK-20649) Reported by: Corey Farrell Patches: + config-cleanup-all.patch (license #5909) patch uploaded by Corey + Farrell ........ Merged revisions 377104 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 377105 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/cli.c, /: Cleanup CLI resources on exit and CLI command + registration errors. (issue ASTERISK-20649) Reported by: Corey + Farrell Patches: cli-leaks-1_8-10.patch (license #5909) patch + uploaded by Corey Farrell cli-leaks-11-trunk.patch (license + #5909) patch uploaded by Corey Farrell Modified ........ Merged + revisions 377073 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 377074 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/cdr.c, /: Cleanup CDR resources on exit. * Simplify + do_reload() return handling since it never returned anything + other than 0. (issue ASTERISK-20649) Reported by: Corey Farrell + Patches: cdr-cleanup-1_8.patch (license #5909) patch uploaded by + Corey Farrell cdr-cleanup-10-11-trunk.patch (license #5909) patch + uploaded by Corey Farrell Modified ........ Merged revisions + 377069 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 377070 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/ccss.c: Fix CCSS CLI commands and logger level not + unregistered. (issue ASTERISK-20649) Reported by: Corey Farrell + Patches: ccss-cleanup-all.patch (license #5909) patch uploaded by + Corey Farrell ........ Merged revisions 377037 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 377038 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-12-03 14:54 +0000 [r377021] Joshua Colp + + * channels/chan_motif.c: Fix an RTP instance reference count leak + in chan_motif. When setting up an RTP instance the RTCP portion + of the instance keeps a reference to the instance itself. In + order to release this reference and stop RTCP the stop API call + must be called before destroying the instance. (closes issue + ASTERISK-20751) Reported by: joshoa + +2012-12-01 00:46 +0000 [r376983] Joshua Colp + + * configs/motif.conf.sample, channels/chan_motif.c: Tweak extension + used for incoming calls received on Motif. Based on feedback from + numerous individuals this patch tweaks incoming calls to first + look for an extension with the name of the endpoint. If no such + extension exists the call will silently fall back to the "s" + extension as it previously did. + +2012-11-30 21:35 +0000 [r376952] Richard Mudgett + + * /, channels/misdn/isdn_lib.c: chan_misdn: Fix sending + RELEASE_COMPLETE in response to SETUP. Fix sending a + RELEASE_COMPLETE in response to a SETUP if chan_misdn does not + have a B channel available to assign to the call. (closes issue + ABE-2869) Reported by: Guenther Kelleter Patches: + setup-reject_2.diff (license #6372) patch uploaded by Guenther + Kelleter Modified ........ Merged revision 376949 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + ........ Merged revisions 376950 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376951 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-30 17:07 +0000 [r376921] Sean Bright + + * /, funcs/func_volume.c: Minor spelling fix to the VOLUME + documentation. ........ Merged revisions 376919 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376920 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-30 16:36 +0000 [r376917] Mark Michelson + + * /, channels/chan_sip.c: Fix potential crashes during SIP attended + transfers. The principal behind this patch is simple. During a + transfer, we manipulate channels that are owned by a separate + thread than the one we currently are running in, so it makes + sense that we need to grab a reference to the channels so that + they cannot disappear out from under us. In the wild, crashes + were sometimes seen when the transferring party would hang up the + call before the transfer target answered the call. The most + common place to see the crash occur was when attempting to send a + connected line update to the transferer channel. (closes issue + ASTERISK-20226) Reported by Jared Smith Patches: + ASTERISK-20226.patch uploaded by Mark Michelson (License #5049) + Tested by: Jared Smith ........ Merged revisions 376901 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376916 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-29 22:59 +0000 [r376866-376870] Richard Mudgett + + * channels/chan_local.c, /: chan_local: Fix local_pvt ref leak in + local_devicestate(). Regression introduced by ASTERISK-20390 fix. + (closes issue ASTERISK-20769) Reported by: rmudgett Tested by: + rmudgett ........ Merged revisions 376868 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376869 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Fix compile error. (issue ASTERISK-20724) + ........ Merged revisions 376864 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376865 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-29 21:57 +0000 [r376836] Michael L. Young + + * /, channels/chan_sip.c: Improve Code Readability And Fix Setting + natdetected Flag For 1.8, 10, 11 and trunk we are are improving + the code readability. For 11 and trunk, auto nat detection was + added. The natdetected flag was being set to 1 when the host + address in the VIA header did not specifiy a port. This patch + fixes this by setting the port on the temporary sock address used + to SIP_STANDARD_PORT in order for the sock address comparison to + work properly. (closes issue ASTERISK-20724) Reported by: Michael + L. Young Patches: asterisk-20724-set-port-v2.diff uploaded by + Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2206/ ........ Merged + revisions 376834 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376835 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-29 17:17 +0000 [r376822] Pedro Kiefer + + * channels/chan_sip.c: Fix chan_sip websocket payload handling + Websocket by default doesn't return an ast_str for the payload + received. When converting it to an ast_str on chan_sip the last + character was being omitted, because ast_str functions expects + that the given length includes the trailing 0x00. payload_len + only has the actual string length without counting the trailing + zero. For most cases this passed unnoticed as most of SIP + messages ends with \r\n. (closes issue ASTERISK-20745) Reported + by: Iñaki Baz Castillo Review: + https://reviewboard.asterisk.org/r/2219/ + +2012-11-29 00:46 +0000 [r376760-376790] Richard Mudgett + + * main/asterisk.c, /, main/astmm.c: Add MALLOC_DEBUG atexit + unreleased malloc memory summary. * Adds the following CLI + commands to control MALLOC_DEBUG reporting of unreleased malloc + memory when Asterisk is shut down. memory atexit list on memory + atexit list off memory atexit summary byline memory atexit + summary byfunc memory atexit summary byfile memory atexit summary + off * Made check all remaining allocated region blocks atexit for + fence violations. * Increased the allocated region hash table + size by about three times. It still isn't large enough + considering the number of malloced blocks Asterisk uses. * Made + CLI "memory show allocations anomalies" use + regions_check_all_fences(). Review: + https://reviewboard.asterisk.org/r/2196/ ........ Merged + revisions 376788 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376789 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/astmm.c: Enhance MALLOC_DEBUG CLI commands. * Fixed CLI + "memory show allocations" misspelling of anomalies option. The + command will still accept the original misspelling. * + Miscellaneous tweaks to CLI "memory show allocations" command + output format. * Made CLI "memory show summary" summarize by line + number instead of by function if a filename is given. * Made CLI + "memory show summary" sort its output by filename or + function-name/line-number depending upon request. * Miscellaneous + tweaks to CLI "memory show summary" command output format. + ........ Merged revisions 376758 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376759 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-28 16:37 +0000 [r376727] Jonathan Rose + + * main/manager.c, /: manager: Make challenge work with + allowmultiplelogin=no Prior to this patch, challenge would yield + a multiple logins error if used without providing the username + (which isn't really supposed to be an argument to challenge) if + allowmultiplelogin was set to no because allowmultiplelogin finds + a user with a zero length login name. This check is simply + disabled for the challenge action when the username is empty by + this patch. (closes issue ASTERISK-20677) Reported by: Vladimir + Patches: challenge_action_nomultiplelogin.diff uploaded by + Jonathan Rose (license 6182) ........ Merged revisions 376725 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 376726 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-28 00:08 +0000 [r376629-376690] Richard Mudgett + + * main/pbx.c, /, UPGRADE.txt: Fix extension matching with the '-' + char. The '-' char is supposed to be ignored by the dialplan + extension matching. Unfortunately, it's treatment is not handled + consistently throughout the extension matching code. * Made the + old exten matching code consistently ignore '-' chars. * Made the + old exten matching code consistently handle case in the matching. + * Made ignore empty character sets. * Fixed ast_extension_cmp() + to return -1, 0, or 1 as documented. The only user of it in + pbx_lua.c was testing for -1. It was originally returning the + strcmp() value for less than which is not usually going to be -1. + * Fix character set sorting if the sets have the same number of + characters and start with the same character. Character set [0-9] + now sorts before [02-9a] as originally intended. * Updated some + extension label and priority already in use warnings to also + indicate if the extension is aliased. (closes issue + ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy" + Harzenetter Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/2201/ ........ Merged + revisions 376688 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376689 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * addons/res_config_mysql.c, /, apps/app_celgenuserevent.c, + pbx/pbx_dundi.c: Remove unnecessary channel module references. * + Removed call to ast_module_user_hangup_all() in + res_config_mysql.c since it is effectively a noop. No channels + can attach a reference to that module. * Removed call to + ast_module_user_hangup_all() in app_celgenuserevent.c. The caller + of unload_module() has already called it. * Removed redundant + channel module references in pbx_dundi.c. The registered dialplan + function callback dispatchers for the read/read2/write callbacks + already reference the module before calling. * pbx_dundi: Moved + unregistering CLI commands, DUNDi switch, and dialplan functions + to the first thing the unload_module() does. This will reduce the + chance of new channels using DUNDi services while the module is + being torn down. ........ Merged revisions 376657 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376658 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, include/asterisk/linkedlists.h: Made AST_LIST_REMOVE() simpler + and use better names. * Update doxygen of AST_LIST_REMOVE(). + ........ Merged revisions 376627 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376628 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-22 23:58 +0000 [r376588] Matthew Jordan + + * main/lock.c, /, main/logger.c, include/asterisk/lock.h: + Re-initialize logmsgs mutex upon logger initialization to prevent + lock errors Similar to the patch that moved the fork earlier in + the startup sequence to prevent mutex errors in the recursive + mutex surrounding the read/write thread registration lock, this + patch re-initializes the logmsgs mutex. Part of the start up + sequence before forking the process into the background includes + reading asterisk.conf; this has to occur prior to the call to + daemon in order to read startup parameters. When reading in a + conf file, log statements can be generated. Since this can't be + avoided, the mutex instead is re-initialized to ensure a reset of + any thread tracking information. This patch also includes some + additional debugging to catch errors when locking or unlocking + the recursive mutex that surrounds locks when the DEBUG_THREADS + build option is enabled. DO_CRASH or THREAD_CRASH will cause an + abort() if a mutex error is detected. (issue ASTERISK-19463) + Reported by: mjordan Tesetd by: mjordan ........ Merged revisions + 376586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 376587 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-20 21:58 +0000 [r376561] David M. Lee + + * res/res_http_websocket.c: Added missing newlines to websocket + ast_logs. Without these newlines, log messages just continue + tacking onto the same line, and do not flush immediately. + +2012-11-20 18:57 +0000 [r376550] Mark Michelson + + * channels/sip/include/sip.h, /, channels/chan_sip.c: Add "Require: + timer" to 200 OK responses when appropriate. The method by which + the Require header is added to 200 responses is inspired by the + method that Olle Johansson uses in his darjeeling-prack branch. + (closes issue ASTERISK-20570) Reported by Matt Jordan, at the + behest of Olle Johansson Review: + https://reviewboard.asterisk.org/r/2172 ........ Merged revisions + 376521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 376522 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-20 17:37 +0000 [r376540] Alec L Davis + + * channels/chan_sip.c: Reduce CLI spam of "Extension Changed" + device state messages. Asterisk 11 follows RFC3265 that states + that after every subscribe or resubscribe a notify should be + sent. Thus the console if filled continuously with the following + after every subscribe; == Extension Changed 8512[phones] new + state IDLE for Notify User cisco1 In Asterisk 1.8 only changes + would be sent. Thus only when a device state changed was anything + emitted to the console. fix: Only print to console when device + state isn't forced. (closes issue ASTERISK-20706) Reported by: + alecdavis Tested by: alecdavis alecdavis (license 585) + +2012-11-19 19:54 +0000 [r376471] Walter Doekes + + * /, channels/chan_sip.c, main/security_events.c, + main/indications.c: Fix most leftover non-opaque ast_str uses. + Instead of calling str->str, one should use ast_str_buffer(str). + Same goes for str->used as ast_str_strlen(str) and str->len as + ast_str_size(str). Review: + https://reviewboard.asterisk.org/r/2198 ........ Merged revisions + 376469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 376470 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-18 20:22 +0000 [r376415-376441] Matthew Jordan + + * main/asterisk.c, /, main/utils.c: Reorder startup sequence to + prevent lockups when process is sent to background Although it is + very rare and timing dependent, the potential exists for the call + to 'daemon' to cause what appears to be a deadlock in Asterisk + during startup. This can occur when a recursive mutex is obtained + prior to the daemon call executing. Since daemon uses fork to + send the process into the background, any threading primitives + are unsafe to re-use after the call. Implementations of pthread + recursive mutexes are highly likely to store the thread + identifier of the thread that previously obtained the mutex. If + the mutex was locked prior to the fork, a subsequent unlock + operation will potentially fail as the thread identifier is no + longer valid. Since the mutex is still locked, all subsequent + attempts to grab the mutex by other threads will block. This + behavior exhibited itself most often when DEBUG_THREADS was + enabled, as this compile time option surrounds the mutexes in + Asterisk with another recursive mutex that protects the storage + of thread related information. This made it much more likely that + a recursive mutex would be obtained prior to daemon and unlocked + after the call. This patch does the following: a) It backports a + patch from Asterisk 11 that prevents the spawning of the + localtime monitoring thread. This thread is now spawned after + Asterisk has fully booted. b) It re-orders the startup sequence + to call daemon earlier during Asterisk startup. This limits the + potential of threading primitives being accessed by + initialization calls before daemon is called. c) It removes calls + to ast_verbose/ast_log/etc. prior to daemon being called. + Developers should send error messages directly to stderr prior to + daemon, as calls to ast_log may access recursive mutexes that + store thread related information. d) It reorganizes when thread + local storage is created for storing lock information during the + creation of threads. Prior to this patch, the read/write lock + protecting the list of threads in ast_register_thread would + utilize the lock in the thread local storage prior to it being + initialized; this patch prevents that. On a very related note, + this patch will *greatly* improve the stability of the Asterisk + Test Suite. Review: https://reviewboard.asterisk.org/r/2197 + (closes issue ASTERISK-19463) Reported by: mjordan Tested by: + mjordan ........ Merged revisions 376428 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376431 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/confbridge/conf_state.c, /: Add a test event that reports + changes in ConfBridge state This patch adds a test event to + ConfBridge that reports transitions between states in ConfBridge. + This is used by tests in the Asterisk Test Suite that verify + state changes based on the entering/leaving of conference + participants. ........ Merged revisions 376414 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-16 19:59 +0000 [r376391] Jonathan Rose + + * res/res_monitor.c, /: monitor: prevent attempts to move/remove + recordings skipped with 'i' and 'o'. The i and o options for + monitor skip the input and output sides of a recording + respectively. This patch addresses a problem in those options + when monitor is called without specifying a specific filename + where monitor will try to move the recording that was skipped. + Since this usually doesn't exist when these options are used, it + would produce a warning when it does this in most cases, but it + is conceivable that there are use cases where this could result + in moving/removing a file unintentionally. (closes issue + ASTERISK-20641) Reported by: Jonathan Rose Review: + https://reviewboard.asterisk.org/r/2190/ ........ Merged + revisions 376389 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376390 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-16 00:09 +0000 [r376339-376343] David M. Lee + + * /, utils/extconf.c: Fixed extconf.c breakage introduced in + r376306. To quote wdoekes: > Note that I'm not confirming + legitimacy of having that file in tree at > all. Is anyone using + aelparse/conf2ael? ........ Merged revisions 376340 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376342 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * utils/Makefile, tests/test_astobj2_thrash.c (added), + utils/utils.xml, /, utils/hashtest.c (removed), + tests/test_hashtab_thrash.c (added), utils/hashtest2.c (removed), + include/asterisk/hashtab.h: Migrate hashtest/hashtest2 to be unit + tests. Both hashtest and hashtest2 are manual testing apps that + thrash hash tables (hashtab and ao2 containers, respectively), by + spinning up several threads that randomly insert, delete, lookup + and iterate over the hash table. If the app doesn't crash, the + hash table probably passes the test. Those utils are not a part + of the typical Asterisk build, so they do not usually get + compiled. This all makes them less that useful. This patch + removes those manual test programs and replaces them with + Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It + also attempts to make the tests more deterministic. * Rather than + spinning up some number of threads that operate on the hash table + randomly, spin up four threads that concurrenly add, remove, + lookup and iterate over the hash table. * Each thread checks the + state of the hash table both during and after execution, and + indicates a test failure if things are not as expected. * Each + thread times out after 60 seconds to prevent deadlocking the unit + test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged + revisions 376306 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376315 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-15 23:03 +0000 [r376310] Jonathan Rose + + * /, apps/app_meetme.c: app_meetme: Fix channels lingering when + hung up under certain conditions Channels would get stuck and + MeetMe would repeatedly display an Unable to write frame to + channel error in the conf_run function if hung up during certain + sound prompts such as during user count announcements. This patch + fixes that by reintroducing a hangup check in the meetme's main + loop (also in conf_run). (closes issue ASTERISK-20486) Reported + by: Michael Cargile Review: + https://reviewboard.asterisk.org/r/2187/ Patches: + meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan + Rose (license 6182) ........ Merged revisions 376307 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376308 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-15 02:08 +0000 [r376264] Rusty Newton + + * apps/app_voicemail.c, /: Patch to play correct sound file when a + voicemail's urgent status is removed We were attempting to play + "vm-urgent-removed", which didn't exist. Now we play + "vm-marked-nonurgent" which exists and is the correct sound file. + Previous behavior was silence and a warning on the CLI. (issue + ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo + Takebe Tested by: Rusty Newton Patches: asterisk20280.patch + uploaded by Rusty Newton (license 5829) ........ Merged revisions + 376262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 376263 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-14 19:53 +0000 [r376234] Richard Mudgett + + * pbx/pbx_spool.c, /: Fix call files when astspooldir is relative. + Future dated call files are ignored when astspooldir is relative + to the current directory. The queue_file() assumed that the qdir + needed to be prepended if the given filename did not start with a + '/'. If astspooldir is relative it is not going to start from the + root directory obviously so it will not start with a '/'. The + filename used in queue_file() ultimately results in qdir + prepended multiple times. * Made queue_file() not prepend qdir if + the filename contains a '/'. (closes issue ASTERISK-20593) + Reported by: James Le Cuirot Patches: + 0004-Fix-future-call-files-from-relative-directories.patch + (license #6439) patch uploaded by James Le Cuirot ........ Merged + revisions 376232 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376233 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-13 18:48 +0000 [r376217] Brent Eagles + + * main/channel.c, /: Patch to prevent stopping the active generator + when it is not the silence generator. This patch introduces an + internal helper function to safely check whether the current + generator is the one that is expected before deactivating it. The + current externally accessible ast_channel_stop_generator() + function has been modified to be implemented in terms of the new + function. (closes issue ASTERISK-19918) Reported by: Eduardo Abad + ........ Merged revisions 376199 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376208 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-12 20:45 +0000 [r376168] Joshua Colp + + * main/pbx.c, /: Properly check if the "Context" and "Extension" + headers are empty in a ShowDialPlan action. The code which + handles the ShowDialPlan action wrongly assumed that a non-NULL + return value from the function which retrieves headers from an + action indicates that the header has a value. This is incorrect + and the contents must be checked to see if they are blank. + (closes issue ASTERISK-20628) Reported by: jkroon Patches: + asterisk-showdialplan-incorrect-error.patch uploaded by jkroon + ........ Merged revisions 376166 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376167 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-12 20:16 +0000 [r376144] Michael L. Young + + * main/pbx.c, /: Fix Dynamic Hints Variable Substition - Underscore + Problem When adding a dynamic hint, if an extension contains an + underscore no variable subsitution is being performed. This patch + changes from checking if the extension contains an underscore to + checking if the extension begins with an underscore. (closes + issue ASTERISK-20639) Reported by: Steven T. Wheeler Tested by: + Steven T. Wheeler, Michael L. Young Patches: + asterisk-20639-dynamic-hint-underscore.diff uploaded by Michael + L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2188/ ........ Merged + revisions 376142 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376143 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-11 17:08 +0000 [r376130] Joshua Colp + + * res/res_rtp_asterisk.c, channels/chan_sip.c, + configs/sip.conf.sample: Remove a fixed size limitation for + producing SDP and change how ICE support is disabled by default. + With ICE support enabled in chan_sip and a large number of + interfaces on the system it was possible for the produced SDP to + be truncated due to some fixed size buffers. These buffers have + now been changed so they will dynamically grow as needed. ICE + support is now also enabled by default in res_rtp_asterisk to + provide a smoother experience for chan_motif users where it is + required. To maintain the previous behavior in chan_sip it is no + longer enabled by default there. (closes issue ASTERISK-20643) + Reported by: coopvr + +2012-11-08 22:08 +0000 [r376089] Mark Michelson + + * /, res/res_fax.c: Fix a "set but not used" warning on newer gccs. + Turns out the "helpful" setting of ms and res in this macro is + completely useless after the timeout antipattern fix. If you're a + new guy looking to write code, don't write a macro like this one. + ........ Merged revisions 376087 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376088 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-08 21:10 +0000 [r376048-376060] Richard Mudgett + + * channels/sig_ss7.c, /: chan_dahdi/SS7: Made reject incoming call + for an in-alarm or blocked channel. If a SS7 call comes in + requesting a CIC that is in-alarm, the call is accepted and + connects if the extension exists in the dialplan. The call does + not have any audio. * Made release the call immediately with + circuit congestion cause. (closes issue ASTERISK-20204) Reported + by: Tuan Le Patches: jira_asterisk_20204_v1.8.patch (license + #5621) patch uploaded by rmudgett ........ Merged revisions + 376058 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 376059 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/asterisk.c, include/asterisk/utils.h, + include/asterisk/astmm.h, /, main/utils.c, main/astmm.c: Add + MALLOC_DEBUG enhancements. * Makes malloc() behave like calloc(). + It will return a memory block filled with 0x55. A nonzero value. + * Makes free() fill the released memory block and boundary + fence's with 0xdeaddead. Any pointer use after free is going to + have a pointer pointing to 0xdeaddead. The 0xdeaddead pointer is + usually an invalid memory address so a crash is expected. * Puts + the freed memory block into a circular array so it is not reused + immediately. * When the circular array rotates out a memory block + to the heap it checks that the memory has not been altered from + 0xdeaddead. * Made the astmm_log message wording better. * Made + crash if the DO_CRASH menuselect option is enabled and something + is found. * Fixed a potential alignment issue on 64 bit systems. + struct ast_region.data[] should now be aligned correctly for all + platforms. * Extracted region_check_fences() from + __ast_free_region() and handle_memory_show(). * Updated + handle_memory_show() CLI usage help. Review: + https://reviewboard.asterisk.org/r/2182/ ........ Merged + revisions 376029 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 376030 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-07 19:03 +0000 [r376014] Mark Michelson + + * include/asterisk/time.h, apps/app_jack.c, apps/app_dial.c, + main/pbx.c, main/rtp_engine.c, /, apps/app_meetme.c, + res/res_fax.c, apps/app_record.c, channels/chan_agent.c, + main/utils.c, include/asterisk/channel.h, apps/app_queue.c, + channels/sig_pri.c, channels/chan_iax2.c, main/channel.c, + channels/chan_dahdi.c, apps/app_waitforring.c, + channels/sig_analog.c: Multiple revisions 375993-375994 ........ + r375993 | mmichelson | 2012-11-07 11:01:13 -0600 (Wed, 07 Nov + 2012) | 30 lines Fix misuses of timeouts throughout the code. + Prior to this change, a common method for determining if a + timeout was reached was to call a function such as + ast_waitfor_n() and inspect the out parameter that told how many + milliseconds were left, then use that as the input to + ast_waitfor_n() on the next go-around. The problem with this is + that in some cases, submillisecond timeouts can occur, resulting + in the out parameter not decreasing any. When this happens + thousands of times, the result is that the timeout takes much + longer than intended to be reached. As an example, I had a + situation where a 3 second timeout took multiple days to finally + end since most wakeups from ast_waitfor_n() were under a + millisecond. This patch seeks to fix this pattern throughout the + code. Now we log the time when an operation began and find the + difference in wall clock time between now and when the event + started. This means that sub-millisecond timeouts now cannot play + havoc when trying to determine if something has timed out. Part + of this fix also includes changing the function ast_waitfor() so + that it is possible for it to return less than zero when a + negative timeout is given to it. This makes it actually possible + to detect errors in ast_waitfor() when there is no timeout. + (closes issue ASTERISK-20414) reported by David M. Lee Review: + https://reviewboard.asterisk.org/r/2135/ ........ r375994 | + mmichelson | 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) | 3 + lines Remove some debugging that accidentally made it in the last + commit. ........ Merged revisions 375993-375994 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375995 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-06 18:59 +0000 [r375966] Richard Mudgett + + * include/asterisk/features.h, main/channel.c, /, + main/channel_internal_api.c, main/features.c, + include/asterisk/channel.h: Fix stuck DTMF when bridge is broken. + When a bridge is broken by an AMI Redirect action or the + ChannelRedirect application, an in progress DTMF digit could be + stuck sending forever. * Made simulate a DTMF end event when a + bridge is broken and a DTMF digit was in progress. (closes issue + ASTERISK-20492) Reported by: Jeremiah Gowdy Patches: + bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by + Jeremiah Gowdy Modified to jira_asterisk_20492_v1.8.patch + jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by + rmudgett Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/2169/ ........ Merged + revisions 375964 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375965 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-12-10 Asterisk Development Team + + * Asterisk 11.1.0 Released. + +2012-12-06 Asterisk Development Team + + * Asterisk 11.1.0-rc3 Released. + + * chan_local: Fix local_pvt ref leak in local_devicestate(). + + Regression introduced by ASTERISK-20390 fix. + + (closes issue ASTERISK-20769) + Reported by: rmudgett + +2012-12-05 Asterisk Development Team + + * Asterisk 11.1.0-rc2 Released. + + * Fix a SIP request memory leak with TLS connections. + + During the TLS re-work in chan_sip some TLS specific code was moved + into a separate function. This function operates on a copy of the + incoming SIP request. This copy was never deinitialized causing a + memory leak for each request processed. + + This function is now given a SIP request structure which it can use + to copy the incoming request into. This reduces the amount of memory + allocations done since the internal allocated components are reused + between packets and also ensures the SIP request structure is + deinitialized when the TLS connection is torn down. + + (closes issue ASTERISK-20763) + Reported by: deti + +2012-11-06 Asterisk Development Team + + * Asterisk 11.1.0-rc1 Released. + +2012-11-06 12:09 +0000 [r375925] Joshua Colp + + * channels/chan_motif.c: Fix a bug where our Motif ICE candidates + were not quite proper, and make us more forgiving. An issue was + reported on the mailing list where calling would result in an + "Incomplete ICE-UDP candidate received on session" error message. + This is the result of the ICE-UDP candidate code not placing a + "network" attribute within the candidates. This is now done. To + increase compatibility though I have removed the requirement for + the "network" attribute to exist within ICE-UDP candidates that + are received since we don't actually require the value. Reported + on the mailing list by Jean-Denis Girard. + +2012-11-05 23:09 +0000 [r375895] Matthew Jordan + + * main/timing.c, main/channel.c, /, res/res_timing_pthread.c, + res/res_timing_dahdi.c, res/res_timing_timerfd.c, + bridges/bridge_softmix.c, funcs/func_jitterbuffer.c, + include/asterisk/timing.h, res/res_musiconhold.c, + channels/chan_iax2.c, res/res_fax_spandsp.c, + res/res_timing_kqueue.c: Refactor ast_timer_ack to return an + error and handle the error in timer users Currently, if an + acknowledgement of a timer fails Asterisk will not realize that a + serious error occurred and will continue attempting to use the + timer's file descriptor. This can lead to situations where errors + stream to the CLI/log file. This consumes significant resources, + masks the actual problem that occurred (whatever caused the timer + to fail in the first place), and can leave channels in odd + states. This patch propagates the errors in the timing resource + modules up through the timer core, and makes users of these + timers handle acknowledgement failures. It also adds some + defensive coding around the use of timers to prevent using bad + file descriptors in off nominal code paths. Note that the patch + created by the issue reporter was modified slightly for this + commit and backported to 1.8, as it was originally written for + Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/ + (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches: + jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license + 6358) ........ Merged revisions 375893 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375894 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-05 21:41 +0000 [r375864] Richard Mudgett + + * main/loader.c, /: Add safety NULL pointer check in module user + references. Made __ast_module_user_remove() check for NULL + pointers. ........ Merged revision 375860 from C.3 ........ + Merged revisions 375862 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375863 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-05 17:59 +0000 [r375847] Jonathan Rose + + * /, UPGRADE.txt: chan_sip: Document a change to user-field + encoding introduced with r303509 The change in question was added + to improve compliance with RFC3261, but at the time of commit, it + wasn't adequately documented in the UPGRADE notes. (closes issue + ASTERISK-20561) Reported by: Deniz Review: + https://reviewboard.asterisk.org/r/2177/ ........ Merged + revisions 375846 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-04 03:09 +0000 [r375729-375802] Matthew Jordan + + * main/manager.c, /: Don't attempt to purge sessions when no + sessions exist Manager's tcp/tls objects have a periodic function + that purge old manager sessions periodically. During shutdown, + the underlying container holding those sessions can be disposed + of and set to NULL before the tcp/tls periodic function is + stopped. If the periodic function fires, it will attempt to + iterate over a NULL container. This patch checks for whether or + not the sessions container exists before attempting to purge + sessions out of it. If the sessions container is NULL, we simply + return. Note that this error was also caught by the Asterisk Test + Suite. ........ Merged revisions 375800 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375801 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, res/res_fax.c: Only deref a reserved gateway session if we + actually reserved one Its perfectly acceptable to have a gateway + session unreserved when we go to first allocate one. Unreffing + the reserved gateway session - when its NULL - will result in an + assertion error. This problem was caught by the Asterisk Test + Suite (once we had enough of the debugging flags enabled) + ........ Merged revisions 375797 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/manager.c, /: Properly clean up manager resources on exit + This patch does two things: 1) It properly unregisters the + manager CLI commands 2) It cleans up AMI users on exit. Prior to + this patch, the AMI users were not being disposed of properly, + resulting in a memory leak. (closes issue ASTERISK-20646) + Reported by: Corey Farrell patches: manager_shutdown.patch + uploaded by Corey Farrell (license 5909) ........ Merged + revisions 375793 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375794 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/db.c, /: Properly finalize prepared SQLite3 statements to + prevent memory leak The AstDB uses prepared SQLite3 statements to + retrieve data from the SQLite3 database. These statements should + be finalized during Asterisk shutdown so that the SQLite3 + database can be properly closed. Failure to finalize the + statements results in a memory leak and a failure when closing + the database. This patch fixes those issues by ensuring that all + prepared statements are properly finalized at shutdown. (closes + issue ASTERISK-20647) Reported by: Corey Farrell patches: + astdb-sqlite3_close.patch uploaded by Corey Farrell (license + 5909) ........ Merged revisions 375761 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/xmldoc.c: Fix memory leaks in XML documentation This patch + fixes two memory leaks: 1) When building XML documentation items, + the 'name' attribute was extracted from XML elements but not + properly freed after being copied into the item being built. 2) + When unloading XML documentation, the doctree container objects + were not properly freed. This patch corrects these memory leaks. + Note that this patch was modified slightly for this commmit, as + the case where the 'name' attribute doesn't exist also wasn't + handled in the item construction. This patch also checks for that + attribute not existing. (closes issue ASTERISK-20648) Reported + by: Corey Farrell Tested by: mjordan patches: + xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909) + + * main/cdr.c, /: Prevent multiple CDR batches from conflicting when + scheduling the CDR write The Asterisk Test Suite caught an error + condition where a scheduled CDR batch write can be deleted twice + if two channels attempt to post their CDRs at the same time. The + batch CDR mutex is locked while the CDRs are appended to the + current batch list; however, it is unlocked prior to actually + scheduling the CDR write. As such, two threads can attempt to + remove the currently scheduled batch write at the same time, + resulting in an assertion error. This patch extends the time that + the mutex is locked to encompass actually scheduling the write. + This prevents two threads from unscheduling the currently + scheduled write at the same time. ........ Merged revisions + 375727 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 375728 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-03 03:17 +0000 [r375702] Andrew Latham + + * README, include/asterisk/doxyref.h: Doxygen Updates Replace links + to missing text files removed in the 1.6.x series with links to + the wiki. Doxygen can handle URLs fine, don't atempt to quote + them. Also update the wiki link in the Readme to get everyone on + the same page. (issue ASTERISK-20259) ........ Merged revisions + 375698 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 375699 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-02 20:59 +0000 [r375661] Richard Mudgett + + * main/channel.c, channels/chan_misdn.c, /, main/ccss.c, + main/format_pref.c: Things don't need to be that const. ........ + Merged revisions 375658 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375659 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-02 20:56 +0000 [r375660] Damien Wedhorn + + * channels/chan_skinny.c: Fix for chan_skinny leaving RTP ports + open Skinny wasn't closing RTP sockets. This patch includes + ast_rtp_instance_stop before ast_rtp_instance_destroy which fixes + the problem. Also add destroy for VRTP (which I believe is + unused, but exists). Review: + https://reviewboard.asterisk.org/r/2176/ + +2012-11-02 18:44 +0000 [r375627] Richard Mudgett + + * channels/misdn/isdn_lib.h, /, channels/misdn/isdn_lib.c: Multiple + revisions 375519-375524 ........ r375519 | rmudgett | 2012-10-30 + 16:06:15 -0500 (Tue, 30 Oct 2012) | 11 lines chan_misdn: Timer + primitives must be handled first. The frm->addr is a different + "address space" than the stack/instance address of other Lx + primitives. The test for B channel instance address could fail. + Patches: patch01_timers.diff (license #6372) patch uploaded by + Guenther Kelleter JIRA ABE-2888 ........ r375520 | rmudgett | + 2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) | 10 lines + chan_misdn: Free memory in error paths and other memory leaks. + The one line commented with BUG is not easily fixable because + there is no de-init function one can call. Patches: + patch02_memory.diff (license #6372) patch uploaded by Guenther + Kelleter JIRA ABE-2888 ........ r375521 | rmudgett | 2012-10-30 + 16:38:41 -0500 (Tue, 30 Oct 2012) | 14 lines chan_misdn: ISDN NT + L2 de-establish/establish * An NT-PTMP cannot de/establish L2 + since it doesn't know the TEIs. * On NT-PTP L2 is started when L1 + is finally active in handle_l1. * L2 deactivation logging + cleanup. * L2 aggregate link status is unknown for NT-PTMP, show + as "UNKN". * Removed unused functions and code for L2 handling. + Patches: patch03_L2estab.diff (license #6372) patch uploaded by + Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 | + rmudgett | 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) | 22 + lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH + prim via lower_id layer (3 or 1) simply does not work. For TE (3) + it returns an error (len=-6) which is not evaluated by + handle_l1(), so the L1 layer status ends up wrong. Instead PH + must be sent via L4, only then does it reach L1 without an error + message. And NT PH prims only reach L1 when they are sent to + layer 2 id. --> use upper_id to send PH primitives. * Check for + errors in PH_(DE)ACTIVATE | CONFIRM. * Debug messages are + improved. * The lower_id is now not used for anything, except: + Why is lower_id layer deleted when it wasn't created? I removed + this code since it looks very wrong. Patches: + patch04_l1activation.diff (license #6372) patch uploaded by + Guenther Kelleter JIRA ABE-2888 ........ r375523 | rmudgett | + 2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) | 31 lines + chan_misdn: Fix loss of B channels if L1 is down. If you make 2 + calls out an NT PTMP port which is not connected to any phone, + the B channel associated with that call becomes unusable until + Asterisk is restarted. The problem is the EVENT_SETUP is queued + when L1 is not up in misdn_lib_send_event(). If L1 cannot be + activated the event won't be dequeued. It gets even worse when + the call is hung up. The queued EVENT_SETUP will be overwritten + by an EVENT_DISCONNECT. The reserved B channel then will never be + freed. If later someone connects a phone to the port, L1 will + eventually activate and the queued EVENT_DISCONNECT is sent down + the stack. However, it is ignored because it is the wrong call + state. The real fix would be that activation and queueing for a + new SETUP is done by the NT stack. But since it doesn't, the + workaround must be removed because it doesn't always work. Fix: + The event is no longer queued but immediately sent to the stack. + If L1 cannot be activated, the L3 state machine that was started + by the EVENT_SETUP will do its work, i.e. a timeout will release + the B channel properly. The SETUP possibly cannot be sent the + first time but is resent by T303 in case L1 could be activated. + Patches: patch05_bchan-loss.diff (license #6372) patch uploaded + by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 | + rmudgett | 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) | 13 + lines chan_misdn: Remove some calls to exit(). Try proper cleanup + when something goes wrong in misdn_lib_init(). Especially do not + call exit()! * Fix memory leak because stack_destroy() does not + free the stack struct. Patches: patch06_cleanup-init.diff + (license #6372) patch uploaded by Guenther Kelleter Modified JIRA + ABE-2888 ........ Merged revisions 375519-375524 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + ........ Merged revisions 375625 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375626 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-02 17:24 +0000 [r375613] Michael L. Young + + * /, channels/chan_sip.c: Fix Wrong Result In Debug Message For SDP + Origin Processing While looking at some debug logs, I noticed + that it was being reported that the SDP origin line was + unsupported or failed. Upon looking into this on my local + machine, I found that I too was getting this debug message yet + everything seemed to be getting processed properly. What was + discovered is, that, the variable to determine what is displayed + in the debug message for the SDP line that was processed, was not + being set for the origin line when the result was successful. + This patch fixes this and was tested on local machine. ........ + Merged revisions 375594 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375601 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-11-01 14:52 +0000 [r375575] Jonathan Rose + + * channels/chan_sip.c, configs/sip.conf.sample: chan_sip: Fix a bug + causing SIP reloads to remove all entries from the registry A + regression was introduced in chan_sip by changes to sip reload + introduced by r349097. That patch moved peer purging from the + beginning of the reload to after the general configuration was + finished. This patch fixes that by undoing the repositioning of + the original peer purging code and using a similar function after + performing general configuration that purges only autocreated + peers that were created when persist mode isn't enabled. (closes + issue ASTERISK-20611) Reported by: Alisher Review: + https://reviewboard.asterisk.org/r/2171/ + +2012-10-31 18:00 +0000 [r375559] Joshua Colp + + * res/res_http_websocket.exports.in: Fix an issue with + res_http_websocket where the chan_sip WebSocket handler could not + be registered. On some systems the optional API support uses the + GCC compiler attribute "weakref" to provide its functionality. + This code changes the function names and prefixes "__" to the + front. The res_http_websocket exports file did not take this into + account, thereby not allowing those functions to be global and + ultimately found. (closes issue ASTERISK-20631) Reported by: + danjenkins + +2012-10-31 14:49 +0000 [r375532] Matthew Jordan + + * res/res_calendar_ews.c, /: Properly extract the Body information + of an EWS calendar item Unlike all other calendar modules, + res_calendar_ews fails to extract the Body information for a + calendar item. This is due, in part, to a quirk in the schema in + the XML - not only does a CalendarItem contain a Body element, + but the CalendarItem exists as a descendant of a different Body + element. The neon parser was erroneously skipping all Body + elements. This patch fixes that by bypassing Body elements that + are not a child of CalendarItem, and parsing the Body element out + if it is a child. Note that the original patch by Terry Wilson + only needed slight modifications to make it properly pull the + Body information out; as such, while I've linked to the patch + that I uploaded for Dmitry, I've attributed the patch to Terry. + (closes issue ASTERISK-19738) Reported by: Dmitry Burilov Tested + by: Dmitry Burilov patches: calendar_ews_body_2012_10_29.diff + uploaded by Terry Wilson (license 6283) ........ Merged revisions + 375528 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 375531 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-30 19:23 +0000 [r375506] Richard Mudgett + + * /, bridges/bridge_softmix.c: Fix ConfBridge crash if no timing + module loaded. (closes issue ASTERISK-19448) Reported by: feyfre + Patches: smfix.patch (license #6099) patch uploaded by feyfre + Modified for coding guidelines. ........ Merged revisions 375496 + from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-30 19:09 +0000 [r375471-375486] Jonathan Rose + + * /, apps/app_mixmonitor.c: mixmonitor: Add a test event This test + event is being used to fix the mixmonitor_audiohook_inherit test. + ........ Merged revisions 375484 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375485 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_confbridge.c: confbridge: Fix a bug which made + conferences not record with AMI/CLI commands When confbridge was + changed to handle conference status with a state machine in + r374658. The function responsible for starting recording for a + conference was refactored with the function actually responsible + for launching the recording thread being split into a function + with another name. The old function name was still used for + manually started recordings through AMI or CLI. This patch fixes + that by switching which function is used to start recording the + conference. (closes issue ASTERISK-20601) Reported by: Vilius + Patches: confbridge_mixmonitor.diff uploaded by Jonathan Rose + (license 6182) ........ Merged revisions 375470 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-30 02:22 +0000 [r375469] Matthew Jordan + + * /, apps/app_queue.c: Ensure that the Queue application tracks + busy members in off nominal situations There are a few code paths + where the Queue application fails to count a paused or in use + queue member as being 'busy'. This can cause callers to get stuck + in the Queue until a paused agent unpauses themselves. (closes + issue ASTERISK-20623) Reported by: Bryan Walters patches: + app_queue.patch uploaded by Bryan Walters (license 5851) ........ + Merged revisions 375450 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375451 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-29 21:23 +0000 [r375437] Mark Michelson + + * /, channels/chan_sip.c: Prevent resetting of NATted realtime peer + address on reload. If a "sip reload" is issued for a SIP peer, + then his IP address will be cleared, thus resulting in forgetting + the public IP address. Asterisk will then attempt to route SIP + traffic to the private IP address. The fix here is to make "sip + reload" ignore realtime peers when "host = dynamic" is spotted. + Realtime peers can now only have their IP address reset if they + have gone from being not dynamic to being dynamic. (closes issue + ASTERISK-18203) reported by daren ferreira (closes issue + ASTERISK-20572) reported by JoshE Patches: fix_nat_realtime.diff + uploaded by JoshE (license #6075) ........ Merged revisions + 375415 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 375417 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-29 19:29 +0000 [r375363-375390] Richard Mudgett + + * /, main/features.c: Fix the Park 'r' option when a channel parks + itself. When a channel uses the Park appliation to park itself + with the 'r' option, the channel hears music-on-hold instead of + the requested ringing. * Added a missing check for the 'r' option + when a channel parks itself. (closes issue ASTERISK-19382) + Reported by: James Stocks Patches by: dsessions Review: + https://reviewboard.asterisk.org/r/2148/ ........ Merged + revisions 375388 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375389 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_dahdi.c, /: chan_dahdi: Fix segfault dereferencing + a NULL tech_pvt. The tech support customer was using the AMI + Redirect action shortly after a call was placed. While the + channel tried to do an ast_read(), the masquerade resulting from + the channel redirect took place. The masquerade in the middle of + the ast_read() resulted in the segfault. (closes issue AST-1025) + Reported by: Trey Blancher Patches: jira_ast_1025_v1.8_v2.patch + (license #5621) patch uploaded by rmudgett ........ Merged + revisions 375361 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375362 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-23 16:22 +0000 [r375288-375327] Jonathan Rose + + * contrib/scripts/ast_tls_cert, /: ast_tls_cert script: Better + response for various exit conditions to openssl (closes issue + ASTERISK-20260) Reported by: Daniel O'Connor Patches: + ast_tls_cert-update.diff uploaded by Daniel O'Connor (license + 6419) ........ Merged revisions 375325 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375326 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/app.c: core: Fix a memory leak in app.c from an early + return ast_app_group_match_get_count allocates memory with the + regcomp function and we previously forgot to free it when bailing + out due to a regex compilation failure against category. (closes + issue AST-1018) Reported by: Guenther Kelleter Patches: + regcomp_memleak.diff uploaded by Guenther Kelleter (license 6372) + ........ Merged revisions 375299 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375300 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, codecs/gsm/src/code.c: GSM: Fix encoding problems with GSM + (closes issue ASTERISK-20457) Reported by: Richard Miller + Patches: code.patch uploaded by Richard Miller (license 5685) + ........ Merged revisions 375272 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375273 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-18 21:44 +0000 [r375219-375247] Jonathan Rose + + * UPGRADE.txt: app_queue: add upgrade notes for 375216 Adds UPGRADE + notes describing behavioral changes to rrmemory strategy caused + by 375216 (issue AST-989) Reported by: Thomas Arimont + + * /, apps/app_queue.c: app_queue: Make ordering of + rrmemory/rrordered persist over add/remove members Prior to this + patch, adding, removing or reloading members to rrmemory would + cause the order to become completely jumbled. Now it behaves more + or less like rrordered other than the fact that it stores the + members on a hash table rather than a linked list. This patch + also prevents removal of members and member reloads from jumbling + rrordered queues. (issue AST-989) Reported by: Thomas Arimont + Review: https://reviewboard.asterisk.org/r/2164/ ........ Merged + revisions 375216 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375217 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-18 20:02 +0000 [r375191] Richard Mudgett + + * Makefile, /, build_tools/make_version, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: + build_tools: Allow Asterisk to report git SHAs in version string. + Make git more attractive for managing work-in-progress. + Especially convenient when a potential patch set needs to be + tested on multiple platforms since one can use git to keep all + the test environments in sync independent of a subversion server. + Now the Asterisk version will show the exact git SHA5 that was + used when building (still appended by "M" if there are local + modifications) from a git clone of the Asterisk repository so the + developer can more easily know what is actually under test. You + will now get this: $ asterisk -V Asterisk GIT-1698298 Instead of + this: $ asterisk -V Asterisk UNKNOWN__and_probably_unsupported + This has zero impact for those not using git with the exception + of an extra test in the configure script to gather git's path. + This is necessary to prevent "sudo make install" from failing + since git may not be in the path in make's shell environment. + (closes issue ASTERISK-20483) Reported by: Shaun Ruffell Patches: + 0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch + (license #5417) patch uploaded by Shaun Ruffell Modified ........ + Merged revisions 375189 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375190 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-17 19:00 +0000 [r375148] Kinsey Moore + + * main/tcptls.c, /: Ensure Asterisk fails TCP/TLS SIP calls when + certificate checking fails When placing a call to a TCP/TLS SIP + endpoint whose certificate is not signed by a configured CA + certificate, Asterisk would issue a warning and continue to + process the call as if there was not an issue with the + certificate. Asterisk now properly fails the call if the + certificate fails verification or if the certificate does not + exist when certificate checking is enabled (the default + behavior). (closes issue ASTERISK-20559) Reported by: kmoore + Review: https://reviewboard.asterisk.org/r/2163/ ........ Merged + revisions 375146 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375147 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-16 21:44 +0000 [r375079-375113] Walter Doekes + + * /, channels/chan_sip.c: Fixes to the fd-oriented SIP TCP reads. + Don't crash on large user input. Allow SIP headers without space. + Optimize code a bit. Review: + https://reviewboard.asterisk.org/r/2162 ........ Merged revisions + 375111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 375112 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Update sip_request_call SIP dial string + documentation. This was missed when merging review r1859. + ........ Merged revisions 375074 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375078 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-16 14:08 +0000 [r375051] Joshua Colp + + * channels/chan_iax2.c: Remove a log message that was left in + accidentally from call-id logging development. + +2012-10-15 21:15 +0000 [r375027] Mark Michelson + + * apps/app_dial.c, /, main/ccss.c, include/asterisk/strings.h, + channels/chan_iax2.c: Fix some potential misuses of ast_str in + the code. Passing an ast_str pointer by value that then calls + ast_str_set(), ast_str_set_va(), ast_str_append(), or + ast_str_append_va() can result in the pointer originally passed + by value being invalidated if the ast_str had to be reallocated. + This fixes places in the code that do this. Only the example in + ccss.c could result in pointer invalidation though since the + other cases use a stack-allocated ast_str and cannot be + reallocated. I've also updated the doxygen in strings.h to + include notes about potential misuse of the functions mentioned + previously. Review: https://reviewboard.asterisk.org/r/2161 + ........ Merged revisions 375025 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 375026 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-15 08:11 +0000 [r375016] Igor Goncharovskiy + + * channels/chan_unistim.c: Fix underscreen buttons warnings apeared + while transfer process + +2012-10-14 11:57 +0000 [r374995] Tzafrir Cohen + + * config.guess, config.sub, /: Update config.guess and config.sub: + 2012-10-10 Update config.guess and config.sub to revision + fb456b34ef4aa02b95dc6be69aaa66fa94a844fb from the + savannah.gnu.org git repo. Adds support for e.g. aarch64 (ARM + 64bit). config.guess:timestamp='2012-09-25' + config.sub:timestamp='2012-10-10' ........ Merged revisions + 374977 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 374991 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-12 21:57 +0000 [r374932] Kinsey Moore + + * apps/app_voicemail.c: Avoid a segfault on invalid format names If + a format name was not found by ast_getformatbyname, a NULL + pointer would be passed into ast_format_rate and immediately + dereferenced. This ensures that a valid pointer is used since the + structure is already allocated on the stack. (closes issue + DPH-523) Reported-by: Steve Pitts + +2012-10-12 16:20 +0000 [r374914] Mark Michelson + + * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h: + Do not use a FILE handle when doing SIP TCP reads. This is used + to solve an issue where a poll on a file descriptor does not + necessarily correspond to the readiness of a FILE handle to be + read. This change makes it so that for TCP connections, we do a + recv() on the file descriptor instead. Because TCP does not + guarantee that an entire message or even just one single message + will arrive during a read, a loop has been introduced to ensure + that we only attempt to handle a single message at a time. The + tcptls_session_instance structure has also had an overflow buffer + added to it so that if more than one TCP message arrives in one + go, there is a place to throw the excess. Huge thanks goes out to + Walter Doekes for doing extensive review on this change and + finding edge cases where code could fail. (closes issue + ASTERISK-20212) reported by Phil Ciccone Review: + https://reviewboard.asterisk.org/r/2123 ........ Merged revisions + 374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 374906 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-11 21:18 +0000 [r374850-374877] Joshua Colp + + * channels/chan_motif.c: Fix a bug where audio on Google Voice + would not work due to ignoring candidates. Instead of ignoring + parts of the message that are not known just ignore the ones we + know may be present and that would cause a problem. + + * res/res_rtp_asterisk.c: Remove code that should not have gotten + in. (issue ASTERISK-20554) + + * res/res_rtp_asterisk.c, channels/chan_motif.c: Fix an issue where + outgoing calls would fail to establish audio due to ICE + negotiation failures. This change removes the requirement for + ufrag and pwd in the transport stanza and also makes us the + controlling agent. (closes issue ASTERISK-20554) Reported by: + mmichelson + +2012-10-11 15:44 +0000 [r374845] Matthew Jordan + + * main/cdr.c, /: Fix incorrect billing duration reported when batch + mode is enabled Similar to r369351, the billing duration can be + skewed when batch mode is enabled. This happened much more rarely + than the duration, as it only occured when the call was answered + (thereby indicating an actual answer time) and immediately hung + up on (indicating a billsec of 0). Since a billing time of '0' + can either mean that the call immediately ended or that the CDR + was improperly answered, we have to use additional information to + know whether or not we can trust the CDR billsec value. Prior to + this patch, we looked to see if we had a valid answer time. If we + did, and billsec was zero, we used the current time to calculate + what billsec value we could from the CDR being written. If batch + mode is enabled, this will incorrectly report a billsec value + being much greater than the actual duration of the call. Instead + of relying on the presence of an answer time to know whether or + not we can re-calculate the billsec for the CDR, we now also use + the presence of the CDR's end time to know if we need to + re-calculate or whether we can trust the billsec value that we + have. This prevents erroneous jumps in the billsec value, while + still making sure that in the worst case, some billing time will + be calculated. (closes issue AST-1016) Reported by: Thomas + Arimont Tested by: Thomas Arimont ........ Merged revisions + 374843 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 374844 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-11 15:31 +0000 [r374842] Mark Michelson + + * channels/chan_sip.c, include/asterisk/sip_api.h, + channels/chan_sip.exports.in (removed), main/sip_api.c (added): + Don't make chan_sip export global symbols. During testing, it was + discovered that having chan_sip export global symbols was + problematic. The biggest problem was that load order was + affected. Trying to use realtime could be problematic since in + all likelihood the necessary realtime driver(s) would not be + loaded before chan_sip. In addition, it was found that it was + impossible to use the Digium Phone Module for Asterisk since it + must be loaded before chan_sip since it must hook into chan_sip's + configuration parsing. The solution is to use a virtual table in + the same manner that other modules in Asterisk do, like + app_voicemail. (closes issue ASTERISK-20545) Reported by: kmoore + +2012-10-11 13:33 +0000 [r374833] Joshua Colp + + * channels/chan_motif.c: Consider the Google Talk content stanza + name (jin:content) valid. + +2012-10-10 21:03 +0000 [r374804] Richard Mudgett + + * /, apps/app_queue.c: app_queue: Made pass connected line updates + from the caller to ringing queue members. Party A calls Party B + Party B puts Party A on hold. Party B calls a queue. Ringing + queue member D sees Party B identification. Party B transfers + Party A to the queue. Queue member D does not get a connected + line update for Party A. Queue member D answers the call and + still sees Party B information. However, if Party A later + transfers the call to Party C then queue member D gets a + connected line update for Party C. * Made pass connected line + updates from the caller to queue members while the queue members + are ringing. (closes issue AST-1017) Reported by: Thomas Arimont + (closes issue ABE-2886) Reported by: Thomas Arimont Tested by: + rmudgett ........ Merged revisions 374801 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + ........ Merged revisions 374802 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374803 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-10 13:35 +0000 [r374792] Kinsey Moore + + * main/manager.c: Fix segfault regression from r370681 Due to usage + of ast_hook_send_action, AMI action handling code should be able + to handle a NULL mansession->session. This would cause a crash on + NULL dereference if action_originate was called from + ast_hook_send_action. (closes issue ASTERISK-20544) + +2012-10-09 22:21 +0000 [r374771] Richard Mudgett + + * main/pbx.c, /: Fix execution of 'i' extension due to + uninitialized variable. The fix for ASTERISK-18243 added code + that could potentially use dst_exten[] uninitialized. As a result + the 'i' exten may not be executed when it should. (closes issue + ASTERISK-20455) Reported by: Richard Miller Patches: + pbx-1.8.16.0.diff (license #5685) patch uploaded by Richard + Miller Made some cosmetic modifications. ........ Merged + revisions 374758 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374763 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-09 21:34 +0000 [r374755-374756] Joshua Colp + + * channels/chan_sip.c: Improve logging for DTLS-SRTP failure + situations. (closes issue ASTERISK-20487) Reported by: mjordan + + * channels/chan_sip.c: Add a log message for when DTLS-SRTP is + requested and the underlying engine does not support it. (closes + issue ASTERISK-20487) Reported by: mjordan + +2012-10-08 22:30 +0000 [r374708-374729] Richard Mudgett + + * configs/chan_dahdi.conf.sample, /: dahdi.conf.sample: Add + description for "buffers" setting. This contains an edited + version of the patch originally created by John Bigelow. (closes + issue ASTERISK-14435) Reported by: John Bigelow Patches: + buffers.patch (license #5091) patch uploaded by John Bigelow + 0001-dahdi.conf.sample-Add-description-for-buffers-settin.patch + (license #5417) patch uploaded by Shaun Ruffell Modified ........ + Merged revisions 374727 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374728 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * pbx/pbx_spool.c, /: Fix deletion of unopenable spool files. If + scan_service() cannot open the spool file, it logs a message + saying that it will delete the file and calls remove_from_queue() + to do it. However, remove_from_queue() fails to delete the spool + file because struct outgoing has not yet been fully initialized. + * Merged allocating a new struct outgoing and init_outgoing() + into new_outgoing(). Allocation is initialization. * Made + apply_outgoing() not initialize the spool filename in struct + outgoing. * Made apply_outgoing() call ast_trim_blanks() and + ast_skip_blanks() rather than manually inlining them. * Reduced + indentation levels in apply_outgoing(). * Fixed a garbled comment + in remove_from_queue(). * Reworked scan_service() to simplify it. + (closes issue ASTERISK-17231) Reported by: David Chappell + Patches: spool_open_failure.diff (license #4997) patch uploaded + by David Chappell Started with this patch. ........ Merged + revisions 374686 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 * Fixed some + memory leaks on off nominal paths in init_outgoing() when merging + into the new_outgoing() function dealing with o->capabilities. + ........ Merged revisions 374695 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-25 Asterisk Development Team + + * Asterisk 11.0.0 Released. + +2012-10-17 Asterisk Development Team + + * Asterisk 11.0.0-rc2 Released. + + * [r374792] Fix segfault regression from r370681 + + Due to usage of ast_hook_send_action, AMI action handling code should + be able to handle a NULL mansession->session. This would cause a + crash on NULL dereference if action_originate was called from + ast_hook_send_action. + + (closes issue ASTERISK-20544) + + * [r374842] Don't make chan_sip export global symbols. + + During testing, it was discovered that having chan_sip export global + symbols was problematic. + + The biggest problem was that load order was affected. + Trying to use realtime could be problematic since in + all likelihood the necessary realtime driver(s) would + not be loaded before chan_sip. + + In addition, it was found that it was impossible to + use the Digium Phone Module for Asterisk since it + must be loaded before chan_sip since it must hook + into chan_sip's configuration parsing. + + The solution is to use a virtual table in the same + manner that other modules in Asterisk do, like + app_voicemail. + + (closes issue ASTERISK-20545) + Reported by: kmoore + + * [r374850] Fix an issue where outgoing calls would fail to establish + audio due to ICE negotiation failures. + + This change removes the requirement for ufrag and pwd in the transport + stanza and also makes us the controlling agent. + + (closes issue ASTERISK-20554) + Reported by: mmichelson + + * [r374851] Remove code that should not have gotten in (r374850) + + (issue ASTERISK-20554) + + * [r374877] Fix a bug where audio on Google Voice would not work due to + ignoring candidates. + + Instead of ignoring parts of the message that are not known just + ignore the ones we know may be present and that would cause a problem. + + * [r375148] Ensure Asterisk fails TCP/TLS SIP calls when certificate + checking fails + + When placing a call to a TCP/TLS SIP endpoint whose certificate is not + signed by a configured CA certificate, Asterisk would issue a warning + and continue to process the call as if there was not an issue with the + certificate. Asterisk now properly fails the call if the certificate + fails verification or if the certificate does not exist when + certificate checking is enabled (the default behavior). + + (closes issue ASTERISK-20559) + Review: https://reviewboard.asterisk.org/r/2163/ + + * [r375051] Remove a log message that was left in accidentally from + call-id logging development. + +2012-10-08 Asterisk Development Team + + * Asterisk 11.0.0-rc1 Released. + +2012-10-08 20:38 +0000 [r374632-374676] Matthew Jordan + + * res/res_rtp_asterisk.c, configs/rtp.conf.sample: Disable ICE + support by default Since there are a number of legacy devices out + there that fail to handle ICE candidates properly (which is a + nice way of saying something much uglier), disable it by default. + Support for ICE candidates can be enabled in rtp.conf using the + icesupport setting. + + * apps/confbridge/conf_state.c (added), + apps/confbridge/conf_state_single.c (added), + apps/confbridge/conf_state_inactive.c (added), + apps/confbridge/conf_state_single_marked.c (added), /, + apps/confbridge/include/confbridge.h, + apps/confbridge/include/conf_state.h (added), + apps/confbridge/conf_state_multi.c (added), + apps/app_confbridge.c, apps/confbridge/conf_state_multi_marked.c + (added), apps/confbridge/conf_state_empty.c (added): Resolve + issues in ConfBridge regarding marked, waitmarked, and unmarked + users Thank's to Neil Tallim (flan)'s tireless testing, issue + reporting, and patches it became clear that app_confbridge had + some complex logic in how it handled interactions between marked, + waitmarked, and unmarked users. In particular, there were some + areas in which the interactions between the users resulted in + inconsistent behavior, and app_confbridge was missing logic in + how to handle some corner cases. Some areas included: * Poor + handling of mixing unmarked and waitmarked users * + Inconsistencies in how MOH and muting was applied to various + users * Handling of various announcements for different user + profile options flan's patches seem to fix the various issues, + but highlighted how hard the code could be to maintain. In an + attempt to make things easier to maintain and to more fully + enumerate the various cases that exist, this patch breaks up the + logic into a state machine-like setup. Please note that the + various state transitioned are documented on the Asterisk wiki: + https://wiki.asterisk.org/wiki/display/AST/Confbridge+state+changes + Review: //https://reviewboard.asterisk.org/r/2072/ Note that for + the following issues, mjordan uploaded the patch, although it was + written by twilson. Any contributor license discrepency is due to + that. (closes issue ASTERISK-19562) Reported by: flan Tested by: + flan, mjordan, jrose patches: + bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by + twilson (license 6283) (closes issue ASTERISK-19726) Reported by: + flan Tested by: flan patches: + bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by + twilson (license 6283) (closes issue ASTERISK-20181) Reported by: + Jonathan White Tested by: Jonathan White patches: + bugASTERISK-19562_ASTERISK-19726_ASTERISK-20181.patch uploaded by + twilson (license 6283) ........ Merged revisions 374652 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * res/pjproject/pjlib/include/pj/sock.h, + res/pjproject/pjlib/src/pj/sock_symbian.cpp, + res/pjproject/pjlib/src/pj/sock_bsd.c, + res/pjproject/pjlib/src/pj/sock_linux_kernel.c: pjproject: Fix + for Solaris builds. Do not undef s_addr. pjproject, in order to + solve build problems on Windows [1], undefines s_addr in one of + it's headers that is included in res_rtp_asterisk.c. On Solaris + s_addr is not a structure member, but defined to map to the real + strucuture member, therefore when building on Solaris it's + possible to get build errors like: [CC] res_rtp_asterisk.c -> + res_rtp_asterisk.o In file included from + /export/home/admin/asterisk-11-svn/include/asterisk/stun.h:29, + from res_rtp_asterisk.c:51: + /export/home/admin/asterisk-11-svn/include/asterisk/network.h: In + function `inaddrcmp': + /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: + error: structure has no member named `s_addr' + /export/home/admin/asterisk-11-svn/include/asterisk/network.h:92: + error: structure has no member named `s_addr' res_rtp_asterisk.c: + In function `ast_rtp_on_ice_tx_pkt': res_rtp_asterisk.c:706: + warning: dereferencing type-punned pointer will break + strict-aliasing rules res_rtp_asterisk.c:710: warning: + dereferencing type-punned pointer will break strict-aliasing + rules res_rtp_asterisk.c: In function + `rtp_add_candidates_to_ice': res_rtp_asterisk.c:1085: error: + structure has no member named `s_addr' make[2]: *** + [res_rtp_asterisk.o] Error 1 make[1]: *** [res] Error 2 make[1]: + Leaving directory `/export/home/admin/asterisk-11-svn' gmake: *** + [_cleantest_all] Error 2 Unfortunately, in order to make this + work, I also had to make sure pjproject only used the typdef + pj_in_addr and not the struct pj_in_addr so that when building + Asterisk I could "typedef struct in_addr pj_in_addr". It's + possible then that the library and users of those interfaces in + Asterisk have a different idea about the type of the argument, + while on the surface it looks like they are all 32 bit big endian + values. [1] http://trac.pjsip.org/repos/changeset/484 (issues + ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang, + mjordan patches: + 0001-pjproject-Fix-for-Solaris-builds.-Do-not-undef-s.patch + uploaded by Shaun Ruffell (license 5417) + + * main/acl.c: Trivial patch to make 'best_score' defined for all + architectures. Fixes trivial build error on Solaris: acl.c: In + function `get_local_address': acl.c:196: error: `best_score' + undeclared (first use in this function) acl.c:196: error: (Each + undeclared identifier is reported only once acl.c:196: error: for + each function it appears in.) make[2]: *** [acl.o] Error 1 (issue + ASTERISK-20366) Reported by: Ben Klang Tested by: Ben Klang + patches: + 0002-main-acl.c-Trivial.-best_score-should-be-defined-for.patch + by Shaun Ruffell (license 5417) + +2012-10-06 03:20 +0000 [r374611-374622] Matthew Jordan + + * res/res_xmpp.c: Handle capability stanzas that fail to provide + node or version information While XEP-0115 states that the node + and ver attributes are both required, some devices fail to + provide either field. Prior to this patch, failure to provide the + node or ver attribute would cause a crash in res_xmpp. While + failing to provide the node or ver attribute is technically + invalid, since this information is not utilized by Asterisk + except for reporting purposes, for interoperability reasons, we + continue to process the capability stanza anyways. (closes issue + ASTERISK-20495) Reported by: Martin W Tested by: Martin W + patches: 20495.patch uploaded by Martin W (license #6434) + + * res/res_xmpp.c, main/message.c: Update documentation for + MessageSend application/command's From field for XMPP When using + the channel technology agnostic application/AMI command + MessageSend, the "From" field is technically optional for the SIP + channel driver. However, if being sent by the XMPP resource + module (either res_xmpp or res_jabber), the "From" field is + necessary, and must correspond to a defined account. This patch + updates the documentation for this application/AMI command to + reflect this. (closes issue ASTERISK-20405) Reported by: Leif + Madsen + +2012-10-05 20:32 +0000 [r374587] dlee : + + * main/manager.c, /: Multiple revisions 374570,374581 ........ + r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | + 22 lines Improve AMI long line error handling In AMI's parser, + when it receives a long line (> 1024 characters), it discards + that line, but continues to process the message normally. + Typically, this is not a problem because a) who has lines that + long and b) usually a discarded line results in an invalid + message. But if that line is specifying an optional field, then + the message will be processed, you get a 'Response: Success', but + things don't work the way you expected them to. This patch + changes the behavior when a line-too-long parse error occurs. * + Changes the log message to avoid way-too-long (and truncated + anyways) log messages * Adds a 'parsing' status flag to Response: + Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line + is too long * Responds with an appropriate error if parsing != + MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow + Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581 + | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line + I've committed too much. Reverting part of r374570. ........ + Merged revisions 374570,374581 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374586 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-05 18:34 +0000 [r374538] Richard Mudgett + + * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, + channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: + Merged revisions 374515-374535 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + ................ r374515 | rmudgett | 2012-10-04 17:52:36 -0500 + (Thu, 04 Oct 2012) | 10 lines chan_misdn: Remove some deadcode * + Made setup_bc() static. Patches: patch1_unused-code.diff (license + #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2882 + ................ r374516 | rmudgett | 2012-10-04 18:01:01 -0500 + (Thu, 04 Oct 2012) | 7 lines chan_misdn: Remove unused bchan + states Patches: patch2_unused-states.diff (license #6372) patch + uploaded by Guenther Kelleter JIRA ABE-2882 ................ + r374517 | rmudgett | 2012-10-04 18:17:51 -0500 (Thu, 04 Oct 2012) + | 16 lines chan_misdn: Remove unnecessary null pointer checks and + checks for stack->nt * cleanup_bc() is always called with valid + bc (or it would've crashed before). * Value of stack->nt is known + in advance at some places. * Rename handle_event() to + handle_event_te(), handle_frm() to handle_frm_te(). Patches: + patch3_checks.diff (license #6372) patch uploaded by Guenther + Kelleter Modified JIRA ABE-2882 ................ r374518 | + rmudgett | 2012-10-04 18:21:59 -0500 (Thu, 04 Oct 2012) | 7 lines + chan_misdn: Fix spelling in log messages Patches: + patch4_spelling.diff (license #6372) patch uploaded by Guenther + Kelleter JIRA ABE-2882 ................ r374519 | rmudgett | + 2012-10-04 18:31:59 -0500 (Thu, 04 Oct 2012) | 15 lines + chan_misdn: Don't cleanup a bc twice. In handle_frm_te() after + calling misdn_lib_send_event(bc, EVENT_RELEASE_COMPLETE) bc is + emptied, cleaned and set not in use, although + misdn_lib_send_event() already did the same. This is bad. When + it's not in use we are not allowed to touch it. * Moved log + message in front of the resulting actions and fixed it to match + the case. Patches: patch5_bccleanup.diff (license #6372) patch + uploaded by Guenther Kelleter JIRA ABE-2882 ................ + r374520 | rmudgett | 2012-10-04 18:43:56 -0500 (Thu, 04 Oct 2012) + | 12 lines chan_misdn: Fix memory leaks, bc, chan not cleaned up + etc., really bad stuff. * Fix return codes of cb_events() for + EVENT_SETUP to use caller's cleanup mechanisms. * Move + cl_queue_chan() call after bearer check. Patches: + patch6_leaks.diff (license #6372) patch uploaded by Guenther + Kelleter JIRA ABE-2882 ................ r374521 | rmudgett | + 2012-10-04 18:48:38 -0500 (Thu, 04 Oct 2012) | 11 lines + chan_misdn: We must initialize cause on sending a DISCONNECT. We + must initialize cause on sending a DISCONNECT, so it is later + correctly indicated to ast_channel in case the answer + (RELEASE/RELEASE_COMPLETE) does not include one. Patches: + patch7_hangupcause.diff (license #6372) patch uploaded by + Guenther Kelleter JIRA ABE-2882 ................ r374522 | + rmudgett | 2012-10-04 19:03:56 -0500 (Thu, 04 Oct 2012) | 7 lines + chan_misdn: Remove unused code for upqueue Patches: + patch8_unused-upqueue.diff (license #6372) patch uploaded by + Guenther Kelleter JIRA ABE-2882 ................ r374523 | + rmudgett | 2012-10-04 19:11:50 -0500 (Thu, 04 Oct 2012) | 7 lines + chan_misdn: Improve debugging (port number, messages fixed, dups + removed) Patches: patch9_debug.diff (license #6372) patch + uploaded by Guenther Kelleter JIRA ABE-2882 ................ + r374533 | rmudgett | 2012-10-05 12:17:18 -0500 (Fri, 05 Oct 2012) + | 8 lines chan_misdn: Better debug: we can print_bc_info even if + there's no ast leg. Patches: patch10_debug-bc-2.diff (license + #6372) patch uploaded by Guenther Kelleter Modified. JIRA + ABE-2882 ................ r374534 | rmudgett | 2012-10-05 + 12:34:10 -0500 (Fri, 05 Oct 2012) | 16 lines chan_misdn: + setup_bc() is called too early for an incoming SETUP on TE. This + prevents the B channel from being setup for HDLC mode when + requested by the bearer capability and config option hdlc=yes. It + violates ETS300102 Ch.5.2.3.2: "The user, in any case, must not + connect to the channel until a CONNECT ACKNOWLEDGE message has + been received." * Call setup_bc() on receipt of + CONNECT_ACKNOWLEGDE for PTMP, and on first response to SETUP for + PTP. Patches: abe-2881-2.diff (license #6372) patch uploaded by + Guenther Kelleter Modified. JIRA ABE-2881 ................ + r374535 | rmudgett | 2012-10-05 12:41:05 -0500 (Fri, 05 Oct 2012) + | 2 lines chan_misdn: Remove some more deadcode. ................ + ........ Merged revisions 374536 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374537 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-04 20:18 +0000 [r374477-374485] Alec L Davis + + * main/dsp.c, /, configs/dsp.conf.sample, CHANGES: dsp.c User + Configurable DTMF_HITS_TO_BEGIN and DTMF_MISSES_TO_END Instead of + a recompile, allow values to be adjusted in dsp.conf For binary + distributions allows easy adjustment for wobbly GSM calls, and + other reasons. Defaults to DTMF_HITS_TO_BEGIN=2 and + DTMF_MISSES_TO_END=3 (closes issue ASTERISK-17493) Tested by: + alecdavis alecdavis (license 585) Review + https://reviewboard.asterisk.org/r/2144/ ........ Merged + revisions 374479 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374481 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/dsp.c, /: dsp.c fix incorrect DTMF Digit_Duration. it's + always short by 'hits_to_begin*DTMF_GSIZE', or 25.5ms if + hitstobegin=2 (issue ASTERISK-16003) Tested by: alecdavis + alecdavis (license 585) Review + https://reviewboard.asterisk.org/r/2145/ ........ Merged + revisions 374475 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374476 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-04 15:42 +0000 [r374428] dlee : + + * main/db.c, /, res/res_agi.c: Fix DBDelTree error codes for AMI, + CLI and AGI The AMI DBDelTree command will return Success/Key + tree deleted successfully even if the given key does not exist. + The CLI command 'database deltree' had a similar problem, but was + saved because it actually responded with '0 database entries + removed'. AGI had a slightly different error, where it would + return success if the database was unavailable. This came from + confusion about the ast_db_deltree retval, which is -1 in the + event of a database error, or number of entries deleted + (including 0 for deleting nothing). * Changed some poorly named + res variables to num_deleted * Specified specific errors when + calling ast_db_deltree (database unavailable vs. entry not found + vs. success) * Fixed similar bug in AGI database deltree, where + 'Database unavailable' results in successful result (closes issue + AST-967) Reported by: John Bigelow Review: + https://reviewboard.asterisk.org/r/2138/ ........ Merged + revisions 374426 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374427 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-04 04:43 +0000 [r374379-374386] Alec L Davis + + * main/dsp.c, /, configs/dsp.conf.sample, CHANGES: dsp.c User + configuration of DTMF_NORMAL_TWIST and DTMF_REVERSE_TWIST values + Asterisk's DTMF Specifications are based on AT&T specs, which may + not be compatible in other countries. Various countries have + different specifications for the maximum power level differences + between the DTMF low group and high group of frequencies. Power + level difference between frequencies for different + Administrations/RPOAs NTT = Max. 5 dB AT&T = 4dB(reverse) to + 8dB(normal) Danish = Max. 6 dB Australian = Max. 10 dB Brazilian + = Max. 9 dB ETSI = Max. 6 dB from ETSI ES 201 235-3 V1.3.1 + (2006-03) Now allow 4 variables to be individually configured in + dsp.conf, with reasonable min/max of 2dB to 20dB. Default is AT&T + specifications Add's the following variables to dsp.conf + ;dtmf_normal_twist=6.31 ;dtmf_reverse_twist=2.51 + ;relax_dtmf_normal_twist=6.31 ;relax_dtmf_reverse_twist=3.98 + (closes issue ASTERISK-20442) Reported by: tbsky Tested by: + tbsky,alecdavis alecdavis (license 585) Review + https://reviewboard.asterisk.org/r/2141/ ........ Merged + revisions 374384 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374385 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /: _dsp_init: bring inline with trunk preparation for clean merge + of DTMF TWIST patch No functional changes, just style. alecdavis + (license 585) Reported by: Alec Davis Tested by: alecdavis + related https://reviewboard.asterisk.org/r/2141 ........ Merged + revisions 374365 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374370 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-04 02:15 +0000 [r374196-374337] Matthew Jordan + + * /, res/res_jabber.c: Check for presence of buddy in info/dinfo + handlers The res_jabber resource module uses the ASTOBJ library + for managing its ref counted objects. After calling + ASTOBJ_CONTAINER_FIND to locate a buddy object, the pointer to + the object has to be checked to see if the buddy existed. Prior + to this patch, the buddy object was not checked for NULL; with + this patch in both aji_client_info_handler and aji_dinfo_handler + the pointer is checked before used and, if no buddy object was + found, the handlers return an error code. This patch does not + take the approach that our JID can be used to log in from another + resource. If that approach is desired, an improvement could be + made to this patch to create the buddy on the fly. This patch + seeks only to prevent Asterisk from crashing. FYI: In Asterisk + 11+, you really should be using res_xmpp. It does not have this + problem, as it moved to the astobj2 library. Note that multiple + people have proposed patches for this issue; the patch being + committed here is based on those. (closes issue ASTERISK-19532) + Reported by: Karsten Wemheuer Tested by: Byron Clark patches: + fix-jabber uploaded by Karsten Wemheuer (license #5930) + xmpp_no_crash_with_ejabberd.patch uploaded by Byron Clark + (license #6157) (closes issue ASTERISK-19557) Reported by: + ulugutz ........ Merged revisions 374335 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374336 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/ccss.c: Destroy the generic_monitors container after the + core_instances in ccss For each item in core_instances disposed + of in the shutdown of ccss, any generic monitor instances + referenced by the objects will be removed from generic_monitors + during their destruction. Hilarity ensues if generic_monitors no + longer exists. Thanks to the Asterisk Test Suite's generic_ccss + test for complaining loudly when it ran into this. ........ + Merged revisions 374300 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/asterisk.c, /: Ensure Shutdown AMI event is still fired + during Asterisk shutdown Richard pointed out that having the + manager dispose of itself gracefully during shutdown meant that + the Shutdown event will no longer get fired. This patch moves the + AMI event just prior to running the atexit callbacks. ........ + Merged revisions 374230 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374231 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/message.c: Fix findings from check-in on r374177 Richard + pointed out two problems with the check-in from r374177: * The + ast_msg_shutdown function declaration doesn't match the prototype + in main/message.c. * The ref/alloc function usage in astobj2 (in + trunk) can use the ao2_t_* variants of the functions to allow the + REF_DEBUG flag to enable/disable their debug counterparts. + ........ Merged revisions 374210 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/db.c, main/asterisk.c, main/xmldoc.c, main/format.c, + main/udptl.c, main/pbx.c, /, main/ccss.c, + include/asterisk/astobj2.h, channels/chan_agent.c, + main/taskprocessor.c, res/res_musiconhold.c, res/res_xmpp.c, + main/cel.c, main/named_acl.c, main/indications.c, + main/format_pref.c, main/astobj2.c, main/channel.c, main/data.c, + main/manager.c, main/features.c, main/config_options.c, + main/event.c, main/message.c: Fix a variety of ref counting + issues This patch resolves a number of ref leaks that occur + primarily on Asterisk shutdown. It adds a variety of shutdown + routines to core portions of Asterisk such that they can reclaim + resources allocate duringd initialization. Review: + https://reviewboard.asterisk.org/r/2137 ........ Merged revisions + 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 374178 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-01 20:26 +0000 [r374133-374150] Sean Bright + + * main/db.c, include/asterisk/astdb.h, /, tests/test_db.c, + apps/app_queue.c: app_queue: Support persisting and loading of + long member lists. Greenlight in #asterisk brought up that he was + receiving an error message "Could not create persistent member + string, out of space" when running app_queue in Asterisk 10. + dump_queue_members() made an assumption that 8K would be enough + to store the generated string, but with queues that have large + member lists this is not always the case. This patch removes the + limitation and uses ast_str instead of a fixed sized buffer. The + complicating factor comes from the fact that ast_db_get requires + a buffer and buffer size argument, which doesn't let us pull back + more than what we pass in, so I introduced a new + ast_db_get_allocated() which returns an ast_strdup()'d copy of + the value from astdb. As an aside, I did some testing on the + maximum size of data that we can store in the BDB library we + distribute and was able to store a 10MB string and retrieve it + with no problems, so I feel this is a safe patch. Review: + https://reviewboard.asterisk.org/r/2136/ ........ Merged + revisions 374108 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 374135 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/db.c, /: Use ast_copy_string instead of strncpy to guarantee + a NUL terminated string. ........ Merged revisions 374132 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-10-01 16:12 +0000 [r374106] Mark Michelson + + * apps/confbridge/conf_config_parser.c: Don't destroy confbridge + config when error is encountered during a reload. Not panicking + means that the old config is kept. (closes issue ASTERISK-20458) + Reported by: Leif Madsen Patches: ASTERISK-20458.patch uploaded + by Mark Michelson(license #5049) Tested by Leif Madsen + +2012-09-29 03:54 +0000 [r374085] Matthew Jordan + + * channels/chan_sip.c: Fix ref leak when adding ICE candidates to + an SDP There was a missing decrement to the reference count for + the current ICE candidate when local candidates are being added + to an outbound SDP. This patch corrects that. + +2012-09-28 19:29 +0000 [r374059] Jonathan Rose + + * /, res/res_jabber.c: res_jabber: Remove CLI command 'jabber test' + The opinion of development was that it is both improper to have + Matt's personal email address used in the source and that the + command wouldn't be useful without it. (closes issue AST-467) + Reported by: Malcolm Davenport ........ Merged revisions 374032 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 374045 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-28 13:02 +0000 [r374019] beagles : + + * res/res_xmpp.c, main/message.c: Reset hangup flags on channels + created through messages and cleanup globals in res_xmpp on + unload. This patch fixes an issue where hangup flags were not + being reset on a channel, affecting subsequent use of that + channel. The patch also adds some additional cleanup to res_xmpp + to fix an issue with reloading the module. (closes + ASTERISK-20360) Reported by: Noah Engelberth Tested by: beagles + Review: https://reviewboard.asterisk.org/r/2134/ + +2012-09-28 12:16 +0000 [r373991] Joshua Colp + + * /, res/res_agi.c: Update documentation to make it explicit that + "stream file" will not restart musiconhold. (issue + ASTERISK-17367) Reported by: oej ........ Merged revisions 373989 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 373990 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-27 22:19 +0000 [r373954] Richard Mudgett + + * /, apps/app_senddtmf.c: Fix SendDTMF crash and channel reference + leak using channel name parameter. The SendDTMF channel name + parameter has two issues. 1) Crashes if the channel name does not + exist. 2) Leaks a channel reference if the channel is the current + channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF + documentation. * Renamed app to senddtmf_name and tweaked the + type. ........ Merged revisions 373945 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373946 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-27 17:05 +0000 [r373880-373914] Joshua Colp + + * channels/chan_sip.c, include/asterisk/http_websocket.h, + res/res_http_websocket.c: Make res_http_websocket an optional + dependency on supported platforms for chan_sip. (closes issue + ASTERISK-20439) Reported by: sruffell Patches: + 0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded + by sruffell (license 5417) + + * main/loader.c, /: loader: Ensure dependent modules are properly + initialized. If an Asterisk module specifies a dependency in + ast_module_info.nonoptreq, it is possible for Asterisk to skip + calling the modules's .load function. Asterisk was loading and + linking the module via load_dynamic_module() but was not adding + the module to the resource_heap. Therefore the module was not + initialized based on it's priority along with the other modules + in the heap. Now use load_resource() instead of + load_dynamic_module() for non-optional requirement. This will add + the module to the resource_heap so the module can be properly + initialized in the correct order. This is required if there are + any module global data structures initialized in the .load() + callback for the module on platforms which do not support weak + references. (issue ASTERISK-20439) Reported by: sruffell Patches: + 0001-loader-Ensure-dependent-modules-are-properly-initial.patch + uploaded by sruffell (license 5417) ........ Merged revisions + 373909 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 373910 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_local.c, /: Fix an issue where Local channels + dialed by app_queue are considered in use immediately. The + chan_local channel driver returns a device state of in use even + if a created Local channel has not yet been dialed. This fix + changes the logic to return a state of not in use until the + channel itself has been dialed. (closes issue ASTERISK-20390) + Reported by: tim_ringenbach Review: + https://reviewboard.asterisk.org/r/2116/ ........ Merged + revisions 373878 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373879 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-26 21:16 +0000 [r373850] Mark Michelson + + * /, channels/chan_sip.c: Move handling of 408 response so there is + no misleading warning message. (closes issue ASTERISK-20060) + Reported by: Walter Doekes ........ Merged revisions 373848 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373849 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-26 18:18 +0000 [r373818] Richard Mudgett + + * /, apps/app_meetme.c: Fixed meetme tab completion and command + documentation. * Removed unnecessary case sensitivity in meetme + list, lock, unlock, mute, unmute, and kick commands. * Separated + meetme lock/unlock, mute/unmute, and kick commands into their own + registered commands to simplify tab completion and parameter + checking. meetme_lock_cmd(), meetme_mute_cmd(), and + meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue + AST-1006) Reported by: John Bigelow Tested by: rmudgett ........ + Merged revisions 373815 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373816 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-26 08:29 +0000 [r373804] Alec L Davis + + * apps/app_queue.c: app_queue: 'agent available' hint, cleanup + restart, and initial state Fix previously untested senarios; 1). + On queue initialisation set queue_avail devstate to INUSE. + Previously was unavailable, which indicated an agent was + available. 2). When removing members, if there are no other + members available, set queue_avail to INUSE. Previously, if a + member interface had become 'unavailable', they were never going + to be removed, particularly when persistant queues is enabled. + 3). When adding a member, check that they are available, if they + are set queue_avail to NOT_INUSE. Previously on reloaded, members + may have been 'unavailable'. 4). When pausing or unpausing a + member, set appropriate queue availability. alecdavis (license + 585) Reported by: Alec Davis Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/2129/ + +2012-09-25 23:09 +0000 [r373738-373775] Mark Michelson + + * /, main/say.c: Fix saying of date in Dutch. The Dutch say the + date before the month. (closes issue ASTERISK-20353) Reported by: + Teun Ouwehand ........ Merged revisions 373773 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373774 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * configs/agents.conf.sample, /, channels/chan_agent.c: Remove dead + code and documentation for nonexistent feature. multiplelogin was + removed from chan_agent back in 1.6.0 when AgentCallbackLogin() + was removed. (closes issue AST-948) reported by Steve Pitts + ........ Merged revisions 373768 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373769 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_voicemail.c, /: Fix error where improper IMAP greetings + would be deleted. (closes issue ASTERISK-20435) Reported by: + fhackenberger Patches: asterisk-20435-imap-del-greeting.diff + uploaded by Michael L. Young (License #5026) (with suggested + modification made by me) ........ Merged revisions 373735 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373737 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-25 20:13 +0000 [r373707] Joshua Colp + + * channels/chan_local.c, /: Fix T.38 support when used with + chan_local in between. Users of the T.38 API can indicate + AST_T38_REQUEST_PARMS on a channel to request that the channel + indicate a T.38 negotiation with the parameters present on the + channel. The return value of this indication is expected to be + AST_T38_REQUEST_PARMS upon success but with chan_local involved + this could never occur. This fix changes chan_local to always + return AST_T38_REQUEST_PARMS for this situation. If the + underlying channel technology on the other side does not support + T.38 this would have been determined ahead of time using + ast_channel_get_t38_state and an indication would not occur. + (closes issue ASTERISK-20229) Reported by: wdoekes Patches: + ASTERISK-20229.patch uploaded by wdoekes (license 5674) Review: + https://reviewboard.asterisk.org/r/2070/ ........ Merged + revisions 373705 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373706 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-25 19:35 +0000 [r373704] Kinsey Moore + + * /: Recorded merge of revisions 373703 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Fix an + issue where media would not flow for situations where the legacy + STUN code is in use. The STUN packets should *not* be blocked by + strict RTP. (closes issue ASTERISK-20415) Reported-by: Michele + Cicciotti Patch-by: Josh Colp (trunk r369817) ........ Merged + revisions 373702 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2012-09-25 18:52 +0000 [r373690] Terry Wilson + + * channels/sip/include/sip.h, /, channels/chan_sip.c, + configs/sip.conf.sample: Properly handle UAC/UAS roles for SIP + session timers The SIP session timer mechanism contains a + mandatory 'refresher' parameter (included in the Session-Expires + header) which is used in the session timer offer/answer signaling + within a SIP Invite dialog. It looks like asterisk is + interpreting the uac resp. uas role only as the initial role of + client and server (caller is uac, callee is uas). The standard + rfc 4028 however assigns the client role to the ((RE)-Invite) + requester, the server role to the ((RE)-Invite) responder. This + patch has Asterisk track the actual refresher as "us" or "them" + as opposed to relying on just the configured "uas" or "uac" + properties. (closes issue AST-922) Reported by: Thomas Airmont + Review: https://reviewboard.asterisk.org/r/2118/ ........ Merged + revisions 373652 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373665 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-25 18:24 +0000 [r373688] Kinsey Moore + + * /, apps/app_queue.c: "show" completion option for "queue" + shouldn't appear twice When tab-completing CLI commands starting + with "queue", "show" appeared twice in the list due to the way + that Asterisk's tab completion functions and the order in which + the commands were registered. The registration order has been + altered to resolve this issue. (closes issue AST-940) + Reported-by: Steve Pitts ........ Merged revisions 373666 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373675 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-25 17:21 +0000 [r373635-373650] Richard Mudgett + + * /, codecs/ilbc/iLBC_encode.c, codecs/ilbc/iLBC_decode.c: Fix + valgrind found memcpy issues in codec_ilbc. Valgrind found + codec_ilbc using memcpy instead of memmove for overlapping memory + blocks. (issue ASTERISK-19890) (closes issue ASTERISK-20231) + Reported by: Walter Doekes Patches: ASTERISK-20231.patch (license + #5674) patch uploaded by Walter Doekes ........ Merged revisions + 373640 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 373645 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * codecs/Makefile, /: Make rebuild GSM, ilbc, or lpc10 codecs if + the respective sources change. ........ Merged revisions 373618 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 373633 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-25 16:31 +0000 [r373632] Jonathan Rose + + * /, channels/chan_sip.c: chan_sip: Set Quality of Service for + video rtp instance (closes issue ASTERISK-20201) Reported by: + ddkprog Patches: chan_sip.c.diff uploaded by ddkprog (license + 6008) ........ Merged revisions 373617 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373631 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-25 14:12 +0000 [r373582] Mark Michelson + + * funcs/func_presencestate.c: "He who go through turnstile sideways + is going to Bangkok" + +2012-09-25 13:29 +0000 [r373580] Kinsey Moore + + * configs/res_odbc.conf.sample, /: Fix documentation for default + username in res_odbc This was previously stated to be "root", but + is actually the name of the context if unspecified. (closes issue + ASTERISK-20258) Reported by: Stefan x ........ Merged revisions + 373578 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 373579 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-25 12:07 +0000 [r373552] Joshua Colp + + * res/res_rtp_multicast.c, /: Fix an issue where a caller to + ast_write on a MulticastRTP channel would determine it failed + when in reality it did not. When sending RTP packets via + multicast the amount of data sent is stored in a variable and + returned from the write function. This is incorrect as any + non-zero value returned is considered a failure while a return + value of 0 is success. For callers (such as ast_streamfile) that + checked the return value they would have considered it a failure + when in reality nothing went wrong and it was actually a success. + The write function for the multicast RTP engine now returns -1 on + failure and 0 on success, as it should. (closes issue + ASTERISK-17254) Reported by: wybecom ........ Merged revisions + 373550 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 373551 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-24 22:17 +0000 [r373508] Matthew Jordan + + * res/res_rtp_asterisk.c, /: Revert change to res_rtp_asterisk + committed in r373236 (1.8) The change committed in r373236 + attempted to account for endpoints that increased their RTP + timestamp in DTMF end of event re-transmissions. This change + attempted to make Asterisk continue to work with endpoints that + failed to follow the RFC while maintaining the fix that allowed + for out of order DTMF to be handled. Unfortunately, there is no + free lunch, and this patch broke any system that sent DTMF + immediately after an RTP session was established or when an SSRC + is updated. As such, that patch is being reverted for the + previous behavior. Endpoints that erroneously increase the RTP + timestamp in DTMF end of event packets will not work properly + with Asterisk. (issue ASTERISK-20424) ........ Merged revisions + 373504 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 373505 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-24 22:12 +0000 [r373502] Richard Mudgett + + * /, channels/chan_sip.c: Be consistent, send From: "Anonymous" + When setting + CALLERID(pres)=unavailable in the dialplan, the From header in + the SIP message contains "Anonymous" + . For consistency, Asterisk + should use a lowercase a in the userpart of the URI. * Make the + From header use a lowercase A in the userpart of the anonymous + URI. (closes issue ASTERISK-19838) Reported by: Antti Yrjola + Patches: chan_sip_patch_ASTERISK-19838.patch (license #6383) + patch uploaded by Antti Yrjola ........ Merged revisions 373500 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 373501 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-24 21:12 +0000 [r373470] Jonathan Rose + + * funcs/func_audiohookinherit.c, /, apps/app_mixmonitor.c: + func_audiohookinherit: Document some missed sources. This patch + also mentions that AUDIOHOOK_INHERIT can be used to transfer + MixMonitor audiohooks. There is also wiki that addresses + audiohooks and the use of AUDIOHOOK_INHERIT at the following + link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks + (closes issue ASTERISK-18220) Reported by: Ishfaq Malik ........ + Merged revisions 373467 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373468 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-24 21:08 +0000 [r373469] Richard Mudgett + + * /, channels/chan_sip.c: Fix potential reentrancy problems in + chan_sip. Asterisk v1.8 and later was not as vulnerable to this + issue. * Made find_call() lock each private as it processes the + found dialogs. (Primary cause of ABE-2876) * Made the other + functions that traverse the dialogs container lock each private + as it examines them. * Fix race condition in sip_call() if the + thread that sent the INVITE is held up long enough for a response + to be processed. The p->initid for the INVITE retransmission + could be added after it was canceled by the response processing. + * Made __sip_destroy() clean up resource pointers after freeing. + This is primarily defensive in case someone has a stale private + pointer. * Removed redundant memset() in reqprep(). The call to + init_req() already does the memset() and is the first reference + to req in reqprep(). * Removed useless set of req.method in + transmit_invite(). The calls to initreqprep() and reqprep() have + to do this because they memset() the req. JIRA ABE-2876 + .......... Merged -r373423 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + ........ Merged revisions 373424 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373466 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-24 19:21 +0000 [r373413-373454] Joshua Colp + + * /, channels/chan_sip.c: Fix a deadlock caused by a race condition + between removing a hint and reloading the dialplan and + subscribing to the removed hint. If conditions were right it was + possible for both the PBX core and chan_sip to deadlock by both + having a lock that the other wants. In the case of the PBX core + it had the contexts lock and wanted a SIP dialog lock, while in + the case of chan_sip it had the SIP dialog lock and wanted the + contexts lock. This fix unlocks the SIP dialog before getting the + extension state so that the other thread will not block on trying + to lock it. Once the extension state is retrieved the SIP dialog + is locked again and life carries on. As the SIP dialog is + reference counted it is not possible for it to go away after + unlocking. (closes issue ASTERISK-20437) Reported by: jhutchins + ........ Merged revisions 373438 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373440 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_sip.c, res/res_format_attr_h264.c: Fix an issue + with H.264 format attribute comparison and fix an issue with + improper SDP being produced. The H.264 format attribute module + compares two format attribute structures to determine if they are + compatible or not. In some instances it was possible for this + check to determine that both structures were incompatible when + they actually should be considered compatible. This check has now + been made even more permissive by assuming that if no attribute + information is available the two structures are compatible. If + both structures contain attribute information a base level + comparison of the H.264 IDC value is done to see if they are + compatible or not. The above issue uncovered a secondary issue in + chan_sip where the SDP being produced would be incorrect if the + formats were considered incompatible. This has now been fixed by + checking that all information required to produce the SDP is + available instead of assuming it is. (closes issue + ASTERISK-20464) Reported by: Leif Madsen + +2012-09-24 12:33 +0000 [r373403] beagles : + + * res/res_rtp_asterisk.c, configs/rtp.conf.sample: + res_rtp_asterisk: Make TURN and STUN server configurations + consistent. This patch removes the turnport configuration + property and changes the turnaddr property to be a combined + host[:port] configuration string. The patch also modifies the + documentation in the example configuration to reflect the + property changes and adds some additional text indicating how the + STUN port is configured. (closes issue ASTERISK-20344) Reported + by: beagles Tested by: beagles Review: + https://reviewboard.asterisk.org/r/2111/ + +2012-09-21 19:29 +0000 [r373318-373368] Jonathan Rose + + * /, channels/iax2-provision.c: iax2-provision: Fix improper return + on failed cache retrieval (closes issue ASTERISK-20337) reported + by: John Covert Patches: iax2-provision.c.patch uploaded by John + Covert (license 5512) ........ Merged revisions 373342 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373343 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_queue.c: app_queue: Make queue reload members and + variants of that work Prior to this patch, 'queue reload members' + cli command did not work at all. This also affects the manager + function 'QueueReload' when supplied with the 'members: yes' + field. (closes issue AST-956) Reported by: John Bigelow ........ + Merged revisions 373298 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373300 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-20 19:16 +0000 [r373246] Joshua Colp + + * /, apps/app_meetme.c: Fix incorrect MeetME conference bridge + reference count decrementing and sometimes premature destruction. + When using the 'e' or 'E' option to MeetMe the configured + conference bridges are loaded and examined to see if any are + empty. If no conference bridges are empty the caller is prompted + to enter the number of one. This operation left around a pointer + to the last created conference bridge still containing + participants. When the caller that was not able to find any empty + conference bridge hung up this pointer was disposed of and the + reference count of the conference bridge decremented. If there + was only a single participant in the conference bridge it was + ultimately destroyed prematurely. (closes issue AST-994) Reported + by: John Bigelow ........ Merged revisions 373242 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373245 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-20 18:59 +0000 [r373235-373240] Matthew Jordan + + * configs/extensions.conf.sample, CHANGES, apps/app_queue.c: + app_queue: Support an 'agent available' hint Sets INUSE when no + free agents, NOT_INUSE when an agent is free. modifes + handle_statechange() scan members loop to scan for a free agent + and updates the Queue:queuename_avial devstate. Previously exited + early if the member was found in the queue. Now Exits later when + both a member was found, and a free agent was found. alecdavis + (license 585) Reported by: Alec Davis Tested by: alecdavis + Review: https://reviewboard.asterisk.org/r/2121/ ~~~~ Support all + ways a member can be available for 'agent available' hints Alec's + patch in r373188 added the ability to subscribe to a hint for + when Queue members are available. This patch modifies the check + that determines when a Queue member is available by refactoring + the availability checks in num_available_members into a shared + function is_member_available. This should now handle the + ringinuse option, as well as device state values other than + AST_DEVICE_NOT_INUSE. + + * res/res_rtp_asterisk.c, /: When processing RFC 2833 DTMF, + accomodate increasing timestamps in End events While endpoints + should not be changing the source timestamp between DTMF event + packets, the fact is there exists those endpoints that do exactly + that. To work around this, we absorb timestamps within the + expected re-transmit period. Note that this period only affects + End of Event packets, so it should not prevent the detection of + new DTMF digits that happen to arrive right on top of each other. + (closes issue ASTERISK-20424) Reported by: Vladimir Mikhelson + Tested by: mjordan, Vladimir Mikhelson Review: + https://reviewboard.asterisk.org/r/2124 ........ Merged revisions + 373236 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 373237 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * configs/extensions.conf.sample, CHANGES, apps/app_queue.c: Add + queue monitoring hints This patch adds support for hints on a + queue. Hints can be added using the nomenclature 'Queue:name', + where name is the name of the queue being monitored. This nifty + feature was done by Alec Davis. Review: + https://reviewboard.asterisk.org/r/1619 Reported by: Alec Davis + Tested by: alecdavis patches: review1619.diff2 by alecdavis + (license 585) + +2012-09-20 18:18 +0000 [r373229] Joshua Colp + + * channels/sip/include/sip.h, res/res_rtp_asterisk.c, + main/rtp_engine.c, channels/chan_sip.c, configure, + include/asterisk/autoconfig.h.in, configure.ac, + configs/sip.conf.sample, include/asterisk/rtp_engine.h: Add + support for DTLS-SRTP to res_rtp_asterisk and chan_sip. As + mentioned on the review for this, WebRTC has moved towards + choosing DTLS-SRTP as the mechanism for key exchange for SRTP. + This commit adds support for this but makes it available for + normal SIP clients as well. Testing has been done to ensure that + this introduces no regressions with existing behavior and also + that it functions as expected. Review: + https://reviewboard.asterisk.org/r/2113/ + +2012-09-20 17:15 +0000 [r373220] Richard Mudgett + + * include/asterisk/features.h, main/channel.c, + apps/app_directed_pickup.c, funcs/func_channel.c, + main/features.c, include/asterisk/channel.h: Named call pickup + groups. Fixes, missing functionality, and improvements. * + ASTERISK-20383 Missing named call pickup group features: + CHANNEL(callgroup) - Need CHANNEL(namedcallgroup) + CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup) Pickup() - + Needs to also select from named pickup groups. * ASTERISK-20384 + Using the pickupexten, the pickup channel selection could fail + even though there was a call it could have picked up. In a call + pickup race when there are multiple calls to pickup and two + extensions try to pickup a call, it is conceivable that the loser + will not pick up any call even though it could have picked up the + next oldest matching call. Regression because of the named call + pickup group feature. * See ASTERISK-20386 for the implementation + improvements. These are the changes in channel.c and channel.h. * + Fixed some locking issues in CHANNEL(). (closes issue + ASTERISK-20383) Reported by: rmudgett (closes issue + ASTERISK-20384) Reported by: rmudgett (closes issue + ASTERISK-20386) Reported by: rmudgett Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/2112/ + +2012-09-20 13:00 +0000 [r373211] Kinsey Moore + + * channels/chan_sip.c: Correct handling of unknown SDP stream types + When the patch to handle arbitrary SDP stream arrangements went + into Asterisk, it also included an ability to transparently + decline unknown stream types. The scanf calls used were not + checked properly causing this part of the functionality to be + broken. (closes issue ASTERISK-20203) + +2012-09-18 20:14 +0000 [r373133] Sean Bright + + * main/manager.c, /: Don't crash when passing a NULL message to + __astman_get_header. Before this commit, __astman_get_header + would blindly dereference the passed in 'struct message *' to + traverse the header list. There are cases, however, such as + '*CLI> sip qualify peer foo' where the message pointer is NULL, + so we need to check for that. ........ Merged revisions 373131 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 373132 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-18 15:47 +0000 [r373119] dlee : + + * Makefile, include/asterisk/utils.h, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add + -fnested-functions compile flag, if needed. In order to use + nested functions on some versions of GCC (e.g. GCC on OS X), the + -fnested-functions flag must be passed to the compiler. This + patch adds detection logic to ./configure to add the flag if + necessary. It also adds a comment to utils.h as to why the nested + function needs a prototype. (closes issue ASTERISK-20399) + Reported by: David M. Lee Review: + https://reviewboard.asterisk.org/r/2102/ + +2012-09-15 00:27 +0000 [r373107] Richard Mudgett + + * channels/sig_ss7.c, /: Made companding law for SS7 calls only + determined by SS7 signaling type. For SS7, the companding law for + a call was chosen inconsistently depending upon ss7type (ITU vs + ANSI) and the DAHDI companding default (T1 vs E1). For incoming + calls, the companding law was determined by ss7type. For outgoing + calls, the companding law was determined by the DAHDI default. + With the wrong combination you would get A-law/u-law conflicts. + An A-law/u-law conflict sounds like bad static on the line. SS7 + ITU signaling with E1 line: ok SS7 ITU signaling with T1 line: + noise SS7 ANSI signaling with E1 line: noise SS7 ANSI signaling + with T1 line: ok * Fix the companding law used to be determined + by the SS7 signaling type only. ........ Merged revisions 373090 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 373101 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-14 19:50 +0000 [r373079] Matthew Jordan + + * main/tcptls.c, /, channels/chan_sip.c, main/libasteriskssl.c: + Resolve memory leaks in TLS initialization and TLS client + connections This patch resolves two sources of memory leaks when + using TLS in Asterisk: 1) It removes improper initialization (and + multiple re-initializations) of portions of the SSL library. + Asterisk calls SSL_library_init and SSL_load_error_strings during + SSL initialization; collectively this obviates the need for + calling any of the following during initialization or client + connection handling: * ERR_load_crypto_strings (handled by + SSL_load_error_strings) * OpenSSL_add_all_algorithms (synonym for + SSL_library_init) * SSLeay_add_ssl_algorithms (synonym for + SSL_library_init) 2) Failure to completely clean up all memory + allocated by Asterisk and by the SSL library for TLS clients. + This included not freeing the SSL_CTX object in the SIP channel + driver, as well as not clearing the error stack when the TLS + client exited. Note that these memory leaks were found by Thomas + Arimont, and this patch was essentially written by him with some + minor tweaks. (closes issue AST-889) Reported by: Thomas Arimont + Tested by: Thomas Arimont patches: (bugAST-889.patch) by Thomas + Arimont (license 5525) Review: + https://reviewboard.asterisk.org/r/2105 ........ Merged revisions + 373061 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 373062 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-13 20:04 +0000 [r373029-373047] dlee : + + * main/Makefile: Fixed make clean when configured + --disable-asteriskssl + + * main/channel.c, /, include/asterisk/channel.h: Fix timeouts for + ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass + its timeout to ast_waitfor_nandfds, expecting it to decrement the + timeout by however many milliseconds were waited. This is a + problem if it consistently waits less than 1ms. The timeout will + never be decremented, and we wait... FOREVER! This patch makes + ast_waitfordigit_full manage the timeout itself. It maintains the + previously undocumented behavior that negative timeouts wait + forever. (closes issue ASTERISK-20375) Reported by: Mark + Michelson Tested by: Mark Michelson Review: + https://reviewboard.asterisk.org/r/2109/ ........ Merged + revisions 373024 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 373025 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-12 20:53 +0000 [r372995] Joshua Colp + + * channels/chan_motif.c: Skip any non-content information when + looking for and handling content. This fixes a bug with Jitsi and + conference calling. Jitsi implements XEP-0298 which places some + conference-info information in the session-initiate request which + chan_motif did not expect to occur. + +2012-09-12 18:23 +0000 [r372984] Jonathan Rose + + * res/res_xmpp.c: res_xmpp: Fix a segfault caused by bodyless + messages (closes issue ASTERISK-20361) Reported by: Noah + Engelberth Review: https://reviewboard.asterisk.org/r/2108/ + +2012-09-12 15:19 +0000 [r372937] Mark Michelson + + * /, channels/chan_sip.c: Add channel name to a warning to make + debugging easier. The "autodestruct with owner in place" message + is typically indicative of a channel reference leak. Printing out + the name of the channel in the message may be helpful when trying + to debug the issue. ........ Merged revisions 372932 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372933 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-12 14:18 +0000 [r372930] dlee : + + * main/Makefile: Fixed r372696 when configured + --disable-asteriskssl; properly install libasteriskssl.dylib on + OS X. I didn't realize that libasteriskssl.c was still compiled, + even when you disable asteriskssl; it simple gets statically + linked into asterisk. + +2012-09-11 22:32 +0000 [r372917] Jonathan Rose + + * channels/chan_local.c, /: chan_local: Switch from using a random + 4 digit hex identifier to unique id Changes chan_local channels + to use an 8 digit hex identifier generated atomically and + sequentially in order to eliminate the chance of having multiple + channels with the same name during high call volume situations. + (issue ASTERISK-20318) Reported by: Dan Cropp Review: + https://reviewboard.asterisk.org/r/2104/ ........ Merged + revisions 372902 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372916 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-11 21:15 +0000 [r372886-372888] Mark Michelson + + * main/asterisk.c, /, include/asterisk/_private.h, main/message.c: + Fix inability to shutdown gracefully due to an unending channel + reference. message.c makes use of a special message queue channel + that exists in thread storage. This channel never goes away due + to the fact that the taskprocessor used by message.c does not get + shut down, meaning that it never ends the thread that stores the + channel. This patch fixes the problem by shutting down the + taskprocessor when Asterisk is shut down. In addition, the thread + storage has a destructor that will release the channel reference + when the taskprocessor is destroyed. (closes issue AST-937) + Reported by Jason Parker Patches: AST-937.patch uploaded by Mark + Michelson (License #5049) Tested by Jason Parker ........ Merged + revisions 372885 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/features.c: Fix bad channel application data reference. + When channels get bridged due to an AMI bridge action or a DTMF + attended transfer, the two channels that get bridged have their + application data pointing to the other channel's name. This means + that if one channel is hung up but the other moves on, it means + that the channel that moves on will have its application data + pointing at freed memory. (issue ASTERISK-20335) Reported by: + aragon ........ Merged revisions 372840 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372841 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-11 17:16 +0000 [r372864] dlee : + + * Makefile, /: Corrects the astsbindir setting when installing the + sample asterisk.conf. (closes issue ASTERISK-20406) ........ + Merged revisions 372863 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-10 20:59 +0000 [r372795-372806] Kinsey Moore + + * /, channels/chan_iax2.c: Ensure iax2 debug output is displayed + when expected When IAX2 debug was changed from iax_showframe to + iax_outputframe, some instances were missed (or added afterward). + This was causing debug output to not be displayed when expected. + (closes issue ASTERISK-20338) Reported-by: John Covert Patch-by: + John Covert ........ Merged revisions 372804 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372805 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_jingle.c, include/asterisk/doxygen/architecture.h, + main/devicestate.c, channels/chan_gtalk.c, res/res_jabber.c: + Deprecate chan_gtalk, chan_jingle, and res_jabber chan_gtalk, + chan_jingle, and res_jabber are now deprecated in favor of using + chan_motif and res_xmpp. They are a feature-equivalent + replacement and are written to be more easily maintainable. + (closes issue ASTERISK-20298) Review: + https://reviewboard.asterisk.org/r/2082/ Reported-by: Leif Madsen + +2012-09-10 19:19 +0000 [r372777] dlee : + + * res/res_rtp_asterisk.c: res_rtp_asterisk: Eliminate "type-punned + pointer" build warning. Removes "res_rtp_asterisk.c:706: warning: + dereferencing type-punned pointer will break strict-aliasing + rules" warning from the build on 32-bit platforms. The problem is + that 'size' was referenced aliased to both (pj_size_t *) and + (pj_ssize_t *). Now just make a copy of size that is the right + type so there isn't any pointer aliasing happening. It also adds + comments and asserts regarding what looks like an inappropriate + use of pj_sock_sendto, but is actually totally fine. (closes + issue ASTERISK-20368) Reported by: Shaun Ruffel Tested by: + Michael L. Young Patches: + 0001-res_rtp_asterisk-Eliminate-type-punned-pointer-build.patch + uploaded by Shaun Ruffel (license 5417) slightly modified by + David M. Lee. + +2012-09-10 18:50 +0000 [r372768] Jonathan Rose + + * /, apps/app_meetme.c: app_meetme: Document that 'p' option will + continue in dialplan. (closes issue AST-991) Reported by John + Bigelow ........ Merged revisions 372765 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372767 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-10 18:37 +0000 [r372766] Kinsey Moore + + * /: Recorded merge of revisions 372764 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Warn on + CLI when UDPTL init fails This adds a CLI warning when a SDP + offer is rejected due to UDPTL initialization failure. + Previously, there was no indication of the reason for offer + rejection in this case. (closes issue ASTERISK-20357) + Reported-by: Francesco Usseglio Gaudi ........ Merged revisions + 372763 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2012-09-10 17:33 +0000 [r372754] Jonathan Rose + + * main/channel.c, /: Masquerade: Retain parkinglot settings made by + CHANNEL function. Prior to this patch, the user would have a + parkinglot set on a channel that was parked and when the channel + was retrieved, any attempt by that channel to park would simply + use the default. This patch makes parkinglot values set in this + way be retained through the masquerade. (closes issue AST-990) + Reported by: Nick Huskinson Patches: + masquerade_parkinglot_patch.diff Uploaded by Jonathan Rose + (license 6182) ........ Merged revisions 372736 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372737 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-09 01:25 +0000 [r372711] Matthew Jordan + + * channels/sip/sdp_crypto.c, /: Only re-create an SRTP session when + needed In r356604, SRTP handling was fixed to accomodate multiple + crypto keys in an SDP offer and the ability to re-create an SRTP + session when the crypto keys changed. In certain circumstances - + most notably when a phone is put on hold after having been + bridged for a significant amount of time - the act of re-creating + the SRTP session causes problems for certain models of phones. + The patch committed in r356604 always re-created the SRTP session + regardless of whether or not the cryptographic keys changed. + Since this is technically not necessary, this patch modifies the + behavior to only re-create the SRTP session if Asterisk detects + that the remote key has changed. This allows models of phones + that do not handle the SRTP session changing to continue to work, + while also providing the behavior needed for those phones that do + re-negotiate cryptographic keys. (issue ASTERISK-20194) Reported + by: Nicolo Mazzon Tested by: Nicolo Mazzon Review: + https://reviewboard.asterisk.org/r/2099 ........ Merged revisions + 372709 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 372710 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-08 05:51 +0000 [r372696] dlee : + + * /, main/Makefile: Recorded merge of revisions 372695 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Add + OPENSSL_INCLUDE to the CFLAGS for ssl.c and tcptls.c. Without + this flag, those files will compile with the system installed + OpenSSL headers (if they exist). This is a real bummer if a + different path was specified using --with-ssl= (closes issue + ASTERISK-20392) ........ Merged revisions 372682 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2012-09-07 23:07 +0000 [r372622-372657] Richard Mudgett + + * /, main/astmm.c: Fix MALLOC_DEBUG version of ast_strndup(). + (closes issue ASTERISK-20349) Reported by: Brent Eagles ........ + Merged revisions 372655 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372656 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, funcs/func_math.c: Remove annoying unconditional debug message + from INC/DEC functions. (closes issue AST-1001) Reported by: + Guenther Kelleter ........ Merged revisions 372628 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372629 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_queue.c: Fix exception path typo in app_queue.c + try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy + Pepper Patches: fix-local-channel-locking.patch (license #6350) + patch uploaded by Jeremy Pepper ........ Merged revisions 372624 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 372625 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_voicemail.c, /: Fix VoicemailUserEntry event headers + ServerEmail and MailCommand reported values. The AMI action + VoicemailUsersList VoicemailUserEntry event headers ServerEmail + and MailCommand did not report the global values if they were not + overridden. The VoicemailUserEntry event header ServerEmail was + not populated with the global value if the voicemail user did not + override it. The VoicemailUserEntry event header MailCommand was + never populated with a value. * Removed unused struct ast_vm_user + member mailcmd[]. (closes issue AST-973) Reported by: John + Bigelow Tested by: rmudgett ........ Merged revisions 372620 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372621 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-07 21:04 +0000 [r372609-372611] dlee : + + * res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin, + res/pjproject/third_party/bin, res/pjproject/third_party/gsm/lib, + res/pjproject/lib, res/pjproject/pjlib/lib, + res/pjproject/third_party/gsm/bin, res/pjproject/pjnath/lib, + res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/lib, + res/pjproject/pjsip/bin, res/pjproject/pjsip-apps/bin, + res/pjproject/pjmedia/lib, res/pjproject/third_party/lib, + codecs/ilbc: svn:ignore cleanup. * pjproject bin and lib + directories should pretty much ignore everything * Ignore *.o in + codecs/ilbc + + * res/Makefile: Fix parallel make for res_asterisk_rtp. Fixes a + build regression introduced in r369517 "Add support for + ICE/STUN/TURN in res_rtp_asterisk and chan_sip." [1]. [1] + http://svnview.digium.com/svn/asterisk?view=revision&revision=369517 + When compiling asterisk in parallel like: $ make -j 10 It's + possible to get errors like the following: + .pjlib-util-test-x86_64-unknown-linux-gnu.depend:120: *** missing + separator. Stop. make[4]: *** [depend] Error 2 make[3]: *** [dep] + Error 1 make[2]: *** + [/home/sruffell/asterisk-working/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a] + Error 2 make[3]: warning: jobserver unavailable: using -j1. Add + `+' to parent make rule. This is because the build system is + trying to build each of the libraries in pjproject in parallel. + Now the build will build pjproject in a single job and link the + results into res_asterisk_rtp. Parallel builds, on one test + system, saves ~1.5 minutes from a default Asterisk build: Single + job: $ git clean -fdx >/dev/null && time ( ./configure >/dev/null + 2>&1 && make >/dev/null 2>&1 ) real 2m34.529s user 1m41.810s sys + 0m15.970s Parallel make: $ git clean -fdx >/dev/null && time ( + ./configure >/dev/null 2>&1 && make -j10 >/dev/null 2>&1 ) real + 1m2.353s user 2m39.120s sys 0m18.850s (closes issue + ASTERISK-20362) Reported by: Shaun Ruffel Patches: + 0001-res_asterisk_rtp-Fix-build-error-when-using-parallel.patch + uploaded by Shaun Ruffel (License #5417) + +2012-09-07 02:26 +0000 [r372531-372583] Matthew Jordan + + * /, apps/app_minivm.c: Free ast_str objects when temp file fails + to be created in MiniVM The previous commit (r372554) was from a + patch that was written before r366880, which ensured that ast_str + objects allocated in the sendmail routine were free'd in off + nominal paths. This commit frees the string objects in the off + nominal path introduced in r372554. (issue ASTERISK-17133) + Reported by: Tzafrir Cohen ........ Merged revisions 372581 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372582 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_minivm.c: Fix file descriptor leak and pointer scope + issue in MiniVM when sending mail When MiniVM sends an e-mail and + it has the volgain option set, it will spawn sox in a separate + process to handle the manipulation of the sound file. In doing + so, it creates a temporary file. There are two problems here: 1) + The file descriptor returned from mkstemp is leaked 2) The + finalfilename character pointer points to a buffer that loses + scope once volgain processing is finished. Note that in r316265, + Russell fixed some gcc warnings by using the return value of the + mkstemp call. A warning was placed in minivm that the file + descriptor was going to be leaked. This patch reverts that + change, as it handles the leak and 'uses' the file descriptor + returned from mkstemp. (closes issue ASTERISK-17133) Reported by: + Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir + Cohen (license #5035) ........ Merged revisions 372554 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372555 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_queue.c: Update QueueMemberStatus event documentation to + include member status values The Status: header in a + QueueMemberStatus event (and other QueueMember* events) is the + numeric value of the device state corresponding to that Queue + Member. As those values are not exactly obvious, listing them in + the documentation is useful. Matt Riddell reported this + indirectly through the wiki page. (closes issue ASTERISK-20243) + Reported by: Matt Riddell + +2012-09-06 22:12 +0000 [r372523] Richard Mudgett + + * /, channels/sig_pri.c: Fix loss of MOH on an ISDN channel when + parking a call for the second time. Using the AMI redirect action + to take an ISDN call out of a parking lot causes the MOH state to + get confused. The redirect action does not take the call off of + hold. When the call is subsequently parked again, the call no + longer hears MOH. * Make chan_dahdi/sig_pri restart MOH on + repeated AST_CONTROL_HOLD frames if it is already in a state + where it is supposed to be sending MOH. The MOH may have been + stopped by other means. (Such as killing the generator.) This + simple fix is done rather than making the AMI redirect action + post an AST_CONTROL_UNHOLD unconditionally when it redirects a + channel and thus potentially breaking something with an + unexpected AST_CONTROL_UNHOLD. (closes issue ABE-2873) Patches: + jira_abe_2873_c.3_bier.patch (license #5621) patch uploaded by + rmudgett ........ Merged revisions 372521 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + ........ Merged revisions 372522 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-06 21:42 +0000 [r372519] Kinsey Moore + + * /, apps/app_queue.c: Ensure listed queues are not offered for + completion When using tab-completion for the list of queues on + "queue reset stats" or "queue reload + {all|members|parameters|rules}", the tab-completion listing for + further queues erroneously listed queues that had already been + added to the list. The tab-completion listing now only displays + queues that are not already in the list. (closes issue AST-963) + Reported-by: John Bigelow ........ Merged revisions 372517 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372518 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-06 18:55 +0000 [r372500] dsessions : + + * channels/chan_sip.c, configs/res_ldap.conf.sample: LDAP Realtime + Peers Cannot Register Prior to 1.8, it was not necessary for an + explicit "type" to be set for an asterisk LDAP realtime peer. Now + the routine find_peer actually checks the type field during + registration and fails to find the peer if it is not set. The + attached patches make the realtime type equal whatever type is + being searched for if the type is 0 upon return from routine + build_peer. (closes issue ASTERISK-17222) Reported by: John + Covert Patch by: David Vossel Tested by: Darren Sessions Review: + https://reviewboard.asterisk.org/r/2095/ + +2012-09-06 15:56 +0000 [r372473] Jonathan Rose + + * /, UPGRADE-1.8.txt: chan_sip: Note change in behavior to how + directmediapermit/deny ACL works r366547 introduced a change to + the directmedia ACL for chan_sip which modified the behavior + significantly. Prior to the patch, this option would bridge peers + with directmedia if a peer's IP address matched its own + directmedia ACL. After that patch, the peer would check the + bridged peer's ACL instead. This change has been present since + 1.8.14.0. That patched failed to document the change in + Upgrade.txt, so this patch adds mention of that change to + UPGRADE.txt (UPGRADE-1.8.txt in newer branches) (issue AST-876) + ........ Merged revisions 372471 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372472 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-06 14:30 +0000 [r372446] Kinsey Moore + + * /, apps/app_queue.c: Ensure "rules" is tab-completable for "queue + show" Previously, tabbing at the end of "queue show" produced a + list of available queues about which information could be shown, + but did not include an alternative command, "rules", to access + information about queue rules. The "rules" item should now be + shown in the list of tab-completable items. (closes issue + AST-958) Reported-by: John Bigelow ........ Merged revisions + 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 372445 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-06 02:50 +0000 [r372392-372419] Matthew Jordan + + * /, pbx/pbx_dundi.c: Fix DUNDi message routing bug when + neighboring peer is unreachable Consider a scenario where DUNDi + peer PBX1 has two peers that are its neighbors, PBX2 and PBX3, + and where PBX2 and PBX3 are also neighbors. If the connection is + temporarily broken between PBX1 and PBX3, PBX1 should not include + PBX3 in the list of peers it sends to PBX2 in a DPDISCOVER + message, as it cannot send messages to PBX3. If it does, PBX2 + will assume that PBX3 already received the message and fail to + forward the message on to PBX3 itself. This patch fixes this by + only including peers in a DPDISCOVER message that are reachable + by the sending node. This includes all peers with an empty + address (00:00:00:00:00:00) and that are have been reached by a + qualify message. This patch also prevents attempting to qualify a + dynamic peer with an empty address until that peer registers. + (closes issue ASTERISK-19309) Reported by: Peter Racz patches: + dundi_routing.patch uploaded by Peter Racz (license 6290) The + patch uploaded by Peter was modified slightly for this commit. + ........ Merged revisions 372417 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372418 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_followme.c: Allow configured numbers for FollowMe to + be greater than 90 characters When parsing a 'number' defined in + followme.conf, FollowMe previously parsed the number in the + configuration file into a buffer with a length of 90 characters. + This can artificially limit some parallel dial scenarios. This + patch allows for numbers of any length to be defined in the + configuration file. Note that Clod Patry originally wrote a patch + to fix this problem and received a Ship It! on the JIRA issue. + The patch originally expanded the buffer to 256 characters. + Instead, the patch being committed duplicates the string in the + config file on the stack before parsing it for consumption by the + application. (closes issue ASTERISK-16879) Reported by: Clod + Patry Tested by: mjordan patches: followme_no_limit.diff uploaded + by Clod Patry (license #5138) Slightly modified for this commit. + ........ Merged revisions 372390 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372391 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-05 19:43 +0000 [r372373] Richard Mudgett + + * main/dsp.c, /: Fix compile error. ........ Merged revisions + 372372 from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-05 19:24 +0000 [r372365] Kinsey Moore + + * main/manager.c, /: Correct documentation for ModuleLoad AMI + action The documentation incorrectly listed 'rtp' as a reloadable + subsystem and left out many other reloadable subsystems. It is + now also documented that subsystems may only be reloaded, not + loaded or unloaded. (closes issue AST-977) Reported-by: John + Bigelow ........ Merged revisions 372354 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372358 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-05 18:46 +0000 [r372342] Alec L Davis + + * main/dsp.c, /: dsp.c: in ast_mf_detect_init incorrectly sets + goertzel samples to 160, should be MF_GSIZE Related + https://reviewboard.asterisk.org/r/2097/ ........ Merged + revisions 372339 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372341 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-05 18:36 +0000 [r372340] Kinsey Moore + + * main/pbx.c, /: Ensure counts generated in + manager_show_dialplan_helper are correct When + manager_show_dialplan_helper was written, the counter increment + for the total number of contexts was placed with the extensions + increment instead of in the enclosing loop. This function should + now generate correct context counts. (closes issue AST-970) + Reported-by: John Bigelow ........ Merged revisions 372337 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372338 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-05 17:35 +0000 [r372327-372328] Richard Mudgett + + * res/res_rtp_asterisk.c: Fix coding guidelines issue with a recent + commit. + + * res/res_rtp_asterisk.c: Fix RTP/RTCP read error message + confusion. The RTP/RTCP read error message can report "fail: + success" when the read failure is because of an ICE failure. * + Changed __rtp_recvfrom() to generate a PJ ICE message when ICE + fails. * Changed RTP/RTCP read error message to indicate an + unspecified error when errno is zero. (closes issue + ASTERISK-20288) Reported by: Joern Krebs Patches: + jira_asterisk_20288_err_msg.patch (license #5621) patch uploaded + by rmudgett (modified) + +2012-09-05 16:04 +0000 [r372311] Mark Michelson + + * res/res_rtp_asterisk.c, main/rtp_engine.c, + include/asterisk/rtp_engine.h: Re-fix sending unnegotiated + payloads during a P2P RTP bridge. The previous fix still would + look in the static_RTP_PT table, which is inappropriate since we + specifically want to find a codec that has been negotiated. + (closes issue ASTERISK-20296) reported by NITESH BANSAL Patches: + codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418) + +2012-09-05 13:47 +0000 [r372289] Matthew Jordan + + * apps/app_voicemail.c, /: Fix memory leaks in app_voicemail when + using IMAP storage or realtime config This patch fixes two memory + leaks: 1. When find_user is called with NULL as its first + parameter, the voicemail user returned is allocated on the heap. + The inboxcount2 function uses find_user in such a fashion when + counting new messages, and fails to free the resulting voicemail + user object. 2. When populate_defaults is called on a voicemail + user, it wipes whatever flags have been set on the object by + copying over the global flags object. If the VM_ALLOCED flag was + ste on the voicemail user prior to doing so, that flag is + removed. This leaks the voicemail user when free_user is later + called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek + patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277) + Patch slightly modified for this commit. Review: + https://reviewboard.asterisk.org/r/2096 ........ Merged revisions + 372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 372288 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-05 12:17 +0000 [r372266] Michael L. Young + + * res/res_rtp_asterisk.c: Fix breakage caused by last merge. + Missing a variable for 11 and trunk. + +2012-09-05 07:41 +0000 [r372214-372241] Alec L Davis + + * main/dsp.c, /: dsp.c: Fix multiple issues when no-interdigit + delay is present, and fast DTMF 50ms/50ms Revert DTMF hit/miss + detector to original -r349249 method with some changes, remove + unnecessary; 1. reseting of hits=0, when no signal, only need to + set it once. 2. incrementing of hits, when the hit is the same as + the current hit. 3. setting of lasthit, when it's the same as + before. Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3 & 3 + spelling mistakes (closes issue ASTERISK-19610) alecdavis + (license 585) Reported by: Jean-Philippe Lord Tested by: + alecdavis Review: https://reviewboard.asterisk.org/r/2085/ + ........ Merged revisions 372239 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372240 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/dsp.c, /: dsp.c: optimize goerztzel sample loops, in + dtmf_detect, mf_detect and tone_detect use a temporary short int + when repeatedly used to call goertzel_sample. alecdavis (license + 585) Reported by: alecdavis Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/2093/ ........ Merged + revisions 372212 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372213 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-05 04:52 +0000 [r372199] Michael L. Young + + * res/res_rtp_asterisk.c, /: Fix Incrementing Sequence Number For + Retransmitted DTMF End Packets In Asterisk 1.4+, a fix was put in + place to increment the sequence number for retransmitted DTMF end + packets. With the introduction of the RTP engine API in 1.8, the + sequence number was no longer being incremented. This patch fixes + this regression as well as cleans up a few lines that were not + doing anything. (closes issue ASTERISK-20295) Reported by: Nitesh + Bansal Tested by: Michael L. Young Patches: + 01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license + 6418) asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L. + Young (license 5026) Review: + https://reviewboard.asterisk.org/r/2083/ ........ Merged + revisions 372185 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372198 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-05 02:25 +0000 [r372175] Matthew Jordan + + * cel/cel_pgsql.c, /: Fix memory leak when CEL is successfully + written to PostgreSQL database PQClear is not called when the + result object of a call to PQExec has a status of + PGRES_COMMAND_OK. Interestingly enough, the off nominal case was + handled properly, so this memory leak only occurred when CEL + records were successfully written. This patch properly clears the + result in the nominal code path. (closes issue ASTERISK-19991) + Reported by: Etienne Lessard Tested by: Etienne Lessard patches: + mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license + #6394) ........ Merged revisions 372158 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372165 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-09-04 15:48 +0000 [r372135-372137] Mark Michelson + + * channels/chan_sip.c: Fix issue where SIP devices were not + notified when custom devices changed to "ringing". The problem + had to do with logic used when checking for what the oldest + ringing channel was. The problem was that if no channel was + found, then no notification would be sent. For custom device + states, there is no associated channel, so no notification would + get sent. This fixes the issue by still sending the notification + even if no associated channel can be found for a ringing device + state change. (closes issue ASTERISK-20297) Reported by Noah + Engelberth + + * main/config_options.c, apps/app_confbridge.c: Prevent crash from + using app_page with no confbridge.conf file provided. Also + prevents other potential crashes when using aco API with + uninitialized aco_info structs. (closes issue ASTERISK-20305) + reported by Noah Engelberth Tested by Noah Engelberth Review: + https://reviewboard.asterisk.org/r/2086 + +2012-08-31 21:14 +0000 [r372118] Mark Michelson + + * res/res_rtp_asterisk.c: Prevent local RTP bridges from sending + inappropriate formats to participants. A change for Asterisk 11 + caused a check for failure to incorrectly check the return value. + This resulted in the possibility of transmitting media that a + party had not negotiated. If this media happened to be G.729, + then this could potentially result in one-way audio if no G.729 + translators are installed. (closes issue ASTERISK-20296) reported + by NITESH BANSAL + +2012-08-30 20:54 +0000 [r372050-372091] Mark Michelson + + * /, apps/app_queue.c: Prevent crash on shutdown due to refcount + error on queues container. When app_queue is unloaded, the queues + container has its refcount decremented, potentially to 0. Then + the taskprocessor responsible for handling device state changes + is unreferenced. If the taskprocessor happens to be just about to + run its task, then it will create and destroy an iterator on the + queues container. This can cause the refcount on the queues + container to increase to 1 and then back to 0. Going back to 0 a + second time results in double frees. This failure was seen + periodically in the testsuite when Asterisk would shut down. + ........ Merged revisions 372089 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 372090 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_queue.c: Help prevent ringing queue members from + being rung when ringinuse set to no. Queue member status would + not always get updated properly when the member was called, thus + resulting in the member getting multiple calls. With this change, + we update the member's status at the time of calling, and we also + check to make sure the member is still available to take the call + before placing an outbound call. (closes issue ASTERISK-16115) + reported by nik600 Patches: app_queue.c-svn-r370418.patch + uploaded by Italo Rossi (license #6409) ........ Merged revisions + 372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 372049 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-30 16:24 +0000 [r371963-372028] Matthew Jordan + + * channels/chan_iax2.c: AST-2012-013: Resolve ACL rules being + ignored during calls by some IAX2 peers When an IAX2 call is made + using the credentials of a peer defined in a dynamic Asterisk + Realtime Architecture (ARA) backend, the ACL rules for that peer + are not applied to the call attempt. This allows for a remote + attacker who is aware of a peer's credentials to bypass the ACL + rules set for that peer. This patch ensures that the ACLs are + applied for all peers, regardless of their storage mechanism. + (closes issue ASTERISK-20186) Reported by: Alan Frisch Tested by: + mjordan, Alan Frisch + + * /: Block r372020 + + * main/manager.c, /, README-SERIOUSLY.bestpractices.txt: + AST-2012-012: Resolve AMI User Unauthorized Shell Access through + ExternalIVR The AMI Originate action can allow a remote user to + specify information that can be used to execute shell commands on + the system hosting Asterisk. This can result in an unwanted + escalation of permissions, as the Originate action, which + requires the "originate" class authorization, can be used to + perform actions that would typically require the "system" class + authorization. Previous attempts to prevent this permission + escalation (AST-2011-006, AST-2012-004) have sought to do so by + inspecting the names of applications and functions passed in with + the Originate action and, if those applications/functions matched + a predefined set of values, rejecting the command if the user + lacked the "system" class authorization. As noted by IBM X-Force + Research, the "ExternalIVR" application is not listed in the + predefined set of values. The solution for this particular + vulnerability is to include the "ExternalIVR" application in the + set of defined applications/functions that require "system" class + authorization. Unfortunately, the approach of inspecting fields + in the Originate action against known applications/functions has + a significant flaw. The predefined set of values can be bypassed + by creative use of the Originate action or by certain dialplan + configurations, which is beyond the ability of Asterisk to + analyze at run-time. Attempting to work around these scenarios + would result in severely restricting the applications or + functions and prevent their usage for legitimate means. As such, + any additional security vulnerabilities, where an + application/function that would normally require the "system" + class authorization can be executed by users with the "originate" + class authorization, will not be addressed. Instead, the + README-SERIOUSLY.bestpractices.txt file has been updated to + reflect that the AMI Originate action can result in commands + requiring the "system" class authorization to be executed. Proper + system configuration can limit the impact of such scenarios. + (closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM + X-Force Research ........ Merged revisions 371998 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371999 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * doc/CODING-GUIDELINES (added), /: Restore CODING-GUIDELINES to + doc folder In r294740, the CODING-GUIDELINES was removed from the + doc folder in favor of the content on the Asterisk wiki. Some + folks still look in the doc folder initially for coding guideline + suggestions; as such, this patch adds a CODING-GUIDELINES file + back into the doc folder. The content of the file merely points + to the correct page on the Asterisk wiki where the coding + guidelines currently live. (closes issue ASTERISK-20279) Reported + by: Andrew Latham Patches: CODING-GUIDELINES.diff uploaded by + Andrew Latham (license 5985) ........ Merged revisions 371961 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 371962 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-29 22:38 +0000 [r371950] Richard Mudgett + + * apps/app_meetme.c: Fix compile errors. + +2012-08-29 21:07 +0000 [r371921] Jonathan Rose + + * /, apps/app_meetme.c: app_meetme: Adding test events for + following activity in MeetMe. ........ Merged revisions 371919 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 371920 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-29 19:56 +0000 [r371862-371893] Richard Mudgett + + * main/channel.c: Fix theoretical compile error with HAVE_EPOLL. + Really shows how much epoll is used since it had not been + reported yet. + + * main/channel.c, /: Initialize file descriptors for dummy channels + to -1. Dummy channels usually aren't read from, but functions + like SHELL and CURL use autoservice on the channel. (closes issue + ASTERISK-20283) Reported by: Gareth Palmer Patches: + svn-371580.patch (license #5169) patch uploaded by Gareth Palmer + (modified) ........ Merged revisions 371888 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371890 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_dial.c, /: Fix hangup cause passthrough regression. The + v1.8 -r369258 change to fix the F and F(x) action logic + introduced a regression in passing the hangup cause from the + called channel to the caller channel. (closes issue + ASTERISK-20287) Reported by: Konstantin Suvorov Patches: + app_dial_hangupcause.patch (license #6421) patch uploaded by + Konstantin Suvorov (modified) Tested by: rmudgett ........ Merged + revisions 371860 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371861 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-29 17:25 +0000 [r371845] Jonathan Rose + + * /, channels/chan_sip.c: chan_sip: Send 408 on retransmit timeout + instead of 603 (closes issue ASTERISK-20124) Reported by: Walter + Doekes ........ Merged revisions 371824 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371825 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-27 21:50 +0000 [r371784-371790] Mark Michelson + + * configs/agents.conf.sample, /: Fix misleading documentation in + agents.conf.sample regarding ackcall usage. The documentation + made it sound as if the DTMF acknowledgment was needed at the + time the agent logs in, rather than when the agent is called. + This is likely a relic from the days when there were multiple + ways of logging in agents. (closes issue AST-962) reported by + Steve Pitts ........ Merged revisions 371787 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371789 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/manager.c, /: Fix incorrect documentation of the + MailboxStatus manager command. The "Waiting" field was + misdocumented as reporting the number of messages waiting. In + reality, it simply indicated the presence or absence of waiting + messages. ........ Merged revisions 371782 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371783 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-27 18:14 +0000 [r371753] dlee : + + * res/pjproject/pjlib-util/bin, res/pjproject/pjnath/build/output, + res/pjproject/pjlib/bin, res/pjproject/pjlib-util/build/output, + res/pjproject/pjnath/bin, res/pjproject/pjlib/build/output: + svn:ignore pjproject bin & output for all platforms. + +2012-08-27 17:51 +0000 [r371749-371750] Mark Michelson + + * /, configs/queues.conf.sample: Fix incorrectly documented option + in queues.conf sharedlastcall defaults to "no" not "yes" (closes + issue AST-979) reported by Steve Pitts ........ Merged revisions + 371747 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 371748 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /: Re-add merge and block properties. + +2012-08-27 16:55 +0000 [r371720] dlee : + + * main/lock.c, /: Fixes ast_rwlock_timed[rd|wr]lock for BSD and + variants. The original implementations simply wrap pthread + functions, which take absolute time as an argument. The spinlock + version for systems without those functions treated the argument + as a delta. This patch fixes the spinlock version to be + consistent with the pthread version. (closes issue + ASTERISK-20240) Reported by: Egor Gorlin Patches: lock.c.patch + uploaded by Egor Gorlin (license 6416) ........ Merged revisions + 371718 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2012-08-27 14:07 +0000 [r371692] Kinsey Moore + + * /, main/utils.c: Implement workaround for BETTER_BACKTRACES crash + When compiling with BETTER_BACKTRACES enabled, Asterisk will + sometimes crash when "core show locks" is run. This happens + regularly in the testsuite since several tests run "core show + locks" to help with debugging. This seems to be a fault with + libraries on certain operating systems (notably CentOS 6.2/6.3) + running on virtual machines and utilizing gcc 4.4.6. (closes + issue ASTERISK-20090) ........ Merged revisions 371690 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371691 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-26 23:07 +0000 [r371664] Alec L Davis + + * main/dsp.c, /: mf_detect: incorrectly used DTMF_GSIZE instead of + MF_GSIZE ........ Merged revisions 371662 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371663 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-22 15:54 +0000 [r371619] Joshua Colp + + * channels/chan_motif.c: Add support for call-id logging to + chan_motif. Review: https://reviewboard.asterisk.org/r/2077/ + +2012-08-21 20:54 +0000 [r371592] Mark Michelson + + * cdr/cdr_tds.c, main/xmldoc.c, apps/app_dial.c, + channels/chan_dahdi.c, /, channels/chan_sip.c, funcs/func_odbc.c, + main/file.c, main/utils.c, apps/app_queue.c, pbx/pbx_config.c, + res/res_jabber.c, apps/app_stack.c, channels/chan_oss.c, + res/res_config_sqlite.c: Fix misuses of asprintf throughout the + code. This fixes three main issues * Change asprintf() uses to + ast_asprintf() so that it pairs properly with ast_free() and no + longer causes MALLOC_DEBUG to freak out. * When ast_asprintf() + fails, set the pointer NULL if it will be referenced later. * Fix + some memory leaks that were spotted while taking care of the + first two points. (Closes issue ASTERISK-20135) reported by + Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071 + ........ Merged revisions 371590 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371591 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-20 20:09 +0000 [r371571] Mark Michelson + + * res/res_rtp_asterisk.c: Use thread-local storage to store + pj_thread_descs. pj_thread_register() takes a parameter of type + pj_thread_desc. It was assumed that pj_thread_register either + used this item temporarily or made a copy of it. Unfortunately, + all it does is keep a pointer to the structure in thread-local + storage. This means that if our pj_thread_desc goes out of scope, + then pjlib will be referencing bogus data quite often, most + commonly on operations involving a pj_mutex_t. In our case, our + pj_thread_desc was on the stack and went out of scope very + shortly after registering our thread with pjlib. With this + change, the pj_thread_desc is stored in thread-local storage so + the pointer that pjlib keeps in thread-local storage will + reference legitimate memory. (closes issue ASTERISK-20237) + reported by Jeremy Pepper Patches: ASTERISK-20237.patch uploaded + by Mark Michelson (license #5049) Tested by Jeremy Pepper + +2012-08-20 15:34 +0000 [r371546] Kinsey Moore + + * main/udptl.c, /: Ignore recovered zero-length secondary UDPTL + packets In some cases, recovering lost packets using the + secondary packet recovery mechanism with UDPTL/T.38 can result in + the recovery of zero-length packets. These must be ignored or the + frame generated from them can cause segfaults and allocation + failures. (closes issue ASTERISK-19762) (closes issue + ASTERISK-19373) Reported-by: Benjamin (bulkorok) Reported-by: Rob + Gagnon (rgagnon) ........ Merged revisions 371544 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371545 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-18 02:35 +0000 [r371492-371530] Matthew Jordan + + * /: Recorded merge of revisions 371529 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Remove + old debug code from http configuration loading (closes issue + ASTERISK-20254) Reported by: Andrew Latham Patches: http.diff + uploaded by Andrew Latham (license #5985) + + * main/http.c: Remove old debug code from http configuration + loading (closes issue ASTERISK-20254) Reported by: Andrew Latham + Patches: http.diff uploaded by Andrew Latham (license #5985) + + * res/res_xmpp.c: Fix typo in JabberSend that looked for '2' + instead of '@' in recipient argument The summary says about all + there is to say. (closes issue ASTERISK-20239) Reported by: + Gregory Porras + + * funcs/func_hangupcause.c: Make the name of the "HangupCauseClear" + application consistent The name of the "HangupCauseClear" + application is "HangupCauseClear", not "HangupcauseClear". The + incorrect case of 'cause' caused the XML documentation to not + register properly. As an aside, this commit message felt very + awkward, but I'm not sure how else to note that "X", which has to + be "X", was referred to as "x". (closes issue ASTERISK-20253) + Reported by: Andrew Latham Patches: hangupcause.diff uploaded by + Andrew Latham (license #5985) + + * build_tools/cflags.xml, utils/utils.xml, res/res_fax.c, + sounds/sounds.xml, res/res_curl.c: Update module support level on + a variety of modules and compiler options Some core support + modules and compiler options were no longer tagged with a module + support level. This patch adds 'core' back to those options. Note + that this patch modifies a few of the patches provided by Andrew + Latham slightly. res_curl and res_fax are both 'core' supported + modules. (closes issue ASTERISK-20215) Reported by: Andrew Latham + Tested by: mjordan Patches: astcanary.diff (license #5985) + uploaded by Andrew Latham cflagsxml.diff (license #5985) uploaded + by Andrew Latham curl_fax.diff (license #5985) uploaded by Andrew + Latham soundsxml.diff (license #5985) uploaded by Andrew Latham + + * main/xmldoc.c, /: Fix memory leak in XML documentation When + formatting documentation fields, the XML documentation parser + calls xmldoc_get_formatted. This function allocates a string + buffer at the beginning of its routine. Unfortunately, on certain + code paths, it also calls xmldoc_string_cleanup, which assumes + that it will create the string buffer. The previously allocated + string buffer is then leaked by the xmldoc_string_cleanup + routine. Now: we don't do that. (closes issue AST-932) Reported + by: Alexander Homig ........ Merged revisions 371469 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371491 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-17 19:49 +0000 [r371482] Joshua Colp + + * channels/chan_sip.c: When a peer registers using WebSocket do not + resolve the Contact provided. (closes issue ASTERISK-20238) + Reported by: james.mortensen + +2012-08-17 15:58 +0000 [r371438] Kinsey Moore + + * main/loader.c, /: Add instrumentation to subsystem reloads When + Asterisk is built with TEST_FRAMEWORK defined, Asterisk will now + generate TestEvent AMI events on subsystem reloads such as cdr, + dnsmgr, extconfig, etc. (issue PQ-1126) ........ Merged revisions + 371436 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 371437 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-17 12:24 +0000 [r371426] Joshua Colp + + * res/res_format_attr_h264.c: Add some additional H.264 attributes, + "max-smbps" and "max-fps", for passthrough. (closes issue + ASTERISK-20206) Reported by: ddkprog Patches: + res_format_attr_h264.c.diff uploaded by ddkprog (license 6008) + +2012-08-17 12:23 +0000 [r371425] Russell Bryant + + * res/res_rtp_asterisk.c: rtp: Ensure defaults are set without + rtp.conf. While building up a new install to test chan_motif, I + ran into a failure due to icesupport being disabled. This was due + to me not having an rtp.conf. It was intended in the code for it + to be enabled by default, but it was only applied if rtp.conf + existed. This patch updates res_rtp_asterisk to be consistent in + how it handles defaults. A few options didn't have their default + values set globally, including icesupport. They are now set and + icesupport is enabled by default, even if you do not have an + rtp.conf. + +2012-08-16 23:02 +0000 [r371399] Terry Wilson + + * main/config.c, /: Handle integer over/under-flow in + ast_parse_args The strtol family of functions will return + *_MIN/*_MAX on overflow. To detect when an overflow has happened, + errno must be set to 0 before calling the function, then checked + afterward. (closes issue ASTERISK-20120) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/2073/ ........ Merged + revisions 371392 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371398 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-16 22:44 +0000 [r371395] Kinsey Moore + + * main/loader.c, /: Add module reload instrumentation for + TEST_FRAMEWORK This adds AMI events for module reloads when + Asterisk is built with TEST_FRAMEWORK enabled and corrects + generation of the module load AMI event. (issue PQ-1126) ........ + Merged revisions 371393 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371394 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-16 19:43 +0000 [r371355-371382] Jonathan Rose + + * /, channels/chan_sip.c: chan_sip: Use pvt outgoing_call variable + to set Remote-Party-ID Header Previously the pvt SIP_OUTGOING + flag was used instead, which will frequently flip during + reinvites. (closes issue AST-897) Reported by: Thomas Arimont + ........ Merged revisions 371357 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371358 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: chan_sip: Trigger reinvite if the SDP + answer is included in the SIP ACK Under certain conditions, a SIP + transaction involving directmedia wouldn't trigger a re-invite + because the SDP answer was included in an ACK instead of in a + message that we would have triggered the invite with. This patch + just queues a source change control frame if the dialog is using + directmedia when we find sdp for an ACK. (closes issue AST-913) + Reported by: Thomas Arimont ........ Merged revisions 371337 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371338 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-15 23:28 +0000 [r371324] Mark Michelson + + * /, apps/app_queue.c: Fix bug where final queue member would not + be removed from memory. If a static queue had realtime members, + then there could be a potential for those realtime members not to + be properly deleted from memory. If the queue's members were + loaded from realtime and then all the members were deleted from + the backend, then the queue would still think these members + existed. The reason was that there was a short- circuit in code + such that if there were no members found in the backend, then the + queue would not be updated to reflect this. Note that this only + affected static queues with realtime members. Realtime queues + with realtime members were unaffected by this issue. (closes + issue ASTERISK-19793) reported by Marcus Haas ........ Merged + revisions 371306 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371313 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-15 20:40 +0000 [r371295] Michael L. Young + + * channels/chan_sip.c: Fix Segfault When Registering SIP Over + WebSockets The helper function, get_address_family_filter, in + chan_sip for dns resolution by address family was not recognizing + the websockets transport and resulting in a null pointer being + sent to functions in netsock2, in an attempt to determine if we + are bound to ANY address ([::]) or not. This patch fixes this + issue by handling the transport types SIP_TRANSPORT_WS and + SIP_TRANSPORT_WSS which results in a sock address being set + properly for use in determining the address family. (closes issue + ASTERISK-20221) Reported by: Sven Beisiegel Tested by: Sven + Beisiegel, James Mortensen Patches: + asterisk-20221-ws-family-filter.diff uploaded by Michael L. Young + (license 5026) + +2012-08-15 20:17 +0000 [r371258-371272] Kinsey Moore + + * /, channels/chan_sip.c: Avoid unconditional NULLing of mwipvt on + relatedpeer on SIP dialog destruction The other instance of this + bug was fixed by jcolp/file in r121496. If we are destroying a + dialog only set the MWI dialog pointer on the related peer to + NULL if it is the dialog currently being destroyed. (closes issue + ASTERISK-20119) Patch-by: Misha Vodsedalek ........ Merged + revisions 371270 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371271 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/sig_ss7.c, channels/chan_dahdi.c, channels/sig_analog.c, + channels/chan_sip.c, channels/chan_iax2.c, channels/sig_pri.c: + Add HANGUPCAUSE information to callee channels This adds + HANGUPCAUSE information to called channels so that hangup + handlers can, in conjunction with predial dialplan execution, + access the hangupcause information when the dialed channel hangs + up on a one-to-one basis instead of a many-to-one basis as with + HANGUPCAUSE usage on the caller channel. Review: + https://reviewboard.asterisk.org/r/2069/ (closes issue + ASTERISK-20198) + +2012-08-13 20:28 +0000 [r371227] Kinsey Moore + + * main/loader.c, /, apps/app_meetme.c: Add test instrumentation + This adds test instrumentation for loading and unloading of + modules and for certain actions in MeetMe to be used in the + testsuite or any other consumer of AMI events. These will only be + generated when Asterisk is built with TEST_FRAMEWORK enabled. + (issue PQ-1131) (issue PQ-1133) ........ Merged revisions 371201 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 371203 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-13 19:52 +0000 [r371200] Mark Michelson + + * /, channels/chan_sip.c: Fix problem where incorrect pointer was + checked for nullity. ........ Merged revisions 371198 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371199 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-10 22:03 +0000 [r371146] Richard Mudgett + + * CHANGES: Update CHANGES for private party ID. + +2012-08-10 21:32 +0000 [r371143] Mark Michelson + + * /, apps/app_queue.c: Fix a couple of documentation problems in + app_queue.c * The RemoveQueueMember app made mention of options + that could be passed in, but no options are supported. I have + removed the listing of options from the documentation. * The + RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value + that could be set. (closes issue AST-949) reported by Steve Pitts + (closes issue AST-954) reported by Steve Pitts ........ Merged + revisions 371141 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371142 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-10 20:08 +0000 [r371121] Matthew Jordan + + * / (added): _ _ _ _ _ _ / \ ___| |_ ___ _ __(_)___| | __ / | / | / + _ \ / __| __/ _ \ '__| / __| |/ / | | | | / ___ \__ \| | __/ | | + \__ \ < | | | | /_/ \_\___/\__\___|_| |_|___/_|\_\ |_| |_| + Because it's one greater than 10. + +2012-08-10 19:54 +0000 [r371120] Richard Mudgett + + * main/channel.c, channels/chan_misdn.c, channels/chan_sip.c, + main/channel_internal_api.c, main/features.c, + include/asterisk/channel.h, channels/sig_pri.c, + funcs/func_callerid.c, main/cli.c: Add private representation of + caller, connected and redirecting party ids. This patch adds the + feature "Private representation of caller, connected and + redirecting party ids", as previously discussed with us (DATUS) + and Digium. 1. Feature motivation Until now it is quite difficult + to modify a party number or name which can only be seen by + exactly one particular instantiated technology channel + subscriber. One example where a modified party number or name on + one channel is spread over several channels are supplementary + services like call transfer or pickup. To implement these + features Asterisk internally copies caller and connected ids from + one channel to another. Another example are extension + subscriptions. The monitoring entities (watchers) are notified of + state changes and - if desired - of party numbers or names which + represent the involving call parties. One major feature where a + private representation of party names is essentially needed, i.e. + where a party name shall be exclusively signaled to only one + particular user, is a private user-specific name resolution for + party numbers. A lookup in a private destination-dependent + telephone book shall provide party names which cannot be seen by + any other user at any time. 2. Feature Description This feature + comes along with the implementation of additional private party + id elements for caller id, connected id and redirecting ids + inside Asterisk channels. The private party id elements can be + read or set by the user using Asterisk dialplan functions. When a + technology channel is initiating a call, receives an internal + connected-line update event, or receives an internal redirecting + update event, it merges the corresponding public id with the + private id to create an effective party id. The effective party + id is then used for protocol signaling. The channel technologies + which initially support the private id representation with this + patch are SIP (chan_sip), mISDN (chan_misdn) and PRI + (chan_dahdi). Once a private name or number on a channel is set + and (implicitly) made valid, it is generally used for any further + protocol signaling until it is rewritten or invalidated. To + simplify the invalidation of private ids all internally generated + connected/redirecting update events and also all + connected/redirecting update events which are generated by + technology channels -- receiving regarding protocol information - + automatically trigger the invalidation of private ids. If not + using the private party id representation feature at all, i.e. if + using only the 'regular' caller-id, connected and redirecting + related functions, the current characteristic of Asterisk is not + affected by the new extended functionality. 3. User interface + Description To grant access to the private name and number + representation from the Asterisk dialplan, the CALLERID, + CONNECTEDLINE and REDIRECTING dialplan functions are extended by + the following data types. The formats of these data types are + equal to the corresponding regular 'non-private' already existing + data types: CALLERID: priv-all priv-name priv-name-valid + priv-name-charset priv-name-pres priv-num priv-num-valid + priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid + priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: + priv-name priv-name-valid priv-name-pres priv-name-charset + priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr + priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag + REDIRECTING: priv-orig-name priv-orig-name-valid + priv-orig-name-pres priv-orig-name-charset priv-orig-num + priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan + priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type + priv-orig-subaddr-odd priv-orig-tag priv-from-name + priv-from-name-valid priv-from-name-pres priv-from-name-charset + priv-from-num priv-from-num-valid priv-from-num-pres + priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid + priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag + priv-to-name priv-to-name-valid priv-to-name-pres + priv-to-name-charset priv-to-num priv-to-num-valid + priv-to-num-pres priv-to-num-plan priv-to-subaddr + priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd + priv-to-tag Reported by: Thomas Arimont Review: + https://reviewboard.asterisk.org/r/2030/ + +2012-08-10 17:56 +0000 [r371113] Mark Michelson + + * channels/chan_sip.c: Fix a comparison that was causing presence + tests to fail. A recent change made it so that device state + changes that were not actual "changes" would not get reported to + subscribers. The problem was that this inadvertently blocked + presence updates as well. + +2012-08-10 16:49 +0000 [r371059-371091] Alexandr Anikin + + * addons/chan_ooh323.c, /: remove ALREADYGONE flag on ooh323 call + data by ooh323_indicate (CONGESTION/BUSY) due to call hasn't gone + there really. This indication arrive from asterisk core not h.323 + stack (closes issue ASTERISK-19308) Reported by: Dmitry Melekhov + Patches: ASTERISK-19308.patch ........ Merged revisions 371089 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 371090 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * addons/ooh323c/src/ooGkClient.c, /: Send re-register packets by + GRQ (gatekeeper request) interval (close issue ASTERISK-20094) + Patches: ASTERISK-20094-2.patch ........ Merged revisions 371060 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 371061 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * addons/ooh323c/src/ooTimer.c: restore calling cb functions by + timer expire this was broken in rev 369602 + +2012-08-10 02:07 +0000 [r371052] Richard Mudgett + + * main/features.c: Fix pickup extension channel reference error. + You cannot unref a pointer and then expect to ref it again later. + * Fix potential NULL pointer deref if the call pickup search + fails. + +2012-08-09 21:35 +0000 [r371036-371043] Alexandr Anikin + + * addons/chan_ooh323.c: Introdue 'ooh323 show gk' cli command that + show status of connection to H.323 Gatekeeper (GkClient state) + + * addons/ooh323c/src/ooGkClient.c, /: Fix to resend GRQ/RRQ if RRJ + (registration reject) is received (close issue ASTERISK-20094) + Patches: ASTERISK-20094.patch ........ Merged revisions 371011 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 371022 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-09 19:22 +0000 [r371030] Richard Mudgett + + * channels/chan_dahdi.c, /, configure, + include/asterisk/autoconfig.h.in, configure.ac, + channels/sig_pri.c, channels/sig_ss7.c: Use better libss7 + detection test and move libpri compile test. ........ Merged + revisions 371012 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 371013 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-09 18:28 +0000 [r371010] Alexandr Anikin + + * /, addons/ooh323c/src/ooh323ep.c: change opening h323 logfile + with append mode instead of overwrite ........ Merged revisions + 370988 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 370989 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-09 17:40 +0000 [r370987] Kinsey Moore + + * /, apps/app_meetme.c: Correct documentation for the MeetMe x flag + The documentation for the x flag for MeetMe incorrectly described + its function as closing down the conference when the last marked + user left. It actually causes the users with that flag to leave + the conference when the last marked user exits. The functionality + of this flag is not changing. ........ Merged revisions 370985 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 370986 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-09 14:52 +0000 [r370979] Mark Michelson + + * main/pbx.c, channels/chan_sip.c, include/asterisk/pbx.h, + channels/sip/include/sip.h: Extend extension state callbacks to + have more information. Quote from review board: This patch + extends the extension state callbacks so that monitoring channels + (as chan_sip) get more information of the devices which are + responsible for an extension state change. The additional + information is needed by chan_sip to present names/numbers of the + caller and callee in an early-state SIP notification. Users of + extenstion state callback not interested in the additional + information are not affected by the changes. Motivation: to + present the involved party's name/number in an early-state + nofification (used by the notified device as a pickup offer) one + after another so that a user can see which call he will pick up + in an undirected pickup. Such a pickup offer to a user shall + indicate the same call (number/name-A calls number/name-B) as the + call which would be picked up when an undirected pickup is + executed. Users interested in additional state info must use the + new functions ast_extension_state_add_extended() resp. + ast_extension_state_add_destroy_extended() to register an + extended state callback. When the callback is registered this + way, an extra member device_state_info of struct + ast_state_cb_info is passed to the callback in addition to the + aggregated extension state. This container holds an object for + every device of the monitored extension hint consisting of the + device name, the device state and a channel reference to the + channel which (presumably) caused the device state. The + information is used by chan_sip for early-state notifications. + When the state of a device changes and the new state contains + AST_EVENT_RINGING, an early-state notification is sent to the + subscribed devices with the caller/callee names/numbers of the + oldest ringing channel of the monitored extension. The notified + user may then invoke a direct pickup, which will pickup exactly + this channel. Users of the old non-extended callbacks will only + be called when the aggregated state did change (same behavior as + before). Users of the extended callback will also be called when + the state is unchanged but does contain AST_EVENT_RINGING. That + could be the case if two channels are ringing at one device and + one of them hangs up, so the aggregated state does not change. + This way the monitoring channel can create a new early-state + notification with the now ringing party-ids. Review: + https://reviewboard.asterisk.org/r/2048 This contribution comes + from Guenther Kelleter + +2012-08-09 14:36 +0000 [r370978] Jonathan Rose + + * pbx/pbx_dundi.c, CHANGES: DUNDi: Add CLI commands DUNDi show + cache and DUNDi show hints (closes issue ASTERISK-18390) Reported + by: Peter Racz Patches: dundi_cli_cache.patch.v2 uploaded by + Peter Racz (license #6290) + ASTERISK-18390_dundi_cli_cache_jrose_mods_v2.diff uploaded by + Jonathan Rose (license #6182) + +2012-08-08 22:45 +0000 [r370955] Michael L. Young + + * /, apps/app_chanspy.c: Fix Not Unreferencing A Spied Channel When + a channel hangs up while being spied upon and the option to exit + the ChanSpy application when the spied on channel hangs up is + set, ast_autochan_destroy is not being called and therefore a + reference to the spied upon channel is not removed. The symptom + being reported was that when using func_group in the dialplan and + calling "group show channels" at the cli, the spied upon channel + was still being shown while "core show channels" showed that the + channel was not up. This patch calls ast_autochan_destroy when a + spied upon channel hangs up and the option to exit the ChanSpy + application is set, removing the reference to the channel + allowing the count for the group that the spied channel was part + of to be decremented. (closes issue ASTERISK-17515) Reported by: + Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young + Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael + L. Young (license 5026) ........ Merged revisions 370952 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370954 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-08 22:41 +0000 [r370951-370953] Mark Michelson + + * CHANGES: Move a SIP change up to the other SIP changes in the + CHANGES file. + + * main/channel.c, main/pbx.c, main/manager.c, pbx/pbx_spool.c, + apps/app_originate.c, include/asterisk/channel.h, + include/asterisk/pbx.h, CHANGES, res/res_clioriginate.c: Allow + support for early media on AMI originates and call files. This is + based on the work done by Olle Johansson on review board. The + idea is that the channel specified in an AMI originate or call + file is typically not connected to the outgoing extension until + the channel has been answered. With this change, an EarlyMedia + header can be specified for AMI originates and an early_media + option can be specified in call files. With this option set, once + early media is received on a channel, it will be connected with + the outgoing extension. (closes issue ASTERISK-18644) Reported by + Olle Johansson Review: https://reviewboard.asterisk.org/r/1472 + +2012-08-08 21:22 +0000 [r370943] Terry Wilson + + * main/manager.c, CHANGES: Add AMI_CLIENT dialplan function + Implementation of a dialplan function for checking manager + accounts. Right now it only returns the number of logged in + sessions for a manager account, but other attributes can be added + later. Patch by: Olle Johansson Review: + https://reviewboard.asterisk.org/r/421/ + +2012-08-08 20:47 +0000 [r370927] Joshua Colp + + * main/rtp_engine.c: Create the payload type if it does not exist + when setting information based on the 'm' line. An rtpmap + attribute is not required for defined payload numbers. + +2012-08-08 20:32 +0000 [r370926] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_analog.h: Convert sig_analog to use a global + callback table. + +2012-08-08 20:30 +0000 [r370925] Kinsey Moore + + * main/channel.c, /: Do not define a cause that doesn't actually + exist AST_CAUSE_NOTDEFINED is a placeholder for usage when there + is no cause information. As such, it should not be defined and + translatable as a cause. ........ Merged revisions 370923 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370924 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-08 20:17 +0000 [r370887-370902] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Fix the analog dial *0 flash-hook of + bridged peer feature. The flash-hook the bridged peer feature now + correctly determines if the bridged peer is another chan_dahdi + channel, that it is an analog channel, and that it has the + correct signaling for an FXO port. It now also flash-hooks the + correct channel. ........ Merged revisions 370900 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370901 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Convert sig_pri to use a global callback table. + + * channels/chan_dahdi.c, channels/sig_ss7.h, channels/sig_ss7.c: + Convert sig_ss7 to use a global callback table. + +2012-08-07 21:58 +0000 [r370881] Damien Wedhorn + + * build_tools/cflags-devmode.xml, channels/chan_skinny.c: Rewrite + of skinny debugging. Debugging messages and associated controls + only compiled in if configured with --enable-dev-mode. Debug + messages provide more detail (including thread id) and are + grouped so the user/dev can limit the type of messages displayed. + Functionally no real change to chan_skinny. Review: + https://reviewboard.asterisk.org/r/2040/ + +2012-08-07 19:59 +0000 [r370860] Joshua Colp + + * main/rtp_engine.c, include/asterisk/rtp_engine.h: Payload and RTP + code are must remain separate since in non-Asterisk format cases + they differ. + +2012-08-07 19:26 +0000 [r370851-370859] Kinsey Moore + + * /: Recorded merge of revisions 370858 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Add + missing AST_CAUSE_* -> text translations ........ Merged + revisions 370856 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * main/channel.c: Add missing AST_CAUSE_* -> text translations A + few of these were missing from the list and are necessary for the + Who Hung Up? functionality. + +2012-08-07 17:47 +0000 [r370832-370845] Joshua Colp + + * main/rtp_engine.c: Fix a bug uncovered by the test suite where + the RTP payload number was not getting set. + + * res/res_rtp_asterisk.c, main/rtp_engine.c, channels/chan_sip.c, + channels/chan_motif.c, include/asterisk/rtp_engine.h: Reduce + memory consumption significantly for users of the RTP engine API + by storing only the payloads present and in use instead of every + possible one. Review: https://reviewboard.asterisk.org/r/2052/ + +2012-08-07 12:46 +0000 [r370820-370831] Matthew Jordan + + * main/channel.c, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, channels/chan_misdn.c, + channels/chan_sip.c, main/channel_internal_api.c, + channels/misdn/chan_misdn_config.h, main/features.c, + configs/misdn.conf.sample, include/asterisk/channel.h, + configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h, + channels/misdn_config.c: Add named callgroups/pickupgroups This + patch adds named calledgroups/pickupgroups to Asterisk. Named + groups are implemented in parallel to the existing numbered + callgroup/pickupgroup implementation. However, unlike the + existing implementation, which is limited to a maximum of 64 + defined groups, the number of defined groups allowed for named + callgroups/pickupgroups is effectively unlimited. Named groups + are configured with the keywords "namedcallgroup" and + "namedpickupgroup". This corresponds to the numbered group + definitions of "callgroup" and "pickupgroup". Note that as the + implementation of named groups coexists with the existing + numbered implementation, a defined named group of "4" does not + equate to numbered group 4. Support for the named groups has been + added to the SIP, DAHDI, and mISDN channel drivers. Review: + https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther + Kelleter(license #6372) + + * contrib/realtime/mysql/voicemail_data.sql: Revert r370820 That + change is wrong, wrong, wrong. + + * contrib/realtime/mysql/voicemail_data.sql: Update the MySQL + voicemail_data contrib script to reflect Asterisk 11 changes All + voicemails now have a 'msg_id' included in their metadata. The + ODBC message storage backend now requires this column; as such, + the MySQL contrib script that creates the voicemail_data table + has been updated with the appropriate column information. + +2012-08-06 15:18 +0000 [r370801] Mark Michelson + + * /, channels/chan_sip.c: Improve debug message for temporary + outbound proxies. Thanks to Paul Belanger for pointing this out. + ........ Merged revisions 370797 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370798 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-03 21:52 +0000 [r370773] Mark Michelson + + * /, channels/chan_sip.c, channels/sip/config_parser.c, + channels/sip/include/sip.h: Multiple revisions 370769-370771 + ........ r370769 | mmichelson | 2012-08-03 16:35:00 -0500 (Fri, + 03 Aug 2012) | 24 lines Fix error in the "IPorHost" section of a + SIP dialstring. This is based on the review request posted by + Walter Doekes (referenced lower in the commit message) The main + fix here is to treat the IPorHost portion of the dial string as a + temporary outbound proxy. This ensures requests get sent to the + proper location. Due to the age of the request, some parts were + no longer relevant. For instance, the request moved outbound + proxy parsing code into a single method. This is done in a + previous commit, so it was not necessary to do again. Also, the + review request fixed some errors with regards to request routing + for CANCEL and ACK requests. This has also been fixed in more + recent commits. (closes issue ASTERISK-19677) reported by Walter + Doekes Review https://reviewboard.asterisk.org/r/1859 ........ + r370770 | mmichelson | 2012-08-03 16:39:35 -0500 (Fri, 03 Aug + 2012) | 3 lines Remove unused variable. ........ r370771 | + mmichelson | 2012-08-03 16:43:52 -0500 (Fri, 03 Aug 2012) | 5 + lines Seriously? Another compilation error fixed. Somebody beat + me. ........ Merged revisions 370769-370771 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370772 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-08-02 15:51 +0000 [r370740] Kinsey Moore + + * channels/chan_sip.c: Fix regression from r370636 When the + chan_sip cleanup went in, a typo was included that caused some + subscriptions of non-Polycom phones to be limited to the same + capabilities as Polycom phones. This resolves the failures in the + test suite resulting from this regression. + +2012-08-01 19:37 +0000 [r370726] Mark Michelson + + * main/manager.c: Fix a possible crash due to passing NULL to + ast_variables_dup() + +2012-08-01 18:52 +0000 [r370720] Richard Mudgett + + * include/asterisk/astobj2.h, main/astobj2.c: Make astobj2.h not + include linkedlists.h. Using astobj2 does not require + linkedlists.h be included even though astob2 uses linked lists + internally. + +2012-08-01 02:26 +0000 [r370699] Kinsey Moore + + * /, utils/extconf.c: Revert alloca changes for utils These changes + were a tad overzealous in the utils directory. Unfortunately, + these don't compile with a "make". ........ Merged revisions + 370697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 370698 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-31 22:28 +0000 [r370681-370691] Mark Michelson + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/sip/include/sip.h: Add headers from SIPAddHeader to + outbound REFER requests. This is a patch from kkm from review + board. This is useful for adding headers to REFER requests that + emanate from a Transfer() dialplan application call. This also + fixes some uses of the Referred-by header, removing an extra set + of angle brackets. I've modified the reporter's original patch to + not require any additions to the sip_refer header and to just + remove the referred_by_name from sip_refer since it is no longer + needed or used. (closes Issue ASTERISK-17639) reported by Kirill + Katsnelson Patches: 019059-sip-refer-addheaders-trunk-353549.diff + uploaded by Kirill Katsnelson (license #5845) Review: + https://reviewboard.asterisk.org/r/1159 + + * main/manager.c, configs/manager.conf.sample, CHANGES: Add + "setvar" option to manager.conf. With this option set, channel + variables can be set on every manager originate. The Variable + header can still be used to set additional channel variables for + individual calls if desired. This work was completed by Olle + Johansson on review board. I have applied the review feedback and + am committing it in order to get this into trunk before Asterisk + 11 is branched. Review: https://reviewboard.asterisk.org/r/1412 + +2012-07-31 21:20 +0000 [r370677] Matthew Jordan + + * /, channels/chan_sip.c: Schedule pokes of registered SIP peers + within a given timespan after SIP reload With a large number of + SIP peers registered, performing a SIP reload causes a flood of + SIP OPTIONS request packets. These are immediately sent out, and, + as responses come back, can cause peers to be flagged as 'lagged' + due to handling of the many response messages. This fix prevents + this "packet storm" and schedules the pokes for a random time. + That time varies between 1 ms and the peer's qualify time, or, if + the qualify time is unknown, the global qualifyfreq setting. The + committed patch has some very small modifications to the patch + schmidts wrote for the review. (closes issue ASTERISK-19154) + Reported by: Nicolo Mazzon patches: issue19154.patch license + #6034 uploaded by schmidts Review: + https://reviewboard.asterisk.org/r/1652 ........ Merged revisions + 370666 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 370672 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-31 20:33 +0000 [r370664] Russell Bryant + + * main/event.c: Move event cache updates into event processing + thread. Prior to this patch, updating the device state cache was + done by the thread that originated the event. It would update the + cache and then queue the event up for another thread to dispatch. + This thread moves the cache updating part to be in the same + thread as event dispatching. I was working with someone on a + heavily loaded Asterisk system and while reviewing backtraces of + the system while it was having problems, I noticed that there + were a lot of threads contending for the lock on the event cache. + By simply moving this into a single thread, this helped + performance *a lot* and alleviated some deadlock-like symptoms. + Review: https://reviewboard.asterisk.org/r/2066/ + +2012-07-31 20:21 +0000 [r370655] Kinsey Moore + + * /, main/say.c, main/threadstorage.c, funcs/func_strings.c, + channels/chan_iax2.c, main/config.c, channels/chan_dahdi.c, + pbx/pbx_spool.c, channels/sig_analog.c, main/strcompat.c, + main/features.c, pbx/pbx_ael.c, main/http.c, pbx/pbx_realtime.c, + channels/chan_alsa.c, channels/sig_ss7.c, main/db.c, + include/asterisk/utils.h, main/pbx.c, funcs/func_cut.c, + tests/test_linkedlists.c, funcs/func_channel.c, apps/app_macro.c, + apps/app_mixmonitor.c, main/asterisk.c, apps/app_voicemail.c, + addons/app_mysql.c, apps/app_meetme.c, apps/app_dictate.c, + main/utils.c, funcs/func_logic.c, cdr/cdr_pgsql.c, + channels/chan_gtalk.c, res/res_jabber.c, + res/res_http_websocket.c, res/ael/pval.c, main/channel.c, + main/manager.c, apps/app_osplookup.c, res/res_agi.c, + apps/app_minivm.c, main/logger.c, main/app.c, + addons/chan_mobile.c, apps/app_while.c, res/res_config_pgsql.c, + channels/chan_sip.c, apps/app_festival.c, pbx/pbx_lua.c, + channels/sig_pri.c, apps/app_getcpeid.c, funcs/func_global.c, + channels/chan_jingle.c, main/tcptls.c, + apps/app_directed_pickup.c, main/file.c, main/callerid.c, + apps/app_sms.c, main/astmm.c, main/event.c, pbx/pbx_dundi.c, + include/asterisk/strings.h, utils/extconf.c, main/dsp.c, + addons/res_config_mysql.c: Clean up and ensure proper usage of + alloca() This replaces all calls to alloca() with ast_alloca() + which calls gcc's __builtin_alloca() to avoid BSD semantics and + removes all NULL checks on memory allocated via ast_alloca() and + ast_strdupa(). (closes issue ASTERISK-20125) Review: + https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes + (wdoekes) ........ Merged revisions 370642 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370643 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-31 19:57 +0000 [r370644] Mark Michelson + + * CHANGES, pbx/pbx_config.c: Add "dialplan remove context" and + modify "dialplan add include" From corruptor's review board + posting: "I've noticed that we can remove particular extension + from context with dialplan remove extension command but in order + to remove all extensions in the context we should delete them on + by one. I've created dialplan remove context command which uses + ast_context_destroy to destroy the whole context with all + extensions. I've created to functions for in pbx_config.c: + handle_cli_dialplan_remove_context which actually removes context + and complete_dialplan_remove_context which completes input. They + are based on other similar functions and pretty trivial but I can + be mistaken somewhere. "I've also modified dialplan add include + into . I've made it similar dialplan add + extension ... command. It creates if it doesn't exist + and I've also modified complete_dialplan_add_include and removed + check for existance of because we can include + non-existent context into another one. (I usually include empty + (non-existent) contexts in advance). Should we raise warning in + this case as it's raised while reading extensions.conf? "I use + those functions with AMI. I think manager commands should be + created in addition to those CLI commands." I've addressed the + latest comments on review board and have made some other coding + guidelines-related cleanup. I also have modified the CHANGES file + to mention these new commands. (closes issue ASTERISK-19292) + reported by Andrey Solovyev Patches: dialplan_add_include.patch + uploaded by Andrey Solovyev (license #5214) + dialplan_remove_context.patch uploaded by Andrey Solovyev + (license #5214) Review: https://reviewboard.asterisk.org/r/2042 + +2012-07-31 19:10 +0000 [r370636] Kinsey Moore + + * channels/chan_sip.c, channels/sip/security_events.c, + channels/sip/include/sip.h: Clean up chan_sip This clean up was + broken out from https://reviewboard.asterisk.org/r/1976/ and + addresses the following: - struct sip_refer converted to use the + stringfields API. - sip_{refer|notify}_allocate -> + sip_{notify|refer}_alloc to match other *alloc functions. - + Replace get_msg_text, get_msg_text2 and get_pidf_body -> No, not + get_pidf_msg_text_body3 but get_content, to match add_content. - + get_body doesn't get the request body, renamed to + get_content_line. - get_body_by_line doesn't get the body line, + and is just a simple if test. Moved code inline and removed + function. - Remove camelCase in struct sip_peer peer state + variables, onHold -> onhold, inUse -> inuse, inRinging -> + ringing. - Remove camelCase in struct sip_request rlPart1 -> + rlpart1, rlPart2 -> rlpart2. - Rename instances of pvt->randdata + to pvt->nonce because that is what it is, no need to update + struct sip_pvt because _it already has a nonce field_. - Removed + struct sip_pvt randdata stringfield. - Remove useless (and + inconsistent) 'header' suffix on variables in + handle_request_subscribe. - Use ast_strdupa on Event header in + handle_request_subscribe to avoid overly complicated strncmp + calls to find the event package. - Move get_destination check in + handle_request_subscribe to avoid duplicate checking for packages + that don't need it. - Move extension state callback management in + handle_request_subscribe to avoid duplicate checking for packages + that don't need it. - Remove duplicate append_date prototype. - + Rename append_date -> add_date to match other add_xxx functions. + - Added add_expires helper function, removed code that manually + added expires header. - Remove _header suffix on + add_diversion_header (no other header adding functions have + this). - Don't pass req->debug to request handle_request_XXXXX + handlers if req is also being passed. - Don't pass req->ignore to + check_auth as req is already being passed. - Don't create a + subscription in handle_request_subscribe if p->expiry == 0. - + Don't walk of the back of referred_by_name when splitting string + in get_refer_info - Remove duplicate check for no dialog in + handle_incoming when sipmethod == SIP_REFER, handle_request_refer + checks for that. Review: https://reviewboard.asterisk.org/r/1993/ + Patch-by: gareth + +2012-07-30 23:26 +0000 [r370565-370598] Richard Mudgett + + * main/test.c: Tweak unit test warning message. + + * funcs/func_presencestate.c, main/test.c: Fix some presence-state + unit test typos. + + * apps/app_confbridge.c: DECLINE to load confbridge if the config + fails to load. + + * channels/chan_misdn.c, /: Release B channel allocation on error + path in chan_misdn. ........ Merged revisions 370563 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370564 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-30 14:52 +0000 [r370548] Jonathan Rose + + * /, apps/app_meetme.c: app_meetme: Change app_meetme support level + to extended from deprecated (closes issue ASTERISK-20134) + Reported by: Leif Madsen ........ Merged revisions 370547 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-30 13:45 +0000 [r370534-370541] Russell Bryant + + * tests/test_event.c: Fix ast_event_new unit test. One of my recent + commits broke this test. The error was: + [test_event.c:event_new_test:214]: Events expected to be + identical have different size: 69 != 59 The difference in size + occurred because the first event had the EID IE added to the + event twice. ast_event_new() now always adds it automatically. + Previously it only added it if there were no IEs specified, which + was kind of weird. + + * include/asterisk/event_defs.h, res/res_corosync.c, main/event.c: + Add a "corosync ping" CLI command. This patch adds a new CLI + command to the res_corosync module. It is primarily used as a + debugging tool. It lets you fire off an event which will cause + res_corosync on other nodes in the cluster to place messages into + the logger if everything is working ok. It verifies that the + corosync communication is working as expected. I didn't put + anything in the CHANGES file for this, because this module is new + in Asterisk 11. There is already a generic "res_corosync new + module" entry in there so I figure that covers it just fine. + + * addons/app_mysql.c, CHANGES: Allow specifying a port number for + the MySQL server. This patch allows you to specify a port number + for the MySQL server. It's useful if a MySQL server is running on + a non-standard port. Even though this module is deprecated in + favor of func_odbc, someone asked for this feature and it seems + pretty harmless to add. It has been tested using a number of + combinations of with/without a port number specified in the + dialplan and changing the port number for mysqld. + +2012-07-26 15:31 +0000 [r370510-370518] Jonathan Rose + + * channels/chan_sip.c, CHANGES: chan_sip: Add SIPpeerstatus command + to AMI This patch was submitted by mnicholson a while back. It + adds a new AMI action which allows users to request SIP peer + status on demand similar to existing PeerStatus events and to the + output you would see from CLI with sip show peer Review: + https://reviewboard.asterisk.org/r/1098/ + + * /, res/res_agi.c: res_agi: Add message indicating need for \n + character in verbose message The while loop responsible for + reading AGI messages from a fastAGI service can end up looping + indefinitely when an AGI script fails to indicate the end of a + message with a \n character. This patch adds an indication that + we are expecting a \n character to end the message to make it + more clear to users that this is necessary if they are receiving + this warning over and over. (issue ASTERISK-20061) Reported by: + Eike Kuiper ........ Merged revisions 370494 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370495 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-25 14:27 +0000 [r370481-370488] Kevin P. Fleming + + * main/Makefile: Repair editline builds using in-tree editline + sources. The previous change to the build system for using a + system-provided editline library was missing a crucial include + directory for building against the copy of the library in the + Asterisk source tree. + + * main/Makefile: Use an absolute path when referring to the + embedded editline directory. This patch changes the build system + to refer to the embedded editline directory using an absolute + path, which will resolve a problem seen on the CentOS automated + build agents. + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, main/Makefile, + main/editline/configure, configure.ac, main/editline/readline + (removed), main/editline/readline.c, main/editline/configure.in, + CHANGES, makeopts.in, main/editline/readline.h (added), + main/asterisk.c, contrib/scripts/install_prereq, main/cli.c: + Enable usage of system-provided NetBSD editline library if + available. This patch changes the Asterisk configure script and + build system to detect the presence of the NetBSD editline + library (libedit) on the system. If it is found, it will be used + in preference to the version included in the Asterisk source + tree. (closes issue ASTERISK-18725) Reported by: Jeffrey C. Ollie + Review: https://reviewboard.asterisk.org/r/1528/ Patches: + 0001-Allow-linking-building-against-an-external-editline.patch + uploaded by jcollie (license #5373) (heavily modified by + kpfleming) + +2012-07-25 03:51 +0000 [r370474] Terry Wilson + + * main/pbx.c, /: Revert a change that broke compilation 1) There is + no such function as ast_ref() 2) The patch was originally + credited as the one uploaded by Guenther Kelleter (license 6372) + via issue AST-921, but the patch committed was not the patch + referenced on the issue. 3) Guenther Kelleter's patch was + actually correct. It moved the ast_free above the + presencechange_cleanup label. I am not committing his change as + it is not technically necesary--calling ast_free(NULL) is + perfectly safe and I worry that moving the ast_free outside of + the label could lead to future bugs if someone ever adds another + failure conditional and expects 'goto presencechange_cleanup;' to + clean up after everything. + +2012-07-24 21:30 +0000 [r370466] Jonathan Rose + + * main/pbx.c, /: Don't attempt free of NULL ptr in pbx.c + handle_presencechange (closes issue AST-921) Reported by: + Guenther Kelleter Patches: nullptr.patch uploaded by Guenther + Kelleter (license 6372) + +2012-07-24 19:12 +0000 [r370453] Kevin P. Fleming + + * tests/test_acl.c: Silence a warning message from older versions + of GCC. Revision 370426 introduced the use of a nested function + in tests/test_acl.c, but the lack of the 'auto' scope specifier + on the function and a forward declaration resulted in compilation + errors on the automated test systems. + +2012-07-24 17:16 +0000 [r370433] Tzafrir Cohen + + * /, channels/chan_oss.c: chan_oss: fix "sample rate" error message + Merged revisions 370428 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 Merged + revisions 370432 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-24 16:54 +0000 [r370426-370431] Kevin P. Fleming + + * main/frame.c, /: Rewrite a comment that didn't adequately explain + the code it was documenting. ........ Merged revisions 370429 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 370430 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * CHANGES: Update CHANGES for list/negation ACL feature. + + * tests/test_acl.c, main/acl.c: Allow permit/deny ACL lines to + contain multiple items and negated entries. Rules in ACLs + (specified using 'permit' and 'deny') can now contain multiple + items (separated by commas), and items in the rule can be negated + by prefixing them with '!'. This simplifies Asterisk Realtime + configurations, since it is no longer necessray to control the + order that the 'permit' and 'deny' columns are returned from + queries. Review: https://reviewboard.asterisk.org/r/1592/ Initial + patch contributed by Tilghman Lesher Unit tests written by Kevin + P. Fleming + +2012-07-24 16:15 +0000 [r370419-370420] Joshua Colp + + * res/res_rtp_asterisk.c: Build is underway so logging can go away. + + * res/res_rtp_asterisk.c: Temporarily enable pj logging to console + for debugging pjnath issue exposed by build slave. + +2012-07-24 08:53 +0000 [r370413] Igor Goncharovskiy + + * channels/chan_unistim.c: Remove code, that operate with cdr in + attempt_transfer(). That was removed somewhere between 1.2 and + 1.4 and acidentaly put back in chan_unistim. (closes issue + ASTERISK-19628) Reported by: Igor Olhovskiy + +2012-07-23 21:27 +0000 [r370407] Kevin P. Fleming + + * codecs/Makefile, build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, + codecs/codec_ilbc.c, CHANGES, makeopts.in: Enable usage of + system-provided iLBC library. The WebRTC version of the iLBC + codec is now package as a library and is available on some + platforms. This patch allows codec_ilbc to be built against that + library if it is present. Review: + https://reviewboard.asterisk.org/r/1964/ + +2012-07-23 21:15 +0000 [r370387] Matthew Jordan + + * tests/test_abstract_jb.c (added), main/abstract_jb.c, + funcs/func_jitterbuffer.c, include/asterisk/abstract_jb.h: Unit + tests for the Jitter Buffer API; remove unnecessary resync This + patch includes the following: * Unit tests for the abstract + Jitter Buffer API. This includes both fixed and adaptive flavors, + testing nominal creation, frame input, frame retrieval, + resyncing; off nominal frame input overflow, out of order, and + others. * Tweaks to the abstract_jb API to remove the unnecessary + resync_threshold parameter from the create function + (resync_threshold is already in the struct passed into the create + function) * Ensure the fixed jitter buffer is empty before + destroying it, to avoid an ASSERT * Don't "resync" the adaptive + jitter buffer. The mechanism that was being used actually causes + the jitter buffer to think its being overflowed by going around + the jitterbuf API and attempting to 'resynch' it improperly. If a + resync is needed, the jitter buffer will do it properly by + itself. Note that this is only an optimization needed for trunk, + as the worst that happens is the loss of three voice packets + before the adaptive jitter buffer will resync anyway. Review: + https://reviewboard.asterisk.org/r/2035 + +2012-07-23 21:10 +0000 [r370386] Mark Michelson + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add + separate configuration options for subscription and registration + minexpiry and maxexpiry. This offers more fine-grained control + over how long subscriptions last without negatively affecting the + expiration range for registrations. Uploaded by: Guenther + Kelleter(license #6372) Review: + https://reviewboard.asterisk.org/r/2051 + +2012-07-23 21:10 +0000 [r370385] Kevin P. Fleming + + * /, funcs/func_shell.c: Improve documentation for the SHELL() + dialplan function. ........ Merged revisions 370383 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370384 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-23 21:02 +0000 [r370382] Mark Michelson + + * UPGRADE.txt: Add notes to UPGRADE.txt about addition of msg_id to + VoiceMails. + +2012-07-23 00:15 +0000 [r370354] Joshua Colp + + * UPGRADE.txt: Update UPGRADE.txt with notes about ICE support and + res_xmpp. + +2012-07-22 23:37 +0000 [r370353] Matthew Jordan + + * CHANGES: Update CHANGES for Asterisk 11 This updates the CHANGES + file with things that were committed for Asterisk 11, but were + not noted in that file. + +2012-07-22 17:03 +0000 [r370347] Joshua Colp + + * res/res_rtp_asterisk.c, channels/chan_sip.c, + configs/sip.conf.sample, channels/sip/include/sip.h: Prevent + multiple local candidates from being added with the same + information and add support for disabling ICE on a per-peer + basis. (closes issue ASTERISK-20088) Reported by: wimpy Review: + https://reviewboard.asterisk.org/r/2044/ + +2012-07-21 13:25 +0000 [r370341] Terry Wilson + + * main/config_options.c, apps/app_confbridge.c, + apps/confbridge/conf_config_parser.c: Fix segfault introduced by + conversion to ACO API The value "none" is specified in the config + file as a valid value for the "video_mode" option. The code prior + to the ACO conversion did not check for "none", but just ignored + it and relied on the default zero value. The parsing with ACO is + more strict, so without handling "none" specifically, parsing + would fail. When parsing failed, but the module loaded anyway, + the config info would never be stored, and one place in the code + did not check for this case and would segfault. It was also + possible that the aco_info struct's internals would be destroyed + and used as well. This patch keeps the module from loading after + parse failures, adds the "none" option to "video_mode", registers + CLI functions only after parsing has completed, checks the config + data for NULL before accessing it, and returns -1 on some + allocation failures when initializing. (closes issue + ASTERISK-20159) Reported by: Birger "WIMPy" Harzenetter Tested + by: Birger "WIMPy" Harzenetter Patches: confbridge_fix3.txt + uploaded by Terry Wilson + +2012-07-20 19:36 +0000 [r370335] Jonathan Rose + + * channels/chan_iax2.c: chan_iax2: Fix a segfault introduced by + call ID logging Didn't previously check that a non NULL IAX + channel was stored in the array at the requested position before + attempting iax_pvt_callid_get (closes issue ASTERISK-20145) + Reported by: Birger "WIMPy" Harzenetter + +2012-07-20 19:08 +0000 [r370329] Matthew Jordan + + * apps/app_dial.c: Clean up ManagerEvent Dial documentation The + paragraph describing the SubEvent belongs with the SubEvent + parameter itself, and not with its enum values. The order of + parsing was placing the description after the last enum, which + isn't correct. + +2012-07-20 18:37 +0000 [r370328] Kinsey Moore + + * channels/chan_misdn.c: Fix build error in chan_misdn from commit + 370316 chan_misdn was not updated properly to account for a + change in parameters for HANGUPCAUSE functionality. It now builds + properly. + +2012-07-20 16:25 +0000 [r370322] Joshua Colp + + * res/res_http_websocket.exports.in: Export the + ast_websocket_set_nonblock function for use by other modules. + +2012-07-20 15:48 +0000 [r370316] Kinsey Moore + + * funcs/func_hangupcause.c (added), main/channel.c, + channels/chan_dahdi.c, channels/sig_analog.c, main/rtp_engine.c, + channels/chan_sip.c, main/channel_internal_api.c, UPGRADE.txt, + include/asterisk/channel.h, channels/chan_iax2.c, + channels/sig_pri.c, include/asterisk/frame.h, channels/sig_ss7.c: + Add hangupcause translation support The HANGUPCAUSE hash (trunk + only) meant to replace SIP_CAUSE has now been replaced with the + HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan functions to better + facilitate access to the AST_CAUSE translations for + technology-specific cause codes. The HangupCauseClear application + has also been added to remove this data from the channel. (closes + issue SWP-4738) Review: https://reviewboard.asterisk.org/r/2025/ + +2012-07-20 15:40 +0000 [r370309-370315] Richard Mudgett + + * CHANGES: Update CHANGES about adding the AccountCode header to + the AMI Hangup event. (issue ASTERISK-19963) + + * main/channel.c: Add the AccountCode header to the AMI Hangup + event. It's harder to correlate the Newchannel and Hangup AMI + events without specifying "AccountCode" in both. (closes issue + ASTERISK-19963) Reported by: Oleg A. Arkhangelsky Patches: + hangup_acctcode.diff (license #6397) patch uploaded by Oleg A. + Arkhangelsky + +2012-07-19 23:21 +0000 [r370303] Terry Wilson + + * include/asterisk/config_options.h, + apps/confbridge/include/confbridge.h, main/config_options.c, + apps/confbridge/conf_config_parser.c: Convert app_confbridge to + use the config options framework Review: + https://reviewboard.asterisk.org/r/2024/ + +2012-07-19 22:25 +0000 [r370298] Richard Mudgett + + * /, main/cel.c: Fix compiler warnings. gcc (GCC) 4.2.4 has + problems casting away constness. ........ Merged revisions 370275 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 370277 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-19 22:17 +0000 [r370272-370278] Matthew Jordan + + * channels/chan_sip.c, res/res_xmpp.c, doc/appdocsxml.dtd, + main/message.c, main/xmldoc.c: Add the ability to specify + technology specific documentation A number of applications/AMI + commands in Asterisk have specific behavioral differences + depending on the resource or channel technology those + applications are executed on. For example, the MessageSend + application/ command is technology agnostic, but how the channel + drivers that support that functionality behave is dependant on + the protocols and channel driver implementation. Prior to this + patch, those details were either documented in the + application/command documentation itself, or were left + undocumented. This patch adds a new element to the documentation + schema, . An info node is essentially a piece of + technology specific reference information that can be included by + any top level XML documentation node. For example, the + MessageSend application can now include XMPP/SIP specific + information, where that technology specific information can be + defined in chan_motif/res_xmpp/ chan_sip. Likewise, that + information can also be included in the MessageSend AMI command. + Review: https://reviewboard.asterisk.org/r/2049 + + * /, main/cel.c: Fix compilation error when MALLOC_DEBUG is enabled + To fix a memory leak in CEL, a channel datastore was introduced + whose destruction function pointer was pointed to the ast_free + macro. Without MALLOC_DEBUG enabled this compiles as fine, as + ast_free is defined as free. With MALLOC_DEBUG enabled, however, + ast_free takes on a definition from a different place then + utils.h, and became undefined. This patch resolves this by using + a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this + calls ast_free; when MALLOC_DEBUG is not enabled, this is defined + to be ast_free, which is defined to be free. (issue AST-916) + Reported by: Thomas Arimont ........ Merged revisions 370273 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370274 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * res/res_rtp_asterisk.c, /: Handle extremely out of order RFC 2833 + DTMF The current implementation of RFC 2833 DTMF handling in + res_rtp_asterisk will, if a packet arrives out of order, drop the + packet. This is to prevent duplicate ton generation in the + Asterisk core. Since the RTP layer does not buffer data itself, + this is the only option the RTP layer currently has for handling + packets that arrive out of order. For the most part, this doesn't + matter. For a particular digit, so long as a BEGIN packet arrives + before the first END packet, the digit will be produced. If + subsequent BEGIN packets arrive interleaved with the ENDs, they + will be dropped; likewise, if the BEGIN or END packets themselves + are out of order, those packets are dropped but sufficient + information is conveyed to the Asterisk core to produce the + appropriate digit. For certain sequences of DTMF packets - most + notably when, for a particular digit, an END packet arrives + before any BEGIN packet for that digit - this is a real problem. + When an END arrives before any BEGINs, the END packet is dropped + - but at the same time, it causes subsequent BEGIN packets for + that digit to be ignored. When the next in order END packet + arrives, it too is dropped - Asterisk believes that there was no + initial BEGIN. The solution this patch provides is to trust the + END packet to convey the information needed for the Asterisk core + to produce the DTMF digit. If we receive an END packet, and it: * + Has a timestamp greater then the last timestamp received from an + END packet * Does not have the same sequence number as the last + received sequence number (and is thus not an END packet + retransmission) Then we send the END frame up to the Asterisk + core. It contains enough DTMF information for Asterisk to produce + the digit. On the other hand, if we receive a BEGIN or + continuation packet that occurs with a timestamp equal to or less + then the last END timestamp, then we've received something out of + order - but we already have received enough information to + produce the digit. These packets are dropped. Much thanks goes to + Olle Johansson (oej) for providing the idea for this solution. + Review: https://reviewboard.asterisk.org/r/2033/ (closes issue + ASTERISK-18404) Reported by: Stephane Chazelas Tested by: Matt + Jordan ........ Merged revisions 370252 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370271 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-19 20:37 +0000 [r370246-370265] Jonathan Rose + + * main/named_acl.c, configs/acl.conf.sample: named_acl: Remove + systemname option from acl.conf, use asterisk.conf value Review: + https://reviewboard.asterisk.org/r/2057/ + + * main/channel_internal_api.c: CallID Logging: Remove new + line/carriage return from callID change test event + +2012-07-19 12:14 +0000 [r370234-370240] Joshua Colp + + * res/Makefile, res/pjproject/build/os-auto.mak.in: Use the + bruteforce method to get debugging enabled for pjproject. + + * res/Makefile: Turn on debugging for pjproject so we can get a + better idea of what is causing the generic CCSS test crash. + +2012-07-18 19:48 +0000 [r370225] Jonathan Rose + + * main/channel_internal_api.c: callid logging: Issue test events + when the callid is changed for a channel Review: + https://reviewboard.asterisk.org/r/2054/ + +2012-07-18 19:18 +0000 [r370187-370211] Kevin P. Fleming + + * /, main/cel.c: Resolve severe memory leak in CEL logging modules. + A customer reported a significant memory leak using Asterisk 1.8. + They have tracked it down to + ast_cel_fabricate_channel_from_event() in main/cel.c, which is + called by both in-tree CEL logging modules (cel_custom.c and + cel_sqlite3_custom.c) for each and every CEL event that they log. + The cause was an incorrect assumption about how data attached to + an ast_channel would be handled when the channel is destroyed; + the data is now stored in a datastore attached to the channel, + which is destroyed along with the channel at the proper time. + (closes issue AST-916) Reported by: Thomas Arimont Review: + https://reviewboard.asterisk.org/r/2053/ ........ Merged + revisions 370205 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370206 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/channel.c, addons/app_mysql.c, main/pbx.c, + funcs/func_curl.c, /, main/ccss.c, funcs/func_odbc.c, + funcs/func_lock.c, apps/app_macro.c, channels/chan_iax2.c, + apps/app_mixmonitor.c, apps/app_stack.c, funcs/func_global.c, + res/res_odbc.c: Ensure that all ast_datastore_info structures are + 'const'. While addressing a bug, I came across a instance of + 'struct ast_datastore_info' that was not declared 'const'. Since + the API already expects them to be 'const', this patch changes + the declarations of all existing instances that were not already + declared that way. ........ Merged revisions 370183 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370184 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-18 15:15 +0000 [r370171-370177] Joshua Colp + + * res/res_rtp_asterisk.c: Fix a crash in pjnath when starting an + ICE connectivity check and immediately destroying the ICE + session. The initial ICE connectivity check is scheduled as a + timer item that is to be executed immediately. It is possible for + this timer item to start executing while the ICE session it is + working on is destroyed. To reduce the chance of this any timer + items that need to be immediately executed will be executed + within the thread that has started the initial ICE connectivity + check. + + * channels/chan_sip.c, include/asterisk/rtp_engine.h: Fix a crash + occurring as a result of excess stack usage. This fix involves + moving the allocation of some temporary codec structures to the + heap and also reduces the number of maximum payloads to something + more sane for both regular and low memory builds. (closes issue + ASTERISK-20140) Reported by: jonnt + +2012-07-18 07:17 +0000 [r370165] Igor Goncharovskiy + + * channels/chan_unistim.c, configs/unistim.conf.sample, CHANGES: + Added option 'interdigit_timer' to unistim.conf to make able + controll hardcoded dial timeout constant. + +2012-07-17 19:05 +0000 [r370152-370157] Joshua Colp + + * res/res_xmpp.c: Add pubsub unsubscription support so + subscriptions do not linger for MWI and device state progatation. + The pubsub code did not attempt to remove subscriptions at all. + This has now changed so that if a client is being disconnected it + will unsubscribe. It will also unsubscribe at connection time so + if it unexpectedly disconnected duplicate subscriptions will not + occur. (closes issue ASTERISK-19882) Reported by: mattvryan + + * include/asterisk/xmpp.h, res/res_xmpp.c: Fix a crash as a result + of propagating MWI or device state over XMPP when the client is + disconnected. The MWI and device state propagation code wrongly + assumes that an XMPP client connection will remain established at + all times. This fix corrects that by making the lifetime of the + subscription the same as the lifetime of the connection itself. + As the connection is established and disconnected the + subscription itself is created and destroyed. (closes issue + ASTERISK-18078) Reported by: elguero + +2012-07-16 19:58 +0000 [r370133] Walter Doekes + + * /, channels/chan_sip.c: Code cleanup and bugfix in chan_sip + outboundproxy parsing. The bug was clearing the global + outboundproxy when a peer-specific outboundproxy was bad. The + cleanup reduces duplicate code. Review: + https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark + Michelson ........ Merged revisions 370131 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370132 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-16 19:14 +0000 [r370111-370126] Joshua Colp + + * res/res_xmpp.c: Fix an issue where a service discovery request + could crash Asterisk. A server sending a service discovery + request to us may or may not put a from attribute in the message. + If the from attribute is present use it in the to attribute for + the result. If the from attribute is not present do not add a to + attribute. (issue ASTERISK-16203) Reported by: wubbla + + * res/res_xmpp.c: Fix a bug where some XMPP servers would reject + authentication. We need to use the user portion of the JID and + not the full configured username. + + * res/res_xmpp.c: Add missing namespace for old non-SASL based + authentication. + + * channels/chan_sip.c: Fix a bug exposed by the testsuite where + text streams would no longer be parsed correctly. + +2012-07-16 14:02 +0000 [r370083] Kinsey Moore + + * /, UPGRADE-10.txt, CHANGES, UPGRADE-1.8.txt: Add comments about + the BUILD_NATIVE change This is a significant change and mention + of it should have gone into UPGRADE.txt and CHANGES. ........ + Merged revisions 370081 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370082 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-16 12:58 +0000 [r370072-370073] Joshua Colp + + * res/res_xmpp.c: Fix an issue where specifying the resource in the + username would cause authentication to fail. + + * channels/sip/sdp_crypto.c, channels/chan_sip.c, + channels/sip/security_events.c, + include/asterisk/http_websocket.h, configs/sip.conf.sample, + CHANGES, res/res_http_websocket.c, channels/sip/include/sip.h: + Add support for SIP over WebSocket. This allows SIP traffic to be + exchanged over a WebSocket connection which is useful for rtcweb. + Review: https://reviewboard.asterisk.org/r/2008 + +2012-07-16 07:38 +0000 [r370066-370067] Igor Goncharovskiy + + * channels/chan_unistim.c: Deactivate timer for dialing entered + number on hook switch hang up. (closes issue ASTERISK-19554) + Reported by: Stefano Villani + + * channels/chan_unistim.c, contrib/unistimLang/fr.po (added), + CHANGES: Add French translation for chan_unistim phones on-screen + menus. + +2012-07-13 18:41 +0000 [r370055-370060] Joshua Colp + + * include/asterisk/format.h, res/res_format_attr_h263.c (added), + res/res_format_attr_h264.c (added): Reduce memory consumption and + add the H.264 and H.263 modules I shamefully neglected to add. + + * main/format.c, channels/chan_sip.c, main/translate.c, + include/asterisk/format.h, res/res_format_attr_silk.c, + res/res_format_attr_celt.c: Add support for parsing SDP + attributes, generating SDP attributes, and passing it through. + This support includes codecs such as H.263, H.264, SILK, and + CELT. You are able to set up a call and have attribute + information pass. This should help considerably with video calls. + Review: https://reviewboard.asterisk.org/r/2005/ + +2012-07-13 00:05 +0000 [r370048] Tzafrir Cohen + + * contrib/scripts/live_ast: live_ast: don't set working directory + contrib/scripts/live_ast currently assumes that it is being run + from the top-level directory of the source tree. It creates a + script that will also set the working directory. This fix avoids + the need to set the working directory if the caller sets + LIVE_AST_BASE_DIR instead. It relies on realpath for that. If + realpath is not available, it will fall back to the original + behaviour. Review: https://reviewboard.asterisk.org/r/2027/ + +2012-07-12 21:43 +0000 [r370043] Terry Wilson + + * include/asterisk/config_options.h, + configs/config_test.conf.sample, main/config_options.c, + tests/test_config.c: Handle deprecated (aliased) option names + with the config options api Add a simple way to register + "deprecated" option names that alias to a different "current" + name. Review: https://reviewboard.asterisk.org/r/2026/ + +2012-07-12 20:28 +0000 [r370037] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Add missing + ast_hangup() calls on some analog exception paths. Make starting + analog_ss_thread() or __analog_ss_thread() failure paths hangup + the channel. ........ Merged revisions 370017 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370025 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-12 20:06 +0000 [r369995-370016] Kinsey Moore + + * /, channels/chan_sip.c: Include Expires header for SIP PUBLISH + requests RFC3903 requres SIP PUBLISH requests to have Expires + headers, so add them. Review: + https://reviewboard.asterisk.org/r/2003/ Patch-by: gareth + ........ Merged revisions 370014 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 370015 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Prevent double uri_escaping in chan_sip + when pedantic is enabled If pedantic mode is enabled, outbound + invites will have double-escaped contacts. This avoids setting an + already-escaped string into a field where it is expected to be + unescaped. (closes issue ASTERISK-20023) Reported by: Walter + Doekes ........ Merged revisions 369993 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369994 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-12 14:38 +0000 [r369972-369974] Michael L. Young + + * /, funcs/func_math.c: Correct Documentation For DEC Function The + documentation for DEC in func_math.c was incorrect. Looks like a + copy and paste error. (Closes issue ASTERISK-20095) Reported by: + Billy Chia Tested by: Michael L. Young Patches: func_math.patch + uploaded by Billy Chia (license 6381) ........ Merged revisions + 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 369971 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * funcs/func_math.c: Reverting last merge since it wasn't completed + properly. + + * funcs/func_math.c: Correct Documentation For DEC Function The + documentation for DEC in func_math.c was incorrect. Looks like a + copy and paste error. (Closes issue ASTERISK-20095) Reported by: + Billy Chia Tested by: Michael L. Young Patches: func_math.patch + uploaded by Billy Chia (license 6381) ........ Merged revisions + 369970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2012-07-11 18:33 +0000 [r369959] Jonathan Rose + + * include/asterisk/acl.h, channels/chan_sip.c, + include/asterisk/config.h, main/acl.c, + include/asterisk/channel.h, configs/manager.conf.sample, + channels/chan_iax2.c, CHANGES, main/named_acl.c (added), + main/config.c, main/loader.c, configs/iax.conf.sample, + main/manager.c, include/asterisk/event_defs.h, + configs/extconfig.conf.sample, configs/sip.conf.sample, + channels/sip/include/sip.h, main/asterisk.c, + configs/acl.conf.sample (added): Named ACLs: Introduces a system + for creating and sharing ACLs This patch adds Named ACL + functionality to Asterisk. This allows system administrators to + define an ACL and refer to it by a unique name. Configurable + items can then refer to that name when specifying access control + lists. It also includes updates to all core supported consumers + of ACLs. That includes manager, chan_sip, and chan_iax2. This + feature is based on the deluxepine-trunk by Olle E. Johansson and + provides a subset of the Named ACL functionality implemented in + that branch. For more information on this feature, see acl.conf + and/or the Asterisk wiki. Review: + https://reviewboard.asterisk.org/r/1978/ + +2012-07-11 17:16 +0000 [r369940] Tilghman Lesher + + * /, main/ast_expr2.h, main/ast_expr2f.c, res/ael/ael_lex.c, + funcs/func_realtime.c, main/ast_expr2.c: Allow the REALTIME() + function to report errors back to the caller. Also, do more error + checking on the arguments specified to the REALTIME() function + and clarify the documentation. While I was editing the file, a + few coding guidelines fixups, as well. Review: + https://reviewboard.asterisk.org/r/2031/ ........ Merged + revisions 369937 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369938 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-11 17:14 +0000 [r369939] Matthew Jordan + + * main/features.c: Don't perform an XInclude to a document node + that may not always be present Because some of the manager events + are defined in the top of the source, due to the macro calls not + containing all necessary information to have the documentation + colocated with the call itself, several include statements were + failing when built with 'make'. While this did not cause any + problems in compilation or validation, it did result in a number + of warnings being dumped to stderr. This patch changes those + references such that they always resolve, regardless of the + documentation build options. + +2012-07-11 16:42 +0000 [r369936] Joshua Colp + + * channels/chan_motif.c: Do not consider failure to read the + configuration file in chan_motif to be a show stopper for loading + Asterisk by returning decline instead of failure. (closes issue + ASTERISK-20103) Reported by: Terry Wilson + +2012-07-11 02:06 +0000 [r369905-369910] Matthew Jordan + + * main/cdr.c, main/channel.c, channels/sig_analog.c, main/logger.c, + channels/sig_pri.c, main/asterisk.c, main/loader.c: Fix + validation errors when producing documentation using default + build script The awk script parses out the first instance of the + DOCUMENTATION tag that it finds within a file. If a file did not + previously have a DOCUMENTATION tag but received one due to it + having an AMI event, then the XML fragment associated with the + AMI event was erroneously placed in the resulting XML file. + Without the python scripts, these XML fragments will not + validate. This patch adds DOCUMENTATION tags at the top of those + files that did not previously have them to prevent the awk script + from pulling AMI event documentation. + + * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c, + channels/chan_local.c, channels/sig_analog.c, main/manager.c, + channels/chan_agent.c, main/features.c, main/logger.c, + channels/sig_pri.c, doc/appdocsxml.dtd, main/asterisk.c, + main/loader.c: Add some additional documentation for core AMI + events This patch adds some basic documentation for a number of + modules. This includes core source files in Asterisk (those in + main), as well as chan_agent, chan_dahdi, chan_local, sig_analog, + and sig_pri. The DTD has also been updated to allow referencing + of AMI commands. + +2012-07-10 15:36 +0000 [r369900] Kinsey Moore + + * channels/chan_sip.c: Fix failing SDP_offer_answer test Asterisk + now generates image stream declinations with the same transport + case that it used to before the stream declination improvements. + (udptl vs UDPTL) (closes issue SWP-4736) + +2012-07-10 15:25 +0000 [r369873-369898] Joshua Colp + + * channels/chan_motif.c: Add additional description stanza names + from the old Google Talk protocol which is used with Google + Voice. (closes issue ASTERISK-20114) Reported by: Malcolm + Davenport + + * channels/chan_motif.c: Respect codec preference order when adding + codecs to a media description. This change allows an endpoint in + motif.conf to be configured with a preference of G.722 and + fallback of ulaw. With Google this allows communication with + Google Talk clients to use G.722 while when using Google Voice + ulaw will be used. (closes issue ASTERISK-20114) Reported by: + Malcolm Davenport + +2012-07-10 13:40 +0000 [r369872] Kinsey Moore + + * main/pbx.c, /, apps/app_stack.c: Improve Goto and GotoIf related + documentation Correct documentation on labeliftrue and + labeliffalse parameters of GotoIf() and update several other + locations that use the same syntax. (closes issue ASTERISK-20007) + Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged + revisions 369869 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369871 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-10 13:34 +0000 [r369870] Matthew Jordan + + * main/libasteriskssl.c: Fix initial loading problem with res_curl + When the OpenSSL duplicate initialization issues were resolved in + r351447, res_curl could fail to load if it checked + SSL_library_init after SSL initialization completed. This is due + to the SSL_library_init stub returning a value of 0 for success, + as opposed to a value of 1. OpenSSL uses a value of 1 to indicate + success - in fact, SSL_library_init is documented to always + return 1. Interestingly, the CURL libraries actually checked the + return value - the fact that nothing else that depends on OpenSSL + was having problems loading probably means they don't check the + return value. (closes issue AST-924) Reported by: Guenther + Kelleter patches: (AST-924.patch license #6372 uploaded by + Guenther Kelleter) + +2012-07-10 11:49 +0000 [r369837-369864] Joshua Colp + + * res/res_rtp_asterisk.c, channels/chan_motif.c: Add required items + for Google video support. This adds legacy STUN support for RTCP + sockets, adds RTCP candidates to the Google transport + information, and adds required codec parameters. (closes issue + ASTERISK-20106) Reported by: Malcolm Davenport + + * main/stun.c: When receiving a STUN binding request send one out + as the Google Talk client uses this as a method to determine if + the remote party is still reachable or not. Failure to do this + results in the Google Talk client ignoring RTP packets after a + specific period of time. This is also done as a result of + receiving a STUN binding request so that the username information + can be used from the inbound request, thus not requiring it to be + stored on a per candidate basis. (closes issue ASTERISK-20107) + Reported by: Malcolm Davenport + + * channels/chan_sip.c: Add support for exposing the received + contact URI and also for setting the request URI in messages. + (closes issue AST-911) + + * channels/chan_motif.c: Force the clock rate of G.722 to be 16000 + when using the Google transports as it is 8000 elsewhere. (closes + issue ASTERISK-20105) Reported by: Malcolm Davenport + + * configs/motif.conf.sample: Document that multiple endpoints using + the same connection is not supported. (closes issue + ASTERISK-20104) Reported by: Malcolm Davenport + +2012-07-09 17:07 +0000 [r369820] Jason Parker + + * configs/sip_notify.conf.sample, /: Add Digium phones context to + sip_notify sample config. This makes it so that they can be + reconfigured remotely. (closes issue ASTERISK-19910) ........ + Merged revisions 369818 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369819 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-09 16:44 +0000 [r369811-369817] Joshua Colp + + * res/res_rtp_asterisk.c: Fix an issue where media would not flow + for situations where the legacy STUN code is in use. The STUN + packets should *not* be blocked by strict RTP. (closes issue + ASTERISK-20102) Reported by: Malcolm Davenport + + * res/res_xmpp.c: Add additional namespaces for Google Talk which + are used for the gmail client. (closes issue ASTERISK-20101) + Reported by: Malcolm Davenport + + * channels/chan_motif.c: Fix dependency to be on res_xmpp. Long ago + in a galaxy far far away it used to use res_jabber. + +2012-07-09 14:54 +0000 [r369794] Jonathan Rose + + * /, channels/chan_sip.c: chan_sip: Fix small behavioral change + accidentally introduced in r369750 When removing the warning for + AST_CONTROL_FLASH from sip_indicate, I also inadvertently changed + the return value, which would likely make the indication not be + sent in audio. This fixes that while still removing the warning + message. ........ Merged revisions 369792 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369793 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-07 17:06 +0000 [r369769] Joshua Colp + + * res/res_xmpp.exports.in (added), include/asterisk/xmpp.h, + channels/chan_motif.c (added), UPGRADE.txt, + channels/chan_gtalk.c, res/res_xmpp.c, CHANGES, res/res_jabber.c, + configs/motif.conf.sample (added): Add a new unified Jingle, + Google Jingle, and Google Talk channel driver written from + scratch called chan_motif. This channel driver is a replacement + for both chan_gtalk and chan_jingle but adds additional features + not found in either. These features include full configuration + reload, video, full codec support, bidirectional cause code + mapping, hold, unhold, and ringing indication. It is also + compliant with the current published Jingle and Google Jingle + specifications. The original Google Talk protocol is also + supported for Google Voice interoperability. You may ask yourself + though where the name motif comes from... and I would say to + you... music! motif: a perceivable or salient recurring fragment + or succession of notes Sorta like a jingle! Review: + https://reviewboard.asterisk.org/r/1917/ + +2012-07-06 22:03 +0000 [r369765] Kinsey Moore + + * channels/chan_dahdi.c, channels/sig_analog.c, + channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c: + Remove unnecessary generation of informational cause frames It is + not necessary to generate information cause code frames on every + protocol event that occurs. This removes all the instances where + the frame was not conveying a cause code and was instead just + conveying a protocol-specific message. This also corrects the + generation of the message associated with disconnects for MFC/R2 + to use the MFC/R2 specific text for the disconnect cause. + +2012-07-06 21:28 +0000 [r369764] Jonathan Rose + + * /, channels/chan_sip.c: chan_sip: Add case for FLASH control + frames so that we don't display a warning. chan_sip channels can + receive flash control frames when connected to analog phones and + possibly for other reasons. There really isn't a reason to warn + when these frames are received, we can safely ignore them. + Patches: dahdi_sip_flash.diff uploaded by Jonathan Rose (license + 6182) ........ Merged revisions 369750 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369751 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-06 18:49 +0000 [r369710-369733] Mark Michelson + + * main/tcptls.c, /: Remove a superfluous and dangerous freeing of + an SSL_CTX. The problem here is that multiple server sessions + share a SSL_CTX. When one session ended, the SSL_CTX would be + freed and set NULL, leaving the other sessions unable to + function. The code being removed is superfluous because the + SSL_CTX structures for servers will be properly freed when + ast_ssl_teardown is called. (closes issue ASTERISK-20074) + Reported by Trevor Helmsley Patches: ASTERISK-20074.diff uploaded + by Mark Michelson (license #5049) Testers: Trevor Helmsley + ........ Merged revisions 369731 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369732 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/bridging.c: Fix bridging thread leak. The bridge thread + was exiting but was never being reaped using pthread_join(). This + has been fixed now by calling pthread_join() in + ast_bridge_destroy(). (closes issue ASTERISK-19834) Reported by + Marcus Hunger Review: https://reviewboard.asterisk.org/r/2012 + ........ Merged revisions 369708 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369709 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-06 14:32 +0000 [r369703] Joshua Colp + + * res/pjproject/pjnath/include/pjnath/ice_session.h, + res/pjproject/pjnath/src/pjnath/ice_session.c: Import revision + 4196 from pjproject trunk. Fix a crash issue when starting ICE + connectivity checks and immediately destroying the ICE session. + This was exposed by the SIP CCSS test. Full fix for this issue + will be worked on as a medium to long term roadmap item. pjroject + issue viewable at https://trac.pjsip.org/repos/ticket/1548 + +2012-07-05 21:36 +0000 [r369681] Matthew Jordan + + * res/res_stun_monitor.c, CHANGES: Add 'stun show status' command + This patch adds a new CLI command, 'stun show status'. This + command will show a table describing all known STUN servers and + statuses. (closes issue ASTERISK-18046) Reported by: Jeremy + Kister Tested by: Jeremy Kister patches: + (stun-show-status-v4-trunk.patch license #6232 uploaded by Jeremy + Kister) Review: https://reviewboard.asterisk.org/r/2001 + +2012-07-05 19:36 +0000 [r369677] Richard Mudgett + + * res/pjproject/pjmedia/include/pjmedia, + res/pjproject/pjsip/include/pjsip, + res/pjproject/pjlib/include/pj/compat, + res/pjproject/pjmedia/include/pjmedia-codec: Make res/pjproject + ignore more files. + +2012-07-05 19:36 +0000 [r369676] Kinsey Moore + + * /, apps/app_voicemail.c: AST-2012-011: Resolve heap corruption + issue with voicemail The heard and deleted arrays in the + voicemail state structure were not handled properly following the + memory leak fix in r354890 and a fix for an invalid free in + r356797. This could result in accessing and writing into freed + memory. The allocation for these arrays has been reworked to + avoid the possibility of invalid frees, access of freed memory, + and crashes that were occurring as a result of this. Locking + around accesses and modifications of the voicemail state + structure members dh_arraysize, heard, and deleted has been added + to prevent simultaneous modification and access when IMAP storage + is in use. If IMAP storage is not in use, this locking is not + compiled in. Review: https://reviewboard.asterisk.org/r/1994/ + (closes issue ASTERISK-19923) Reported by: Dan Delaney Tested by: + Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by + kmoore (license 6273) ........ Merged revisions 369652 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369653 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-05 19:32 +0000 [r369666-369673] Richard Mudgett + + * res/pjproject/pjsip/src/pjsip-ua, + res/pjproject/pjsip-apps/src/ipjsystest/ipjsystest.xcodeproj, + res/pjproject/pjnath/src/pjnath-test, + res/pjproject/third_party/build/speex, + res/pjproject/third_party/build/gsm/output, + res/pjproject/pjmedia/include/pjmedia-codec, + res/pjproject/third_party/build/baseclasses, + res/pjproject/third_party/build/srtp, + res/pjproject/pjsip-apps/src/samples, + res/pjproject/pjlib-util/lib, res/pjproject/pjmedia/bin, + res/pjproject/pjlib/include/pj++, + res/pjproject/tests/pjsua/scripts-call, + res/pjproject/third_party/srtp/doc, + res/pjproject/pjsip-apps/src/pocketpj/output, + res/pjproject/pjnath/bin, + res/pjproject/third_party/srtp/crypto/replay, + res/pjproject/pjsip/include/pjsip, + res/pjproject/third_party/build/speex/speex, + res/pjproject/build.symbian, res/pjproject/third_party/bin, + res/pjproject/pjsip/src/pjsua-lib, + res/pjproject/third_party/srtp/include, + res/pjproject/third_party/portaudio/doc, res/pjproject/lib, + res/pjproject/pjmedia/include/pjmedia-videodev, + res/pjproject/pjlib/bin, + res/pjproject/third_party/srtp/crypto/cipher, + res/pjproject/third_party/build/speex/output, + res/pjproject/pjlib-util/src/pjlib-util, + res/pjproject/third_party/portaudio/test, + res/pjproject/third_party/build/gsm, + res/pjproject/third_party/portaudio/include, + res/pjproject/pjsip-apps/src/pjsua_wince, + res/pjproject/pjsip/include/pjsip-simple, + res/pjproject/pjmedia/src/pjmedia-codec, + res/pjproject/tests/pjsua, + res/pjproject/pjsip-apps/src/pocketpj/res, + res/pjproject/pjsip-apps/src/3rdparty_media_sample, + res/pjproject/third_party/gsm/inc, + res/pjproject/pjsip-apps/build/wince-evc4, + res/pjproject/pjsip-apps/src/ipjsua/Resources-iPad, + res/pjproject/third_party/portaudio/src/hostapi, + res/pjproject/third_party/portaudio/build, res/pjproject/build, + res/pjproject/third_party/build/resample, + res/pjproject/third_party/speex/include, + res/pjproject/pjsip/src/pjsip, + res/pjproject/pjlib/build/wince-evc4, + res/pjproject/pjsip-apps/src/symbian_ua_gui/group, + res/pjproject/pjsip-apps/src/symbian_ua, + res/pjproject/tests/pjsua/wavs, + res/pjproject/third_party/portaudio/src/os/win, + res/pjproject/pjsip-apps/src/ipjsua/Classes, + res/pjproject/pjmedia/include/pjmedia, + res/pjproject/tests/pjsua/scripts-sendto, + res/pjproject/third_party/gsm/src, + res/pjproject/third_party/portaudio/build/msvc, + res/pjproject/pjsip-apps/src/confbot, + res/pjproject/pjnath/src/pjturn-client, + res/pjproject/pjlib-util/build/output, + res/pjproject/third_party/BaseClasses, + res/pjproject/third_party/portaudio/src/hostapi/wasapi, + res/pjproject/third_party/portaudio/src/hostapi/wdmks, + res/pjproject/pjlib/src/pj/compat, + res/pjproject/third_party/srtp/crypto/include, + res/pjproject/third_party/speex/include/speex, + res/pjproject/third_party/gsm/add-test, + res/pjproject/pjsip/build, + res/pjproject/pjsip-apps/src/pjsua_wince/output, + res/pjproject/third_party/gsm/lib, res/pjproject/pjsip, + res/pjproject/pjsip-apps/src/pjsystest, + res/pjproject/third_party/portaudio/src, + res/pjproject/third_party/speex/libspeex, + res/pjproject/pjsip/build/wince-evc4/output, + res/pjproject/pjlib-util/src/pjlib-util-test, + res/pjproject/pjsip-apps/src/symsndtest, + res/pjproject/third_party/srtp/tables, + res/pjproject/third_party/g7221, res/pjproject/pjmedia/include, + res/pjproject/pjlib/include/pj, + res/pjproject/third_party/build/portaudio/output, + res/pjproject/pjsip-apps/bin, + res/pjproject/pjsip-apps/src/ipjsua/ipjsua.xcodeproj, + res/pjproject/pjsip-apps/src/pjsua, + res/pjproject/third_party/srtp/test, + res/pjproject/pjsip/include/pjsip-ua, + res/pjproject/third_party/resample, + res/pjproject/third_party/build/ilbc, + res/pjproject/pjmedia/src/pjmedia-audiodev, + res/pjproject/pjsip-apps/src/ipjsua, + res/pjproject/third_party/srtp/srtp, + res/pjproject/third_party/build/milenage, + res/pjproject/pjmedia/src/pjmedia, res/pjproject/pjlib-util, + res/pjproject/third_party/portaudio/src/common, + res/pjproject/third_party/portaudio/bindings/cpp, + res/pjproject/pjlib-util/build/wince-evc4/output, + res/pjproject/third_party/srtp/crypto/kernel, + res/pjproject/tests/pjsua/scripts-pres, res/pjproject/pjnath, + res/pjproject/pjsip/build/output, + res/pjproject/pjsip-apps/build/output, + res/pjproject/pjsip-apps/build, res/pjproject/tests/automated, + res/pjproject/pjnath/build/wince-evc4/output, + res/pjproject/third_party/portaudio/src/hostapi/asio, + res/pjproject/pjnath/include/pjnath, + res/pjproject/pjsip/src/test, + res/pjproject/pjsip-apps/src/symbian_ua_gui/gfx, + res/pjproject/pjsip/bin, + res/pjproject/third_party/build/portaudio, + res/pjproject/pjlib/build/output, res/pjproject/pjmedia/src, + res/pjproject/pjlib/src/pj, res/pjproject/pjlib, + res/pjproject/pjlib/build/wince-evc4/output, + res/pjproject/pjmedia/src/test/vectors, + res/pjproject/third_party/portaudio/src/hostapi/jack, + res/pjproject/pjmedia/src/pjmedia-codec/g722, + res/pjproject/third_party/portaudio/src/hostapi/coreaudio, + res/pjproject/pjmedia/build/output, + res/pjproject/pjlib-util/include/pjlib-util, + res/pjproject/third_party/portaudio/src/hostapi/asihpi, + res/pjproject/third_party/milenage, res/pjproject/pjnath/src, + res/pjproject/tests/pjsua/scripts-run, + res/pjproject/pjlib-util/build/wince-evc4, + res/pjproject/pjmedia/lib, res/pjproject/pjmedia/src/test, + res/pjproject/third_party/speex/symbian, + res/pjproject/third_party/speex/win32, + res/pjproject/third_party/srtp/crypto/test, + res/pjproject/pjlib-util/bin, + res/pjproject/third_party/portaudio/build/scons, + res/pjproject/tests/cdash, + res/pjproject/tests/pjsua/scripts-media-playrec, + res/pjproject/third_party/build/portaudio/src, + res/pjproject/pjlib/src, res/pjproject/third_party/mp3, + res/pjproject/pjnath/lib, res/pjproject/third_party/build/g7221, + res/pjproject/third_party/gsm/man, + res/pjproject/third_party/portaudio/src/os/unix, + res/pjproject/third_party/portaudio/bindings, + res/pjproject/pjsip-apps/src/python, + res/pjproject/pjnath/src/pjnath, res/pjproject/third_party/lib, + res/pjproject/third_party/portaudio/src/os/mac_osx, + res/pjproject/third_party/srtp/crypto/ae_xfm, + res/pjproject/pjsip-apps/bin/samples, + res/pjproject/pjnath/src/pjturn-srv, + res/pjproject/third_party/portaudio/pablio, + res/pjproject/pjlib/lib, res/pjproject/third_party/g7221/decode, + res/pjproject/pjlib/include/pj/compat, + res/pjproject/third_party/gsm, + res/pjproject/third_party/build/baseclasses/output, + res/pjproject/third_party/build/srtp/output, + res/pjproject/third_party/srtp, res/pjproject/pjnath/build, + res/pjproject/tests/pjsua/scripts-sipp, res/pjproject/pjsip-apps, + res/pjproject/pjnath/build/wince-evc4, + res/pjproject/third_party/srtp/crypto/rng, + res/pjproject/pjsip/build/wince-evc4, + res/pjproject/pjsip-apps/build/wince-evc4/output, + res/pjproject/third_party/gsm/tst, + res/pjproject/third_party/portaudio/src/hostapi/dsound, + res/pjproject/third_party/portaudio/testcvs, + res/pjproject/pjsip-apps/src/ipjsystest/Classes, + res/pjproject/pjlib/build, res/pjproject/third_party/portaudio, + res/pjproject/third_party/portaudio/src/hostapi/wmme, + res/pjproject/pjlib-util/docs, + res/pjproject/pjmedia/include/pjmedia-audiodev, + res/pjproject/pjsip-apps/src/vidgui, + res/pjproject/pjlib/src/pjlib-test, + res/pjproject/pjsip-apps/src/py_pjsua, + res/pjproject/third_party/portaudio/src/os, + res/pjproject/pjsip/include, + res/pjproject/pjmedia/build/wince-evc4, + res/pjproject/pjmedia/src/pjmedia-videodev, + res/pjproject/pjsip-apps/src, res/pjproject/third_party/speex, + res/pjproject/third_party/gsm/tls, + res/pjproject/third_party/g7221/common, + res/pjproject/tests/pjsua/tools, + res/pjproject/third_party/resample/include, + res/pjproject/third_party/build/samplerate/output, + res/pjproject/third_party/build/samplerate, + res/pjproject/third_party/gsm/bin, + res/pjproject/pjsip/src/pjsip-simple, + res/pjproject/third_party/g7221/encode, + res/pjproject/pjlib/src/pjlib-samples, + res/pjproject/pjsip-apps/lib, + res/pjproject/pjsip-apps/src/ipjsystest, + res/pjproject/pjlib-util/include, + res/pjproject/third_party/build/resample/output, + res/pjproject/third_party/build/ilbc/output, + res/pjproject/third_party/srtp/crypto, + res/pjproject/pjsip-apps/src/python/samples, res/pjproject/tests, + res/pjproject/pjsip-apps/src/symbian_ua_gui/sis, + res/pjproject/pjnath/include, + res/pjproject/pjsip-apps/src/symbian_ua_gui, + res/pjproject/pjmedia/build, res/pjproject/pjmedia, + res/pjproject/third_party/build/milenage/output, + res/pjproject/pjlib-util/build, res/pjproject/pjsip/src, + res/pjproject/pjmedia/build/wince-evc4/output, + res/pjproject/third_party/portaudio/src/hostapi/alsa, + res/pjproject/pjsip-apps/docs, + res/pjproject/pjsip-apps/src/symbian_ua_gui/inc, + res/pjproject/pjsip-apps/src/symbian_ua_gui/data, + res/pjproject/tests/pjsua/scripts-pesq, + res/pjproject/third_party/srtp/pjlib, + res/pjproject/pjlib/include, res/pjproject/pjnath/build/output, + res/pjproject/third_party/srtp/crypto/hash, + res/pjproject/build/vs, res/pjproject/pjlib/docs, + res/pjproject/third_party/build, + res/pjproject/third_party/resample/src, + res/pjproject/third_party, res/pjproject/pjlib/src/pjlib++-test, + res/pjproject/third_party/build/g7221/output, + res/pjproject/third_party/srtp/crypto/math, + res/pjproject/pjsip/lib, res/pjproject/pjsip-apps/src/pocketpj, + res/pjproject/tests/pjsua/scripts-recvfrom, + res/pjproject/third_party/portaudio/build/dev-cpp, + res/pjproject/pjsip/include/pjsua-lib, + res/pjproject/pjsip-apps/src/symbian_ua_gui/src, res/pjproject, + res/pjproject/third_party/portaudio/src/hostapi/oss, + res/pjproject/pjlib-util/src, res/pjproject/third_party/ilbc: + Make res/pjproject ignore some generated files. + + * include/asterisk/utils.h: Tweak some comments and whitespace in + utils.h + +2012-07-05 18:11 +0000 [r369644] Jonathan Rose + + * apps/app_mixmonitor.c: app_mixmonitor: Fix a reference leak in + manager_mixmonitor function Manager_mixmonitor included an early + return on failed executions of mixmonitor that would result in a + leaked channel reference. (closes issue ASTERISK-19943) Reported + by: Mark Murawski Patches: mixmonitor-trunk-368394.patch uploaded + by Mark Murawski (license 5791) + +2012-07-05 17:03 +0000 [r369628] Matthew Jordan + + * /, channels/chan_sip.c: Do not send a BYE when a provisional + response arrives during a re-INVITE Commits r369557 and r369579 + were done to improve handling of re-INVITEs when the UA that was + supposed to receive the re-INVITE fails to respond. A limitation + of those patches occurred when a UA sent a provisional response + to the re-INVITE. This triggered a sending of a BYE in + check_pending. This patch tweaks the handling of the re-INVITE + such that a BYE is not sent in response to those messages. (issue + ASTERISK-19992) Reported by: Steve Davies Tested by: Steve Davies + patches: (reinvite_tweak.diff license #5012 by Steve Davies) + ........ Merged revisions 369626 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369627 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-05 11:42 +0000 [r369602-369620] Alexandr Anikin + + * addons/ooh323c/src/ooCmdChannel.c, + addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c: + Fix dev mode ooh323 warnings + + * addons/chan_ooh323.c, addons/ooh323c/src/ooq931.h, + addons/ooh323c/src/ooCalls.h, configs/chan_ooh323.conf.sample + (removed), addons/ooh323c/src/ooh323ep.c, CHANGES, + configs/ooh323.conf.sample (added), + addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooStackCmds.c, + addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooLogChan.h, + addons/ooh323c/src/ooStackCmds.h, addons/ooh323c/src/ooh245.c, + addons/ooh323cDriver.c, addons/ooh323c/src/ooh245.h, + addons/ooh323c/src/ooCmdChannel.c, addons/ooh323c/src/ooq931.c: + Added direct media support to ooh323 channel driver options are + documented in config sample sample config rename to proper name - + ooh323.conf To change media address ooh323 send empty TCS if + there was completed TCS exchange or send facility + forwardedelements with new fast start proposal if not. Then close + transmit logical channels and renew TCS exchange. If new fast + start proposal is received then ooh323 stack call back channel + driver routine to change rtp address in the rtp instance. If + empty TCS is received then close transmit logical channels and + renew TCS exchange Review: + https://reviewboard.asterisk.org/r/1607/ + + * addons/ooh323cDriver.c: fix small mistake in the previous + + * addons/ooh323c/src/ooTimer.c, addons/ooh323c/src/ooCapability.c, + addons/ooh323c/src/decode.c, addons/ooh323c/src/perutil.c, + addons/ooh323cDriver.c, addons/ooh323c/src/ooSocket.c, + addons/ooh323c/src/ooq931.c: Fix modern gcc warning Review: + https://reviewboard.asterisk.org/r/1767 + +2012-07-03 17:07 +0000 [r369559-369581] Terry Wilson + + * /, channels/chan_sip.c: More improvements to re-INVITEs timing + out after a provisional response There is no need to call + check_pendings() on a final response to an INVITE when destroying + the scheduler entry as it will be done later during normal + processing. (issue ASTERISK-19992) ........ Merged revisions + 369579 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 369580 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Better handle + re-INVITEs with provisional but no final repsonses A previous + attempt at fixing this issue had negative side effects related to + attended transfers which this patch should resolve. Many thanks + to Steve Davies for all of the good suggestions and testing. + (closes issue ASTERISK-19992) Reported by: Steve Davies Tested + by: Steve Davies, Terry Wilson Review: + https://reviewboard.asterisk.org/r/2009/ ........ Merged + revisions 369557 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369558 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-07-02 14:06 +0000 [r369517-369527] Joshua Colp + + * configs/xmpp.conf.sample (added), include/asterisk/xmpp.h + (added), configs/cli_aliases.conf.sample, res/res_xmpp.c (added): + Add a cleaned up drop-in replacement for res_jabber called + res_xmpp. This provides the same externally facing functionality + but is implemented differently internally. This is currently not + built by default but this will be changed once chan_jingle2 + (insert actual name in your head when reading this after it has + been merged) is in the tree. Review: + https://reviewboard.asterisk.org/r/1983/ + + * res/res_rtp_asterisk.c: Ensure the timer heap is protected by a + lock. + + * res/pjproject/pjlib/include/pj/config_site.h: Enable IPv6 support + in pjproject. + + * res/res_rtp_asterisk.c: Don't try to send connectivity checks on + RTCP if RTCP is no longer present and don't do multiple ICE + connectivity checks at once. + + * res/pjproject/pjlib/src/pj/sock_qos_common.c (added), + res/pjproject/pjlib-util/src/pjlib-util/crc32.c (added), + res/pjproject/pjsip/src/pjsip-simple/xpidf.c (added), + res/pjproject/third_party/gsm/src/gsm_implode.c (added), + res/pjproject/tests/pjsua/scripts-sipp/uas-cancel-no-final.xml + (added), res/pjproject/build.symbian/pjmedia.mmp (added), + res/pjproject/third_party/build/portaudio/src/pa_hostapi.h + (added), res/pjproject/pjlib/src/pjlib-test/fifobuf.c (added), + res/pjproject/pjlib/src/pj/file_access_unistd.c (added), + res/pjproject/third_party/gsm/src/toast_ulaw.c (added), + res/pjproject/pjsip/include/pjsip/sip_transport_tls.h (added), + res/pjproject/pjsip/include/pjsip/sip_multipart.h (added), + res/pjproject/pjmedia/src/pjmedia/errno.c (added), + res/pjproject/pjsip-apps/src/pjsua_wince/pjsua_wince.vcp (added), + res/pjproject/third_party/speex/COPYING (added), + res/pjproject/pjlib/src/pj/os_core_darwin.m (added), + res/pjproject/third_party/ilbc/packing.c (added), + res/pjproject/third_party/build/portaudio/src/pa_mac_core_internal.h + (added), + res/pjproject/tests/pjsua/scripts-sendto/300_srtp_receive_crypto_tag_zero.py + (added), res/pjproject/third_party/ilbc/packing.h (added), + res/pjproject/pjlib/src/pj/pool_caching.c (added), + res/pjproject/pjnath/include/pjnath/errno.h (added), + res/pjproject/pjmedia/include/pjmedia-codec/h264_packetizer.h + (added), res/pjproject/pjmedia/include/pjmedia/sdp_neg.h (added), + res/pjproject/third_party/speex/libspeex/lsp_bfin.h (added), + res/pjproject/third_party/portaudio/aclocal.m4 (added), + res/pjproject/third_party/mp3/mp3_port.h (added), + res/pjproject/third_party/BaseClasses/ctlutil.cpp (added), + res/pjproject/pjsip-apps/src/pocketpj/PocketPJDlg.cpp (added), + res/pjproject/tests/pjsua/scripts-recvfrom/240_publish_scenarios.py + (added), res/pjproject/README-RTEMS (added), + res/pjproject/third_party/build/portaudio/output (added), + res/pjproject/pjsip-apps/build/Makefile (added), + res/pjproject/tests/pjsua/scripts-sipp/prack_fork.xml (added), + res/pjproject/pjlib-util/src/pjlib-util-test/stun.c (added), + res/pjproject/pjlib-util/src/pjlib-util/dns_dump.c (added), + res/pjproject/pjmedia/include/pjmedia/circbuf.h (added), + res/pjproject/pjlib/build/os-darwinos.mak (added), + res/pjproject/third_party/srtp/test/rtpw.c (added), + res/pjproject/tests/pjsua/scripts-sipp/uas-answer-180-multiple-fmts.xml + (added), + res/pjproject/third_party/srtp/crypto/include/cryptoalg.h + (added), res/pjproject/third_party/portaudio/bindings/cpp + (added), + res/pjproject/tests/pjsua/scripts-sipp/uas-answer-200-reinvite-without-sdp.xml + (added), res/pjproject/third_party/portaudio/configure.in + (added), res/pjproject/pjmedia/include/pjmedia-codec/g722.h + (added), res/pjproject/pjsip-apps/src/vidgui/pj-pkgconfig.mak + (added), res/pjproject/pjmedia/include/pjmedia-codec/speex.h + (added), res/pjproject/config.guess (added), + res/pjproject/tests/cdash/cfg_site_sample.py (added), + res/pjproject/third_party/portaudio/src/common/pa_skeleton.c + (added), + res/pjproject/pjsip-apps/src/symbian_ua_gui/inc/symbian_ua_guiSettingItemList.hrh + (added), res/pjproject/third_party/srtp/test/getopt_s.c (added), + res/pjproject/pjmedia/src/pjmedia-codec/g722 (added), + res/pjproject/tests/pjsua/scripts-pesq/201_codec_g722.py (added), + res/pjproject/pjnath/src/pjturn-client/client_main.c (added), + res/pjproject/third_party/gsm/src/short_term.c (added), + res/pjproject/build.symbian/libg7221codec.mmp (added), + res/pjproject/pjmedia/src/pjmedia/wsola.c (added), + res/pjproject/pjlib-util/include/pjlib-util/hmac_sha1.h (added), + res/pjproject/pjlib/include/pj++/list.hpp (added), + res/pjproject/third_party/ilbc/anaFilter.c (added), + res/pjproject/third_party/mp3 (added), + res/pjproject/pjmedia/src/pjmedia/tonegen.c (added), + res/pjproject/pjsip-apps/src/samples/stateful_proxy.c (added), + res/pjproject/third_party/ilbc/anaFilter.h (added), + res/pjproject/pjsip-apps/src/symsndtest/app_main.cpp (added), + res/pjproject/pjsip-apps/src/pocketpj/SettingsDlg.cpp (added), + res/pjproject/tests/pjsua/scripts-sipp/uas-invite.xml (added), + res/pjproject/third_party/g7221/encode/sam2coef.c (added), + res/pjproject/pjlib/src/pj/compat/string.c (added), + res/pjproject/pjlib/include/pj/compat/cc_gcce.h (added), + res/pjproject/pjlib/include/pj/config_site_sample.h (added), + res/pjproject/third_party/build/srtp/output (added), + res/pjproject/tests/pjsua/scripts-pesq/200_codec_speex_8000.py + 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res/pjproject/pjnath/src/pjnath-test/test.c (added), + res/pjproject/pjsip-apps/src/samples/siprtp_report.c (added), + res/pjproject/pjnath/src/pjnath-test/test.h (added), + res/pjproject/third_party/srtp (added), + res/pjproject/third_party/build/g7221/libg7221codec.vcproj + (added), + res/pjproject/pjlib-util/build/wince-evc4/pjlib_util_test_wince.vcp + (added), res/pjproject/build/m-i386.mak (added), + res/pjproject/pjlib-util/include/pjlib-util/srv_resolver.h + (added), res/pjproject/tests/pjsua/scripts-call/150_srtp_3_2.py + (added), res/pjproject/pjsip/include/pjsip_simple.h (added), + res/pjproject/pjmedia/src/test/audio_tool.c (added), + res/pjproject/pjlib/src/pj/exception_symbian.cpp (added), + res/pjproject/pjmedia/build/m-i386.mak (added), + res/pjproject/third_party/BaseClasses/wxutil.h (added), + res/pjproject/pjsip-apps/src/vidgui (added), + res/pjproject/pjsip/src/pjsua-lib/pjsua_media.c (added), + res/pjproject/pjlib-util/build/pjlib_util.vcproj (added), + res/pjproject/pjnath/include/pjnath/stun_transaction.h (added), + res/pjproject/third_party/portaudio/src/hostapi/oss/recplay.c + (added), res/pjproject/third_party/resample/include (added), + res/pjproject/pjmedia/include/pjmedia/transport.h (added), + res/pjproject/pjlib-util/src/pjlib-util/srv_resolver.c (added), + res/pjproject/build.symbian/pjsua_libU.def (added), + res/pjproject/pjsip/src/pjsip-simple/presence_body.c (added), + res/pjproject/pjmedia/include/pjmedia/stereo.h (added), + res/pjproject/tests/pjsua/scripts-call/301_ice_public_b.py + (added), res/pjproject/tests/automated/configure.py (added), + res/pjproject/pjsip-apps/src/symbian_ua/symbian_ua_reg.rss + (added), res/pjproject/pjsip-apps/src/pocketpj/PopUpWnd.h + (added), + res/pjproject/third_party/speex/libspeex/high_lsp_tables.c + (added), res/pjproject/pjlib/src/pj/ssl_sock_ossl.c (added), + res/pjproject/tests/pjsua/scripts-sendto/313_srtp1_unsupported_crypto.py + (added), res/pjproject/pjsip-apps/src/symbian_ua_gui (added), + res/pjproject/third_party/build/milenage/libmilenage.vcp (added), + res/pjproject/pjmedia/include/pjmedia/transport_loop.h (added), + res/pjproject/third_party/build/gsm/libgsmcodec.vcp (added), + res/pjproject/third_party/speex/libspeex/window.c (added), + res/pjproject/tests/pjsua/scripts-sendto/125_sdp_with_multi_audio_2.py + (added), res/pjproject/pjsip-apps/src/symbian_ua_gui/data + (added), res/pjproject/pjsip/src/pjsip/sip_transport_wrap.cpp + (added), res/pjproject/pjmedia/include/pjmedia-videodev/errno.h + (added), res/pjproject/pjlib/src/pj/os_time_common.c (added), + res/pjproject/third_party/resample/src (added), + res/pjproject/pjlib/docs (added), + res/pjproject/tests/pjsua/scripts-sendto/161_err_replaces_dlg_not_found.py + (added), res/pjproject/pjsip-apps/src/pocketpj (added), + res/pjproject/pjsip-apps/src/samples/simple_pjsua.c (added), + res/pjproject/pjsip-apps/src/symbian_ua_gui/src (added), + res/pjproject/tests/pjsua/scripts-media-playrec/100_resample_lf_11_48.py + (added), res/pjproject/pjmedia/include/pjmedia/rtcp.h (added), + res/pjproject/tests/pjsua/scripts-sendto/300_srtp_invalid_crypto_tag_non_numeric.py + (added), res/pjproject/tests/pjsua/scripts-pres/100_peertopeer.py + (added), res/pjproject/pjmedia/src/pjmedia/vid_codec_util.c + (added), res/pjproject/third_party/gsm/MACHINES (added), + res/pjproject/tests/pjsua/scripts-sipp/uac-subscribe.xml (added), + res/pjproject/third_party/build/baseclasses (added), + res/pjproject/third_party/srtp/include/srtp.h (added), + res/pjproject/tests/pjsua/scripts-sendto/173_timer_offer_refresher_uac.py + (added), res/pjproject/pjmedia/src/pjmedia/stream.c (added), + res/pjproject/tests/pjsua/scripts-recvfrom/209a_reg_handle_423_ok.py + (added), res/pjproject/pjlib/src/pjlib-samples/log.c (added), + res/pjproject/third_party/build/portaudio/src/pa_mac_core_old.c + (added), res/pjproject/pjsip/src/pjsip/sip_transport_tcp.c + (added), + res/pjproject/tests/pjsua/scripts-sendto/150_err_extension.py + (added), + res/pjproject/pjlib-util/src/pjlib-util-test/encryption.c + (added), res/pjproject/lib (added), + res/pjproject/pjmedia/include/pjmedia/codec.h (added), + res/pjproject/pjmedia/src/pjmedia/converter_libswscale.c (added), + res/pjproject/pjlib/src/pj/ip_helper_win32.c (added), + res/pjproject/pjmedia/include/pjmedia-videodev/avi_dev.h (added), + res/pjproject/pjlib-util/src/pjlib-util/scanner_cis_bitwise.c + (added), res/pjproject/third_party/gsm/README (added), + res/pjproject/pjlib-util/src/pjlib-util (added), + res/pjproject/third_party/build/gsm (added), + res/pjproject/pjlib/include/pj/compat/cc_msvc.h (added), + res/pjproject/pjsip-apps/src/pjsua_wince (added), + res/pjproject/tests/pjsua (added), + res/pjproject/pjlib/include/pj++/timer.hpp (added), + res/pjproject/build.symbian/pjlib.mmp (added), + res/pjproject/pjsip/src/test/test.c (added), + res/pjproject/third_party/portaudio/build (added), + res/pjproject/pjsip/src/test/test.h (added), + res/pjproject/pjsip/include/pjsip_auth.h (added), + res/pjproject/pjlib/src/pj/errno.c (added), + res/pjproject/third_party/BaseClasses/wxdebug.cpp (added), + res/pjproject/pjsip/include/pjsip-simple/rpid.h (added), + res/pjproject/pjlib/include/pj/compat/os_sunos.h (added), + res/pjproject/third_party/portaudio/install-sh (added), + res/pjproject/pjlib/src/pj/os_info.c (added), + res/pjproject/tests/pjsua/scripts-sipp/uas-reinv-no-media.xml + (added), + res/pjproject/tests/pjsua/scripts-sendto/172_timer_supported_but_not_used.py + (added), + res/pjproject/third_party/build/resample/libresample_dll.vcproj + (added), res/pjproject/pjmedia/include (added), + res/pjproject/third_party/portaudio/src/hostapi/asio/ASIO-README.txt + (added), res/pjproject/pjsip-apps/src/python/samples/presence.py + (added), + res/pjproject/build/vs/pjproject-vs8-debug-static-defaults.vsprops + (added), res/pjproject/pjmedia/src/pjmedia/transport_srtp.c + (added), + res/pjproject/pjmedia/include/pjmedia-codec/amr_sdp_match.h + (added), res/pjproject/pjsip/src/pjsip-simple/rpid.c (added), + res/pjproject/pjlib-util/src/pjlib-util/dns_server.c (added), + res/pjproject/tests/pjsua/runall.py (added), + res/pjproject/pjlib/include/pj/compat/m_armv4.h (added), + res/pjproject/pjsip/src/pjsip/sip_util_proxy.c (added), + res/pjproject/pjlib-util/include/pjlib-util/crc32.h (added), + res/pjproject/pjlib-util/build/os-auto.mak.in (added), + res/pjproject/tests/pjsua/scripts-sipp/uas-subscribe-multipart-notify.xml + (added), res/pjproject/pjlib/build/wince-evc4/pjlib_wince.vcp + (added), res/pjproject/pjmedia/include/pjmedia/sound.h (added), + res/pjproject/pjsip/build/output (added), res/pjproject/pjnath + (added), res/pjproject/INSTALL.txt (added), + res/pjproject/tests/pjsua/mod_call.py (added), + res/pjproject/pjlib/build/wince-evc4/pjlib_wince.vcw (added), + res/pjproject/pjsip/src/test/dlg_core_test.c (added), + res/pjproject/tests/pjsua/scripts-pesq/200_codec_g711u.py + (added), + res/pjproject/tests/pjsua/scripts-sendto/411_fmtp_amrnb_offer_band_eff.py + (added), res/pjproject/third_party/build/resample/config.h + (added), res/pjproject/pjsip-apps/src/pjsua_wince/pjsua_wince.rc + (added), res/pjproject/pjlib/build/output (added), + res/pjproject/pjlib/include/pj/compat/m_powerpc.h (added), + res/pjproject/pjsip/src/test/msg_logger.c (added), + res/pjproject/pjsip-apps/src/pjsua_wince/resource.h (added), + res/pjproject/pjsip/src/pjsip/sip_auth_parser_wrap.cpp (added), + res/pjproject/aconfigure.ac (added), + res/pjproject/tests/pjsua/scripts-sendto/140_sdp_with_direction_attr_in_session_1.py + (added), + res/pjproject/pjsip-apps/src/pjsystest/pjsystest_wince.rc2 + (added), res/pjproject/pjlib/include/pj/compat/os_win32.h + (added), res/pjproject/pjmedia/include/pjmedia/doxygen.h (added), + res/pjproject/pjsip/src/test/main_rtems.c (added), + res/pjproject/pjlib-util/include/pjlib-util/scanner_cis_bitwise.h + (added), res/pjproject/pjsip-apps/src/ipjsystest/main.m (added), + res/pjproject/build.symbian/pjsip.mmp (added), + res/pjproject/third_party/speex/include/speex/speex_jitter.h + (added), res/pjproject/tests/pjsua/run.py (added), + res/pjproject/third_party/speex/symbian (added), + res/pjproject/tests/pjsua/scripts-pesq/200_codec_l16_8000_stereo.py + (added), res/pjproject/pjsip-apps/src/samples/auddemo.c (added), + res/pjproject/tests/pjsua/scripts-sendto/300_srtp_crypto_case_insensitive.py + (added), res/pjproject/third_party/g7221/common/basic_op.c + (added), + res/pjproject/pjnath/build/wince-evc4/pjnath_test_wince.vcp + (added), res/pjproject/third_party/g7221/common/basic_op.h + (added), res/pjproject/third_party/portaudio/config.guess + (added), res/pjproject/third_party/portaudio/src/os/unix (added), + res/pjproject/third_party/speex/libspeex/cb_search_sse.h (added), + res/pjproject/tests/pjsua/tools/Makefile (added), + res/pjproject/pjlib/src/pj/compat/longjmp_i386.S (added), + res/pjproject/third_party/portaudio/pablio (added), + res/pjproject/build.symbian/symbian_ua_udeb.pkg (added), + res/pjproject/README.txt (added), + res/pjproject/third_party/srtp/srtp.vcproj (added), + res/pjproject/pjnath/build (added), + res/pjproject/third_party/portaudio/src/hostapi/dsound (added), + res/pjproject/tests/automated/prepare.xml.template (added), + res/pjproject/pjsip/src/pjsua-lib/pjsua_pres.c (added), + res/pjproject/tests/pjsua/scripts-sipp/uas-reinv-and-ack(same-branch)-without-sdp.xml + (added), res/pjproject/pjlib/build (added), + res/pjproject/third_party/build/baseclasses/libbaseclasses.vcproj + (added), + res/pjproject/third_party/speex/include/speex/speex_preprocess.h + (added), res/pjproject/pjlib/src/pjlib-test (added), + res/pjproject/pjsip-apps/src/symbian_ua_gui/data/symbian_ua_gui.l01 + (added), res/pjproject/pjlib/build/privkey.pem (added), + res/pjproject/pjmedia/src/pjmedia/alaw_ulaw_table.c (added), + res/pjproject/configure-legacy (added), + res/pjproject/tests/pjsua/scripts-sendto/200_ice_success_1.py + (added), res/pjproject/pjsip/include/pjsip/sip_transport.h + (added), res/pjproject/pjnath/src/pjturn-srv/server.c (added), + res/pjproject/pjmedia/build/os-linux.mak (added), + res/pjproject/pjlib/include/pj/compat/os_win32_wince.h (added), + res/pjproject/pjsip/src/pjsip-ua/sip_replaces.c (added), + res/pjproject/third_party/portaudio/src/common/pa_util.h (added), + res/pjproject/pjsip-apps/src/symbian_ua_gui/inc/symbian_ua_guiDocument.h + (added), res/pjproject/pjlib/src/pj/fifobuf.c (added), + res/pjproject/third_party/gsm/tls/sour1.dta (added), + res/pjproject/pjsip/include/pjsip/sip_types.h (added), + res/pjproject/pjlib/include/pj/compat/time.h (added), + res/pjproject/pjsip/src/pjsip/sip_auth_msg.c (added), + res/pjproject/tests/pjsua/scripts-sendto/001_torture_4475_3_1_1_1.py + (added), res/pjproject/pjsip/include/pjsip_ua.h (added), + res/pjproject/pjlib/build/Makefile (added), + res/pjproject/third_party/srtp/README (added), + res/pjproject/tests/pjsua/scripts-sendto/311_srtp1_recv_avp.py + (added), res/pjproject/pjsip-apps/src/pjsua/main_rtems.c (added), + res/pjproject/pjsip-apps/src/pocketpj/res/invisibl.bmp (added), + res/pjproject/pjlib/src/pjlib-test/rtems_network_config.h + (added), res/pjproject/third_party/srtp/crypto/math/stat.c + (added), res/pjproject/third_party/srtp/test/replay_driver.c + (added), res/pjproject/pjmedia/src/pjmedia-audiodev/audiotest.c + (added), res/pjproject/pjlib/src/pjlib++-test (added), + res/pjproject/pjsip-apps/src/samples/streamutil.c (added), + res/pjproject/pjmedia/src/pjmedia/ffmpeg_util.c (added), + res/pjproject/tests/pjsua/scripts-sendto/500_pres_subscribe_with_bad_event.py + (added), res/pjproject/third_party/srtp/install-sh (added), + res/pjproject/tests/pjsua/scripts-pesq/200_codec_speex_16000.py + (added), + res/pjproject/third_party/srtp/crypto/cipher/null_cipher.c + (added), res/pjproject/pjmedia/src/pjmedia/ffmpeg_util.h (added), + res/pjproject/pjlib-util/src (added), + res/pjproject/pjsip/include/pjsip/sip_config.h (added), + res/pjproject/pjlib/docs/doxygen.cfg (added): Add support for + ICE/STUN/TURN in res_rtp_asterisk and chan_sip. Review: + https://reviewboard.asterisk.org/r/1891/ + +2012-06-29 20:32 +0000 [r369512] Mark Michelson + + * main/rtp_engine.c, /: Fix apparent copy and paste error where + incorrect "glue" is used. ........ Merged revisions 369511 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-29 17:02 +0000 [r369493] Richard Mudgett + + * apps/app_dial.c, main/channel.c, main/autoservice.c, main/pbx.c, + channels/chan_local.c, funcs/func_channel.c, + main/channel_internal_api.c, main/features.c, + configs/cdr.conf.sample, include/asterisk/channel.h, + include/asterisk/pbx.h, CHANGES, apps/app_followme.c, + apps/app_queue.c: Hangup handlers - Dialplan subroutines that run + when the channel hangs up. Hangup handlers are an alternative to + the h extension. They can be used in addition to the h extension. + The idea is to attach a Gosub routine to a channel that will + execute when the call hangs up. Whereas which h extension gets + executed depends on the location of dialplan execution when the + call hangs up, hangup handlers are attached to the call channel. + You can attach multiple handlers that will execute in the order + of most recently added first. (closes issue ASTERISK-19549) + Reported by: Mark Murawski Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/2002/ + +2012-06-29 16:56 +0000 [r369492] Joshua Colp + + * /, channels/chan_sip.c: With some configurations a transport is + not actually specified so assume UDP in these cases. ........ + Merged revisions 369490 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369491 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-29 16:42 +0000 [r369489] Richard Mudgett + + * main/channel_internal_api.c, .cleancount: Remove obsolete struct + ast_channel note. The opaquing the ast_channel struct no longer + requires .cleancount to be changed when the struct is changed. * + Bump .cleancount value one last time because of struct + ast_channel for old times sake. + +2012-06-29 15:33 +0000 [r369473] Joshua Colp + + * /, channels/chan_sip.c: Make the address family filter specific + to the transport. (closes issue ASTERISK-16618) Reported by: Leif + Madsen Review: https://reviewboard.asterisk.org/r/1667/ ........ + Merged revisions 369471 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369472 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-28 01:12 +0000 [r369449-369454] Terry Wilson + + * include/asterisk/config_options.h, + configs/config_test.conf.sample, main/config_options.c, + tests/test_config.c: Add the ability to set flags via the config + options api Allows the setting of flags via the config options + api. For example, code like this: #define OPT1 1 << 0 #define + OPT2 1 << 1 #define OPT3 1 << 2 struct thing { unsigned int + flags; }; and a config like this: [blah] opt1=yes opt2=no + opt3=yes Review: https://reviewboard.asterisk.org/r/2004/ + + * /, channels/chan_sip.c, channels/sip/include/sip.h: AST-2012-010: + Clean up after a reinvite that never gets a final response The + basic problem is that if a re-INVITE is sent by Asterisk and it + receives a provisional response, but no final response, then the + dialog is never torn down. In addition to leaking memory, this + also leaks file descriptors and will eventually lead to Asterisk + no longer being able to process calls. This patch just keeps + track of whether there is an outstanding re-INVITE, and if there + is goes ahead and cleans up everything as though there was no + outstanding reinvite. Review: + https://reviewboard.asterisk.org/r/2009/ (closes issue + ASTERISK-19992) Reported by: Steve Davies Tested by: Steve + Davies, Terry Wilson ........ Merged revisions 369436 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369437 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-26 21:45 +0000 [r369414] Jonathan Rose + + * include/asterisk/logger.h, channels/chan_dahdi.c, + main/autoservice.c, main/pbx.c, channels/chan_local.c, + channels/sig_analog.c, main/channel_internal_api.c, + channels/chan_agent.c, main/features.c, main/logger.c, + channels/chan_iax2.c, channels/sig_pri.c, channels/sig_ss7.c, + main/bridging.c, main/cli.c: Unique Call ID logging Phases III + and IV Adds call ID logging changes to specific channel drivers + that weren't handled handled in phase II of Call ID Logging. Also + covers logging for threads for threads created by systems that + may be involved with many different calls. Extra special thanks + to Richard for rigorous review of chan_dahdi and its various + signalling modules. review: + https://reviewboard.asterisk.org/r/1927/ review: + https://reviewboard.asterisk.org/r/1950/ + +2012-06-26 13:23 +0000 [r369370-369392] Matthew Jordan + + * /, main/adsi.c: Fix crash in unloading of res_adsi module When + res_adsi is unloaded, it removes the ADSI functions that it + previously installed by passing a NULL adsi_funcs pointer to + ast_adsi_install_funcs. This function was not checking whether or + not the adsi_funcs pointer passed in was NULL before + dereferencing it to check whether or not the version of the + functions matches what the core was expecting it. This patch + makes it so that the version is only checked if a potentially + valid adsi_funcs pointer was passed in. Passing in NULL removes + the installed functions, bypassing the version check. ........ + Merged revisions 369390 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369391 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/manager.c: Update "manager show event" to support tab + completion Thank you rmudgett for pointing out that I was missing + this in the initial check-in for AMI event documentation + (r369346) + + * main/cdr.c, /: Fix incorrect duration reporting in CDRs created + in batch mode Certain places in core/cdr.c would, if the duration + value were 0, calculate the duration as being the delta between + the current time and the time at which the CDR record was + started. While this does not typically cause a problem in + non-batch mode, this can cause an issue in batch mode where CDR + records are gathered and written long after those calls have + ended. In particular, this affects calls that were never + answered, as those are expected to have a duration of 0. Often, + this would result in CDR logs with a significant number of calls + with lengthy durations, but dispositions of "BUSY". Note that + this does not affect cdr_csv, as that backend does not use + ast_cdr_getvar and instead directly reports the duration value. + The affected core backends include cdr_apative_odbc and + cdr_custom; other extended or deprecated CDR backends may + potentially still directly manipulate the duration values. (issue + ASTERISK-19860) Reported by: Thomas Arimont (issue AST-883) + Reported by: Thomas Arimont Tested by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/1996/ ........ Merged + revisions 369351 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369369 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-25 19:26 +0000 [r369367] Mark Michelson + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Re-fix how + local tag is generated when sending a 481 to an INVITE. Match our + local tag to whatever to-tag was sent in the initial INVITE. + Because the size of the to-tag may not fit in the buffer in the + sip_pvt, it has been changed to a string field. (closes issue + ASTERISK-19892) reported by Walter Doekes Review: + https://reviewboard.asterisk.org/r/1977 ........ Merged revisions + 369352 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 369353 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-25 17:59 +0000 [r369346] Matthew Jordan + + * apps/app_dial.c, apps/app_meetme.c, configure.ac, + apps/app_userevent.c, CHANGES, apps/app_queue.c, Makefile, + build_tools/get_documentation.py (added), main/manager.c, + configure, build_tools/post_process_documentation.py (added), + include/asterisk/xmldoc.h, apps/app_confbridge.c, makeopts.in, + apps/app_stack.c, apps/app_chanspy.c, doc/appdocsxml.dtd, + main/xmldoc.c, apps/app_voicemail.c: Add AMI event documentation + This patch adds the core changes necessary to support AMI event + documentation in the source files of Asterisk, and adds + documentation to those AMI events defined in the core application + modules. Event documentation is built from the source by two new + python scripts, located in build_tools: get_documentation.py and + post_process_documentation.py. The get_documentation.py script + mirrors the actions of the existing AWK get_documentation + scripts, except that it will scan the entirety of a source file + for Asterisk documentation. Upon encountering it, if the + documentation happens to be an AMI event, it will attempt to + extract information about the event directly from the manager + event macro calls that raise the event. The + post_process_documentation.py script combines manager event + instances that are the same event but documented in multiple + source files. It generates the final core-[lang].xml file. As + this process can take longer to complete than a typical 'make + all', it is only performed if a new make target, 'full', is + chosen. Review: https://reviewboard.asterisk.org/r/1967/ + +2012-06-25 16:07 +0000 [r369329] Richard Mudgett + + * /, main/features.c: Fix Bridge application occasionally returning + to the wrong location. * Fix do_bridge_masquerade() getting the + resume location from the zombie channel. The code must not touch + a clone channel after it has masqueraded it. The clone channel + has become a zombie and is starting to hangup. (closes issue + ASTERISK-19985) Reported by: jamicque Patches: + jira_asterisk_19985_v1.8.patch (license #5621) patch uploaded by + rmudgett Tested by: jamicque ........ Merged revisions 369327 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 369328 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-25 15:55 +0000 [r369304-369326] Mark Michelson + + * include/asterisk/adsi.h, /, main/Makefile, res/res_adsi.c, + main/adsi.c (added), res/res_adsi.exports.in (removed): Multiple + revisions 369323-369324 ........ r369323 | mmichelson | + 2012-06-25 10:35:43 -0500 (Mon, 25 Jun 2012) | 9 lines Eliminate + embedding of res_adsi.so module. The way this is done is to stop + using the optional API. Instead, res_adsi.so, when loaded fills + in a table of function pointers. Review: + https://reviewboard.asterisk.org/r/1991 ........ r369324 | + mmichelson | 2012-06-25 10:50:17 -0500 (Mon, 25 Jun 2012) | 2 + lines Forgot to svn add this file in my last commit. ........ + Merged revisions 369323-369324 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369325 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Be more consistent with the return code + for requests received from invalid domain. When Asterisk receives + an INVITE from an external domain when allowexternaldomains=no + send a 403 instead of a 404. This is consistent with Asterisk's + behavior when receiving a REGISTER in this situation. (Closes + issue ASTERISK-19601) Reported by Matthew Jordan Patches: + ASTERISK-19601-no401.patch uploaded by Mark Michelson (License + #5049) ........ Merged revisions 369302 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369303 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-23 00:33 +0000 [r369237-369296] Richard Mudgett + + * main/features.c: Fix F and F(x) action logic in Bridge + application. + + * /, main/features.c: Fix Bridge application and AMI Bridge action + error handling. * Fix AMI Bridge action disconnecting the AMI + link on error. * Fix AMI Bridge action and Bridge application not + checking if their masquerades were successful. * Fix Bridge + application running the h-exten when it should not. * Made + do_bridge_masquerade() return if the masquerade was successful so + the Bridge application and AMI Bridge action could deal with it + correctly. * Made bridge_call_thread_launch() hangup the passed + in channels if the bridge_call_thread fails to start. Those + channels would have been orphaned. * Made builtin_atxfer() check + the success of the transfer masquerade setup. ........ Merged + revisions 369282 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369283 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_queue.c: Explicitly check caller hangup in app Queue + rather than a polluted res2 value. ........ Merged revisions + 369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 369263 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_queue.c: Fix F and F(x) action logic in Queue + application. + + * apps/app_dial.c, /: Check if PBX was started and fix F and F(x) + action logic in Dial application. ........ Merged revisions + 369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 369259 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/ccss.c: Check if PBX was started for generic CCSS recall. + ........ Merged revisions 369238 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369239 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Change incorrect chan_sip zombie hangup + debug message. They are all zombies now. ........ Merged + revisions 369235 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369236 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-22 20:05 +0000 [r369217] Terry Wilson + + * /, channels/chan_sip.c: Don't crash on a guest directmedia call A + sip_pvt may not have relatedpeer set if a call doesn't match up + with a peer. If there is no relatedpeer, there is no direct media + ACL to apply, so just return that it is allowed. (closes issue + ASTERISK-20040) Reported by: Terry Wilson ........ Merged + revisions 369214 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369215 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-22 19:54 +0000 [r369184-369216] Kinsey Moore + + * channels/chan_dahdi.c: Fix wrong variable name in the R2 + disconnect callback + + * /, channels/chan_sip.c: Don't parse media stream state for SIP + video streams The sendonly/recvonly/sendrecv/inactive media + stream attributes were parsed for video, but nothing was ever + done with them. With this code removed, an UNSUPPORTED message is + produced when these attributes are used in conjunction with a + video stream which is the better behavior since they were never + really supported in the first place. ........ Merged revisions + 369195 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 369206 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_dahdi.c: Add HANGUPCAUSE hash implementation for + DAHDI MFC/R2 subtech This adds a minimal implementation of the + "Who Hung Up?" Asterisk 11 work to chan_dahdi.c for the MFC/R2 + DAHDI subtech. Given the way that OpenR2 interfaces with + chan_dahdi, it is much harder to expose the type of protocol + information that is available in PRI, SS7, or other channel + technologies. + + * channels/sig_analog.c, channels/sig_pri.c: Add HANGUPCAUSE hash + support for analog and PRI DAHDI subtechs This is part of the + DAHDI support for the Asterisk 11 "Who Hung Up?" project and + covers the implementation for the technologies implemented in + sig_analog.c and sig_pri.c. Tested on a local machine to verify + protocol and cause information is available. Review: + https://reviewboard.asterisk.org/r/1953/ (issue SWP-4222) + + * channels/sig_ss7.c: Add "Who Hung Up?" implementation for DAHDI + SS7 subtechnology Testing was done on a local machine to verify + that protocol and cause information was being sent properly. + Review: https://reviewboard.asterisk.org/r/1955/ (issue SWP-4222) + +2012-06-20 21:33 +0000 [r369166-369167] Richard Mudgett + + * main/logger.c: Don't waste time initializing the whole + call_identifer_str[]. The array is either setup with a callid + string or only the first element needs to be initialized. + + * channels/chan_misdn.c: Fix chan_misdn compile error. + +2012-06-20 17:48 +0000 [r369148] Alexandr Anikin + + * /, addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: fix + locking issue on empty callList (issue ASTERISK-19298) Reported + by: Dmitry Melekhov Patches: ASTERISK-18322-2.patch ........ + Merged revisions 369146 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369147 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-20 11:47 +0000 [r369142] Sean Bright + + * apps/app_externalivr.c: Remove declaration of eivr_connect_socket + because it no longer exists. + +2012-06-20 11:20 +0000 [r369141] Alexandr Anikin + + * addons/chan_ooh323.c: use right definition for channel name + +2012-06-20 03:18 +0000 [r369110-369126] Michael L. Young + + * main/manager.c, CHANGES: Add IPv6 Support To Manager This patch + adds IPv6 support to AMI. (Closes issue ASTERISK-19965) Reported + by: Michael L. Young Tested by: Michael L. Young Patches: + ami_ipv6_v3.diff uploaded by Michael L. Young (license 5026) + Review: https://reviewboard.asterisk.org/r/1968/ + + * main/netsock2.c, /, include/asterisk/netsock2.h: Fix NULL pointer + segfault in ast_sockaddr_parse() While working with + ast_parse_arg() to perform a validity check, a segfault occurred. + The segfault occurred due to passing a NULL pointer to + ast_sockaddr_parse() from ast_parse_arg(). According to the + documentation in config.h, "result pointer to the result. NULL is + valid here, and can be used to perform only the validity checks." + This patch fixes the segfault by checking for a NULL pointer. + This patch also adds documentation to netsock2.h about why it is + necessary to check for a NULL pointer. (Closes issue + ASTERISK-20006) Reported by: Michael L. Young Tested by: Michael + L. Young Patches: asterisk-20006-netsock-null-ptr.diff uploaded + by Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/1990/ ........ Merged + revisions 369108 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369109 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-19 23:36 +0000 [r369092] Alexandr Anikin + + * addons/chan_ooh323.c, /: check rtptimeouts in ooh323 channels as + per config file (rtp voice, video, udptl except rtcp) (closes + issue ASTERISK-19179) Reported by: TSAREGORODTSEV Yury Patches: + 19179-ooh323-ast10.patch ........ Merged revisions 369091 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-19 21:13 +0000 [r369086] Kinsey Moore + + * main/channel.c, channels/chan_dahdi.c, channels/chan_misdn.c, + main/rtp_engine.c, include/asterisk/channel.h, + channels/chan_iax2.c: Ensure that pvt cause information does not + break native bridging Channel drivers that allow native bridging + need to handle AST_CONTROL_PVT_CAUSE_CODE frames and previously + did not handle them properly, usually breaking out of the native + bridge. This change corrects that behavior and exposes the + available cause code information to the dialplan while native + bridges are in place. This required exposing the HANGUPCAUSE hash + setter outside of channel.c, so additional documentation has been + added. + +2012-06-19 15:44 +0000 [r369068] Mark Michelson + + * /, channels/chan_sip.c: Fix request routing issue when + outboundproxy is used. Asterisk was incorrectly setting the + destination of CANCELs and ACKs for error responses to the URI of + the initial INVITE. This resulted in further requests, such as + INVITEs with authentication credentials, to be routed + incorrectly. Instead, when these CANCEL or ACKs are to be sent, + we should simply keep the destination the same as what it + previously was. There is no need to alter it any. (closes issue + ASTERISK-20008) Reported by Marcus Hunger Patches: + ASTERISK-20008.patch uploaded by Mark Michelson (license #5049) + ........ Merged revisions 369066 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369067 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-18 22:56 +0000 [r369061] Kinsey Moore + + * main/features.c: Fix AST_CONTROL_PVT_CAUSE_CODE handling When the + IAX2 Who Hung Up? changes were added, they uncovered a bug in the + way AST_CONTROL_PVT_CAUSE_CODE was handled in + feature_request_and_dial(). This particular frame subtype was + being treated like more terminal control frames causing the + function to be exited prematurely. + +2012-06-18 18:25 +0000 [r369057] Richard Mudgett + + * /, main/features.c: Fix monitoring calls put in a parking lot. * + Fix a regression that was introduced by -r366167 which + effectively disabled monitoring parked calls. (closes issue + ASTERISK-20012) Reported by: sdolloff Tested by: rmudgett + ........ Merged revisions 369043 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 369044 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-15 21:18 +0000 [r369034] Damien Wedhorn + + * channels/chan_skinny.c: Various small chan_skinny fixes and + cleanup Added test to skinny_register to only allow device to + register against a device that is not already registered. Addback + l->device test for skinny_show_lines. Fixes segfault if a line is + configured but not configured to a device. Reverses part of + r368680. Removed redundant l->device tests in subsubstate and + dumpsub. l->device will always be valid if these routines are + called. Reverses 368948 - discussed with mjordan on irc. Some + indentation cleanup. + +2012-06-15 17:13 +0000 [r369028] Kinsey Moore + + * channels/chan_sip.c, channels/sip/include/sip.h: Allow chan_sip + to decline unwanted media streams This change replaces the static + array of four representable media streams with an AST_LIST so + that chan_sip can keep track of offered media streams. This + allows chan_sip to deal with offers containing multiple same-type + streams and many other situations without rejecting the SDP offer + in its entirety, yet still generating a valid response. This also + covers cases where Asterisk can not comprehend the offer if it is + in the correct format. Previously, chan_sip would reject SDP + offers or entirely ignore individual stream offers in an effort + to be more compatible which would often result in invalid SDP + responses. Review: https://reviewboard.asterisk.org/r/1988/ + +2012-06-15 16:30 +0000 [r369027] Jason Parker + + * /, apps/app_voicemail.c: Fix voicemail API tests by using the + correct argument order for create/destroy. ........ Merged + revisions 369024 from + http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 + ........ Merged revisions 369026 from + http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones + +2012-06-15 16:20 +0000 [r369013] Kevin P. Fleming + + * main/format.c, main/udptl.c, main/netsock2.c, main/autoservice.c, + main/rtp_engine.c, main/frame.c, main/security_events.c, /, + main/say.c, main/threadstorage.c, channels/console_video.c, + main/devicestate.c, main/astfd.c, main/taskprocessor.c, + main/format_pref.c, main/astobj2.c, main/indications.c, + main/config.c, main/loader.c, main/term.c, + apps/confbridge/conf_config_parser.c, main/cli.c, + channels/sig_analog.c, main/framehook.c, main/strcompat.c, + main/plc.c, main/fskmodem_int.c, main/syslog.c, + main/stdtime/localtime.c, main/bridging.c, main/db.c, + channels/sig_ss7.c, main/datastore.c, main/sched.c, + channels/sip/sdp_crypto.c, main/strings.c, main/pbx.c, + channels/vcodecs.c, channels/sip/security_events.c, + main/libasteriskssl.c, channels/iax2-provision.c, + pbx/dundi-parser.c, main/aoc.c, main/cel.c, utils/astdb2bdb.c, + channels/iax2-parser.c, main/chanvars.c, main/netsock.c, + build_tools/find_missing_support_level (added), main/data.c, + main/srv.c, channels/chan_misdn.c, main/privacy.c, + main/fixedjitterbuf.c, channels/sip/dialplan_functions.c, + main/test.c, main/audiohook.c, codecs/codec_dahdi.c, main/alaw.c, + main/asterisk.c, main/timing.c, main/global_datastores.c, + main/fskmodem_float.c, main/ccss.c, + channels/sip/reqresp_parser.c, main/xml.c, + channels/misdn/isdn_msg_parser.c, main/utils.c, main/autochan.c, + channels/misdn/isdn_lib.c, main/enum.c, main/presencestate.c, + main/fskmodem.c, channels/misdn_config.c, main/io.c, + main/channel.c, main/cdr.c, res/ael/pval.c, main/ulaw.c, + main/dial.c, main/format_cap.c, main/tdd.c, + channels/console_gui.c, main/heap.c, channels/misdn/ie.c, + main/logger.c, main/app.c, channels/console_board.c, + main/image.c, main/message.c, main/dns.c, main/lock.c, + main/stun.c, channels/sip/srtp.c, main/dnsmgr.c, + main/slinfactory.c, main/channel_internal_api.c, + main/translate.c, main/jitterbuf.c, main/acl.c, + utils/astdb2sqlite3.c, channels/sip/utils.c, channels/sig_pri.c, + apps/app_system.c, funcs/func_realtime.c, main/tcptls.c, + main/hashtab.c, funcs/func_presencestate.c, + apps/app_celgenuserevent.c, main/abstract_jb.c, main/callerid.c, + main/file.c, main/config_options.c, res/snmp/agent.c, + main/astmm.c, main/event.c, channels/misdn/portinfo.c, + channels/sip/config_parser.c, channels/vgrabbers.c, main/dsp.c, + main/xmldoc.c: Multiple revisions 369001-369002 ........ r369001 + | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 + lines Add support-level indications to many more source files. + Since we now have tools that scan through the source tree looking + for files with specific support levels, we need to ensure that + every file that is a component of a 'core' or 'extended' module + (or the main Asterisk binary) is explicitly marked with its + support level. This patch adds support-level indications to many + more source files in tree, but avoids adding them to third-party + libraries that are included in the tree and to source files that + don't end up involved in Asterisk itself. ........ r369002 | + kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 + lines Add a script to enable finding source files without + support-levels defined. ........ Merged revisions 369001-369002 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 369005 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-15 16:17 +0000 [r369007] Kinsey Moore + + * main/frame.c, channels/chan_iax2.c, include/asterisk/frame.h: Add + HANGUPCAUSE hash support to IAX2 Continuing with the Who Hung Up? + project for Asterisk 11, this adds support to IAX2 for the + HANGUPCAUSE hash. Additionally, this breaks out some + functionality in frame.c for getting information about frame + types and subclasses. Review: + https://reviewboard.asterisk.org/r/1941/ (issue SWP-4222) + +2012-06-15 15:33 +0000 [r369000] Jason Parker + + * /, apps/app_voicemail.exports.in: Remove some symbol exports that + got missed in the removal of global symbols. (issue AST-807) + (issue AST-901) (issue AST-908) ........ Merged revisions 368998 + from + http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 + ........ Merged revisions 368999 from + http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones + +2012-06-15 00:55 +0000 [r368972-368991] Richard Mudgett + + * /: Remove remaining properties mmichelson left laying around from + phones branch merge. + + * apps/app_dial.c, main/channel.c, include/asterisk/app.h, + main/ccss.c, main/app.c, apps/app_followme.c, apps/app_queue.c, + apps/app_stack.c: Allow non-normal execution routines to be able + to run on hungup channels. * Make non-normal dialplan execution + routines be able to run on a hung up channel. This is preparation + work for hangup handler routines. * Fixed ability to support + relative non-normal dialplan execution routines. (i.e., The + context and exten are optional for the specified dialplan + location.) Predial routines are the only non-normal routines that + it makes sense to optionally omit the context and exten. Setting + a hangup handler also needs this ability. * Fix Return + application being able to restore a dialplan location exactly. + Channels without a PBX may not have context or exten set. * Fixes + non-normal execution routines like connected line interception + and predial leaving the dialplan execution stack unbalanced. + Errors like missing Return statements, popping too many stack + frames using StackPop, or an application returning non-zero could + leave the dialplan stack unbalanced. * Fixed the AGI gosub + application so it cleans up the dialplan execution stack and + handles the autoloop priority increments correctly. * Eliminated + the need for the gosub_virtual_context return location. Review: + https://reviewboard.asterisk.org/r/1984/ + + * main/pbx.c: Make the Hangup application set a softhangup flag. + The Hangup application used to just return -1 to cause normal + dialplan execution to hangup a channel. For the non-normal + execution routines like predial and connected-line interception + routines, the hangup request would exit the routine early but + otherwise be ignored. * Made the Hangup application not allow + setting a cause code of zero. A zero cause code is not defined. + + * include/asterisk/app.h: Move vm defines to group them better. + +2012-06-14 19:40 +0000 [r368966] Jason Parker + + * include/asterisk/app.h, /, tests/test_voicemail_api.c, + main/app.c, include/asterisk/app_voicemail.h (removed), + apps/app_voicemail.c: Multiple revisions 368963,368965 ........ + r368963 | qwell | 2012-06-14 13:47:03 -0500 (Thu, 14 Jun 2012) | + 14 lines Remove global symbol requirement from app_voicemail. + This uses the existing "function installation" stuff that already + existed for other functions, like getting message counts. (closes + issue AST-807) (issue AST-901) (issue AST-908) Review: + https://reviewboard.asterisk.org/r/1965/ ........ Merged + revisions 368962 from + http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 + ........ r368965 | qwell | 2012-06-14 14:04:57 -0500 (Thu, 14 Jun + 2012) | 11 lines These functions that were moved need to be + static. Also wrap test functions in a #ifdef. (issue AST-807) + (issue AST-901) (issue AST-908) ........ Merged revisions 368964 + from + http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 + ........ Merged revisions 368963,368965 from + http://svn.asterisk.org/svn/asterisk/branches/10-digiumphones + +2012-06-14 17:34 +0000 [r368948] Matthew Jordan + + * /, channels/chan_skinny.c: AST-2012-009: Fix crash in chan_skinny + due to Key Pad Button Message handling AST-2012-008 (r367844) + fixed a denial of service attack exploitable in the Skinny + channel driver that occurred when certain messages are sent after + a previously registered station sends an Off Hook message. + Unresolved in that patch is an issue in the Asterisk 10 releases, + wherein, if a Station Key Pad Button Message is processed after + an Off Hook message, the channel driver will inappropriately + dereference a NULL pointer. This patch fixes those places where + the message handling or the channel callback functions would + attempt to dereference the line's pointer to the device. (issue + ASTERISK-19905) Reported by: Christoph Hebeisen Tested by: + mjordan, Christoph Hebeisen Patches: AST-2012-009-10.diff + uploaded by mjordan (license 6283) ........ Merged revisions + 368947 from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-14 15:28 +0000 [r368929] Mark Michelson + + * /, main/Makefile: Revert Makefile change to remove embedding + res_adsi.so The change has resulted in a linking error for + certain versions of GCC. This is much worse than the original + issue, so for now, temporarily revert the change. A more thorough + change will be sought out. ........ Merged revisions 368927 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368928 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-14 13:41 +0000 [r368920-368921] Terry Wilson + + * include/asterisk/config_options.h, main/config_options.c: Add a + post_apply callback to the Config Options API This adds a + callback that only fires when changes have been successfully + applied via the Config Options API. Review: + https://reviewboard.asterisk.org/r/1980/ + + * include/asterisk/config_options.h, main/config_options.c: Add + filename alias support to the Config Options API This adds the + ability to handle a single filename alias for a config file. This + is useful if a config filename has changed, but the old filename + should be supported for backwards compatibility. Review: + https://reviewboard.asterisk.org/r/1981/ + +2012-06-13 21:17 +0000 [r368900] Mark Michelson + + * /, funcs/func_volume.c: Fix a deadlock that occurs when + func_volume is used on a local channel. This was discovered by + trying to perform a call forward to an extension that makes use + of func_volume. When the local channel is optimized away, the + datastore on the local;2 channel would have its audiohook + destroyed rather than detaching the audiohook from the channel + and then destroying it. With this patch, func_volume's datastore + destructor takes the proper route of detaching the audiohook and + then destroying it. (closes issue ASTERISK-19611) reported by + Volker Sauer Patches: ASTERISK-19611.patch uploaded by Mark + Michelson (license #5049) ........ Merged revisions 368898 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368899 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-13 20:28 +0000 [r368896] Matthew Jordan + + * res/res_smdi.c, /, res/res_adsi.c: Mark res_smdi/res_adsi as + 'core' supported modules Recently, various issues surrounding + weak symbols have caused problems with modules that rely on that + feature to be enabled in menuselect. This includes app_voicemail + and chan_dahdi, as they both rely upon res_smdi and res_adsi, + which, in certain circumstances, may not be enabled by default in + menuselect. Because res_smdi/res_adsi are dependencies for + chan_dahdi/app_voicemail, this patch marks both as 'core' + supported modules. This will allow both app_voicemail and + chan_dahdi to be enabled as well, regardless of whether or not + that system supports weak symbols. (issue AST-900) Reported by: + Thomas Arimont (issue AST-885) Reported by: Denis Alberto + Martinez ........ Merged revisions 368894 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368895 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-13 19:51 +0000 [r368886] Mark Michelson + + * /, main/Makefile: Remove forced linking of res_adsi.o In GCC 4.5+ + the result is that Asterisk has a phantom module loaded at + startup, claiming to be res_adsi. (closes issue ASTERISK-19920) + reported by Leif Madsen ........ Merged revisions 368873 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368885 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-13 14:55 +0000 [r368832-368855] Matthew Jordan + + * Makefile: Replace MODULES_DIR with ASTMODDIR in Makefile's + INSTALLDIRS Post Asterisk 10, the MODULES_DIR variable no longer + exists, and was replaced with ASTMODDIR. + + * Makefile, /: Do not install empty directories; add ASTLIBDIR + r368830 modified the installation script to only create a + directory if that directory does not exist. If some directory + variable was empty, it would attempt to create the empty + location. It also failed to create the ASTLIBDIR directory. This + patch fixes it such that the correct directories are made and + only created if a value specifying them actually exists. ........ + Merged revisions 368852 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368853 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * Makefile, /: Do not perform install on existing directories If a + directory already exists, performing a 'make install' will remove + the permissions associated with the current directory and replace + them with the permissions of the user executing the install. This + patch changes this behavior to only perform an install on the + directory if the directory does not exist. Thus, if a user later + changes the permissions on that directory, those permissions will + be preserved in subsequent installs. Review: + https://reviewboard.asterisk.org/r/1986 Review: + https://reviewboard.asterisk.org/r/1864 (closes issue + ASTERISK-19492) Reported by: Karl Fife Tested by: Paul Belanger, + Tilghman Lesher patches: ASTERISK-19492 by pabelanger (uploaded + by mjordan) ........ Merged revisions 368830 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368831 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-12 15:46 +0000 [r368809] Mark Michelson + + * /, channels/chan_sip.c: Set the Caller ID "tag" on peers even if + remote party information is present. On incoming calls, we were + setting the cid_tag on the dialog only if there was no remote + party information (Remote-Party-ID or P-Asserted-Identity) + present. The Caller ID tag is an invented parameter, though, and + should be set no matter the circumstance. (closes issue + ASTERISK-19859) Reported by Thomas Arimont (closes issue AST-884) + Reported by Trey Blancher ........ Merged revisions 368807 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368808 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-12 14:09 +0000 [r368793-368794] Matthew Jordan + + * /: Update merge property information + + * channels/chan_sip.c: Fix deadlock in SIP transfers that involve a + REFER request In r367163, "send to voicemail" functionality was + added to the SIP channel driver. This required updating the party + redirecting information for the channel based on the headers + provided in the REFER request. When the redirecting party + information is updated on the channel, a call to + ast_indicate_data occurs. Because handle_request_refer still had + the sip_pvt locked, a deadlock could occur between the pbx_thread + and the do_monitor thread servicing the REFER request. This patch + preserves the proper locking order between the channel and the + sip_pvt by ensuring that the sip_pvt is unlocked prior to + updating the party redirecting information on the channel. + (closes issue AST-903) Reported by: Matt Jordan patches: + jira_ast_903_trunk.patch by rmudgett (license 5621) + +2012-06-12 04:03 +0000 [r368784] Kinsey Moore + + * channels/chan_sip.c, UPGRADE.txt: Parse ANI2 information from SIP + From header parameters ANI2 information is now parsed out of SIP + From headers when present in the oli, isup-oli, and ss7-oli + parameters and is available via the CALLERID(ani2) dialplan + function. (closes issue ASTERISK-19912) Patch-by: Rob Gagnon + Review: https://reviewboard.asterisk.org/r/1947/ + +2012-06-11 17:34 +0000 [r368772] Richard Mudgett + + * main/channel.c, channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_sip.c, include/asterisk/channel.h, + channels/chan_iax2.c: Fix deadlock potential with + ast_set_hangupsource() calls. Calling ast_set_hangupsource() with + the channel lock held can result in a deadlock because the + function also locks the bridged channel. (issue ASTERISK-19537) + (closes issue AST-891) Reported by: Guenther Kelleter Tested by: + Guenther Kelleter (closes issue ASTERISK-19801) Reported by: Alec + Davis ........ Merged revisions 368759 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368760 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-11 15:23 +0000 [r368722-368751] Kinsey Moore + + * channels/sip/sdp_crypto.c, /, channels/chan_sip.c, main/say.c, + res/res_fax.c, channels/sip/reqresp_parser.c, apps/app_queue.c, + main/loader.c, channels/chan_dahdi.c, res/res_config_odbc.c, + channels/sip/dialplan_functions.c, apps/app_directory.c, + pbx/pbx_config.c, res/res_odbc.c, res/res_speech.c, + apps/app_voicemail.c: Fix coverity UNUSED_VALUE findings in core + support level files Most of these were just saving returned + values without using them and in some cases the variable being + saved to could be removed as well. (issue ASTERISK-19672) + ........ Merged revisions 368738 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368739 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /: Recorded merge of revisions 368721 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Fix + compilation in dev-mode Backport a compilation fix in md5.c from + trunk that only showed up in dev-mode under certain compiler + versions. ........ Merged revisions 368719 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2012-06-08 21:08 +0000 [r368712-368714] Richard Mudgett + + * main/manager.c, main/utils.c, include/asterisk/strings.h: Fix + error paths in action_hangup() for AMI Hangup action. * Check + allocation function return values for failure. Crashing is bad. * + Tweak ast_regex_string_to_regex_pattern() parameters for proper + ast_str usage. + + * main/channel.c, include/asterisk/channel.h: Tweak + ast_channel_softhangup_withcause_locked() to take a typed + parameter. + +2012-06-08 08:32 +0000 [r368688] Igor Goncharovskiy + + * channels/chan_unistim.c: Fix MWI update so LED display correct + voicemail state after phone usage. Also fixes few warnings. + (closes issue #19675) Reported by: dbohling Patches: fixmwi.patch + uploaded by dbohling (license 6378) + +2012-06-07 21:44 +0000 [r368680-368681] Damien Wedhorn + + * channels/chan_skinny.c: Skinny cleanup (mwi_event_cb). Original + was testing for d->session, setting and testing again (all + nested). Removed duplicate testing and restructured function to + test/return and then the main code. + + * channels/chan_skinny.c: Skinny cleanup. Removed d->registered + which was mirroring d->session. Changed relevant references to + use d->session instead. Moved setting and unsetting of l->device + from session register to device configuration. As such, l->device + will always be valid unless it is has not been configured to a + device. Revised various test where checking if a device is + registered to use l->device->session. + +2012-06-07 20:39 +0000 [r368674-368675] Richard Mudgett + + * apps/app_queue.c: Fix app_queue debug message use of args.options + after the string has been parsed. + + * apps/app_queue.c: Fix inverted test in app_queue for ringinuse. + Regression from -r367080 ringinuse commit. (issue ASTERISK-19536) + +2012-06-07 20:32 +0000 [r368673] Terry Wilson + + * main/udptl.c, include/asterisk/config_options.h, apps/app_skel.c, + main/config_options.c, tests/test_config.c: Fix reloading an + unchanged file with the Config Options API Adding multiple file + support broke reloading an unchanged file. This adds an enum for + return values for the aco_process_* functions and ensures that + the config is not applied if res is not ACO_PROCESS_OK. Review: + https://reviewboard.asterisk.org/r/1979/ + +2012-06-07 20:00 +0000 [r368668] Tzafrir Cohen + + * formats/format_ogg_vorbis.c: Fix a typo in format_ogg_vorbis.c: + suport Review: https://reviewboard.asterisk.org/r/1970/ + +2012-06-07 15:43 +0000 [r368663] Terry Wilson + + * include/asterisk/config_options.h, main/config_options.c, + tests/test_config.c: Add default handler documentation and + standardize acl handler Added documentation describing what flags + and arguments to pass to aco_option_register for default option + types. Also changed the ACL handler to use the flags parameter to + differentiate between "permit" and "deny" instead of adding an + additional vararg parameter. Review: + https://reviewboard.asterisk.org/r/1969/ + +2012-06-06 21:34 +0000 [r368646] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Fix POTS flash + hook to orignate a second call deadlock. A deadlock can occur + when a POTS phone tries to flash hook to originate a second call + for 3-way or transfer. If another process is scanning the + channels container when the POTS line flash hooks then a deadlock + will occur. * Release the channel and private locks when creating + a new channel as a result of a flash hook. (closes issue + ASTERISK-19842) Reported by: rmudgett Tested by: rmudgett + ........ Merged revisions 368644 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368645 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-06 19:25 +0000 [r368637] Mark Michelson + + * /, channels/chan_sip.c: Fix a specific scenario where ACKs are + not matched. If a dialog-starting INVITE contains a to-tag, then + Asterisk will respond with a 481. In this case, the resulting + incoming ACK would not be matched, so Asterisk would continue + retransmitting the 481 until the transaction times out. There + were two issues. Asterisk, upon creating a sip_pvt would generate + a local tag. However, when the time came to transmit the 481, + since there was a to-tag in the INVITE, Asterisk would place this + original to-tag in the 481 response. When the ACK came in, + Asterisk would attempt to match the to-tag in the ACK to the + generated local tag. Unfortunately, Asterisk never actually + transmitted a response with the generated local tag, so the + to-tag in the ACK would not match. The other problem was that + when the 481 was sent, nothing was set on the sip_pvt to indicate + what CSeq is expected in the ACK. To fix the first problem, we + zero out the to-tag seen in the incoming INVITE. This way, + Asterisk, when time to send a response, will send its generated + local tag instead. To fix the second problem, we set the + sip_pvt's pendinginvite to the CSeq of the INVITE when we send a + 481. (closes issue ASTERISK-19892) Reported by Mark Michelson + ........ Merged revisions 368625 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368629 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-06 17:22 +0000 [r368606] Matthew Jordan + + * /, build_tools/make_version: Add feature modifier to versions + produced from branches Certain branches, such as Certified + Asterisk, may have a modifier added to them that specifies the + features available in that branch. For branches, this modifier is + expected to be reflected in the location of the branch in + subversion. For example, a subversion of URL of + /certified/branches/1.8.11 would have a feature modifier of + 'certified'. This is slightly different then how features are + determined for tags, where the feature is part of the actual tag + name, e.g., "10.5.0-digiumphones". In keeping with the + nomenclature used for tags, the feature specifier for branches is + translated and placed after the revision numbers. For the example + given previously, this would result in a branch version of + "Asterisk SVN-branch-1.8.11-cert-rXXXXXX". ........ Merged + revisions 368604 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368605 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-06 16:11 +0000 [r368588] Kinsey Moore + + * /, channels/chan_sip.c: Ensure overlapping hold flags do not + conflict When changing between different modes of hold, the flags + were not being cleared out properly causing a failure to change + hold states. (closes issue ASTERISK-19919) Patch-by: Morten + Tryfoss Reported-by: Morten Tryfoss ........ Merged revisions + 368586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 368587 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-06 01:11 +0000 [r368566-368569] Richard Mudgett + + * /, main/features.c: Fix parked call performing a DTMF blind + transfer after being retrieved. When a parked call was retrieved + from the parking lot, it could not do a blind transfer because it + caused the involved calls to be hung up unconditionally. * Made + the ParkedCall application return the ast_bridge_call() return + value. (closes issue ABE-2862) Reported by: Vlad Povorozniuc + ........ Merged revisions 368567 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368568 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/features.c: Make builtin_blindtransfer() fully use + ast_async_goto() abilities. + +2012-06-05 16:25 +0000 [r368550] Jonathan Rose + + * CHANGES: Merge 'core' and 'core changes' sections in CHANGES + file. + +2012-06-05 15:28 +0000 [r368519-368537] Kinsey Moore + + * /: Recorded merge of revisions 368536 from + http://svn.asterisk.org/svn/asterisk/branches/10 ........ Resolve + some build warnings My newly upgraded compiler caught these + usages of uninitialized values. They weren't actually used. + ........ Merged revisions 368533 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * /, apps/app_voicemail.c: Ensure that pages and emails are sent + using RFC822-compliant date format When localization was added to + app_voicemail, these headers were altered when they should have + remained in en_US format for RFC compliance. This reverts the + changes to those two lines. (closes issue ASTERISK-19876) + ........ Merged revisions 368520 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368524 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_dial.c, channels/chan_unistim.c, channels/chan_local.c, + channels/chan_sip.c, main/channel_internal_api.c, + main/features.c, include/asterisk/channel.h, apps/app_queue.c: + Convert AST_FLAG_ANSWERED_ELSEWHERE usage to + AST_CAUSE_ANSWERED_ELSEWHERE This was essentially duplicated + functionality where normal channels used + AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used + AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts + that usage into AST_CAUSE_ANSWERED_ELSEWHER usage. Review: + https://reviewboard.asterisk.org/r/1944 (closes issue + ASTERISK-19865) Patch-by: Birger Harzenetter + +2012-06-04 22:12 +0000 [r368500] Mark Michelson + + * /, channels/chan_sip.c: Relay proper SIP responses on calling + side. Revision 351130 broke corect HANGUPCAUSE setting for the + 404 case in chan_sip. Other cases were also potentially broken. + This patch fixes the relaying of causes to be what they used to + be. (closes issue ASTERISK-19914) Reported by Pavel Troller + Tested by Walter Doekes (via a reviewboard test to be committed + later) Patches: chan_sip.diff uploaded by Pavel Troller (license + #6302) ........ Merged revisions 368498 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368499 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-04 21:18 +0000 [r368472] Richard Mudgett + + * /, UPGRADE.txt: Document BLINDTRANSFER behavior change. (issue + ASTERISK-19322) (closes issue ASTERISK-19875) Reported by: call + ........ Merged revisions 368469 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368470 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-04 20:53 +0000 [r368435-368467] Mark Michelson + + * contrib/editors/asterisk.vim: Also have vim syntax-highlight + type=network. + + * contrib/editors/asterisk.vim: Add vim syntax highlighting for + type=line, type=phone, and type=application. (closes issue + ASTERISK-19800) Reported by: Billy Chia Patches: + asterisk.vim.patch uploaded by Billy Chia (license #6381) + + * main/channel.c, apps/app_mixmonitor.c: Remove some extra + debugging I forgot to remove in the merge of Digium phone + support. + + * /: Remove automerge properties. + + * /, contrib/realtime/mysql/voicemail_messages.sql, + main/presencestate.c (added), main/config.c, main/channel.c, + include/asterisk/callerid.h, include/asterisk/file.h, + main/manager.c, channels/chan_skinny.c, + include/asterisk/event_defs.h, include/asterisk/sip_api.h + (added), tests/test_voicemail_api.c (added), main/features.c, + apps/app_voicemail.exports.in, main/app.c, main/message.c, + channels/sip/include/sip.h, main/pbx.c, channels/chan_sip.c, + include/asterisk/presencestate.h (added), + include/asterisk/config.h, include/asterisk/app_voicemail.h + (added), configs/manager.conf.sample, apps/app_queue.c, + include/asterisk/manager.h, include/asterisk/app.h, + funcs/func_presencestate.c (added), include/asterisk/message.h, + main/file.c, main/callerid.c, main/event.c, + include/asterisk/pbx.h, tests/test_config.c, + channels/chan_sip.exports.in (added), apps/app_mixmonitor.c, + main/asterisk.c, apps/app_voicemail.c: Merge changes dealing with + support for Digium phones. Presence support has been added. This + is accomplished by allowing for presence hints in addition to + device state hints. A dialplan function called PRESENCE_STATE has + been added to allow for setting and reading presence. Presence + can be transmitted to Digium phones using custom XML elements in + a PIDF presence document. Voicemail has new APIs that allow for + moving, removing, forwarding, and playing messages. Messages have + had a new unique message ID added to them so that the APIs will + work reliably. The state of a voicemail mailbox can be obtained + using an API that allows one to get a snapshot of the mailbox. A + voicemail Dialplan App called VoiceMailPlayMsg has been added to + be able to play back a specific message. Configuration hooks have + been added. Configuration hooks allow for a piece of code to be + executed when a specific configuration file is loaded by a + specific module. This is useful for modules that are dependent on + the configuration of other modules. chan_sip now has a public + method that allows for a custom SIP INFO request to be sent + mid-dialog. Digium phones use this in order to display progress + bars when files are played. Messaging support has been expanded a + bit. The main visible difference is the addition of an AMI action + MessageSend. Finally, a ParkingLots manager action has been added + in order to get a list of parking lots. + +2012-06-04 19:46 +0000 [r368421] Richard Mudgett + + * main/channel.c, /: Fix potential deadlock between masquerade and + chan_local. * Restructure ast_do_masquerade() to not hold channel + locks while it calls ast_indicate(). * Simplify many calls to + ast_do_masquerade() since it will never return a failure now. If + it does fail internally because a channel driver callback + operation failed, the only thing ast_do_masquerade() can do is + generate a warning message about strange things may happen and + press on. * Fixed the call to ast_bridged_channel() in + ast_do_masquerade(). This change fixes half of the deadlock + reported in ASTERISK-19801 between masquerades and chan_iax. + (closes issue ASTERISK-19537) Reported by: rmudgett Tested by: + rmudgett Review: https://reviewboard.asterisk.org/r/1915/ + ........ Merged revisions 368405 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368407 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-02 21:13 +0000 [r368359] Joshua Colp + + * include/asterisk/utils.h, res/res_http_websocket.exports.in + (added), include/asterisk/http_websocket.h (added), main/utils.c, + res/res_http_websocket.c (added): Add res_http_websocket module + which implements the WebSocket protocol according to RFC 6455. + Review: https://reviewboard.asterisk.org/r/1952/ + +2012-06-01 23:53 +0000 [r368311] Richard Mudgett + + * /, apps/app_stack.c: Fix deadlock when Gosub used with alternate + dialplan switches. Attempting to remove a channel from + autoservice with the channel lock held will result in deadlock. * + Restructured gosub_exec() to not call ast_parseable_goto() and + ast_exists_extension() with the channel lock held. (closes issue + ASTERISK-19764) Reported by: rmudgett Tested by: rmudgett + ........ Merged revisions 368308 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368310 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-01 20:42 +0000 [r368268-368269] Kevin P. Fleming + + * channels/chan_sip.c: Improve SDP offer/answer RFC compliance + Asterisk should not accept SDP offers that contain unknown RTP + profiles (for audio/video streams) or unknown top-level media + types. When it does, it answers with an SDP that does not match + the offer properly, and this will nearly always result in a + broken call. This patch causes such offers to be rejected. + Review: https://reviewboard.asterisk.org/r/1811/ + + * /, channels/chan_sip.c: Improve SDP parsing warning messages * + 'Unsupported media type' is only reported when that is in fact + the case, not when a supported media type is included in an 'm' + line that has an invalid format. * All warning messages related + to parsing 'm' lines now include the 'm' line contents. * (minor + bugfix) newline added to port-number-zero warning messages. * + Warning messages improved to use RFC-specified terminology for + various items. * Warnings for offers that include more than one + port for a single media type now include the media type. Review: + https://reviewboard.asterisk.org/r/1811/ ........ Merged + revisions 368218 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368267 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-01 18:20 +0000 [r368181-368221] Terry Wilson + + * configs/config_test.conf.sample (added): Add missing config for + config API test + + * main/udptl.c, include/asterisk/utils.h, + include/asterisk/astobj2.h, configure.ac, + include/asterisk/config.h, main/astobj2.c, main/config.c, + Makefile, include/asterisk/config_options.h (added), configure, + main/asterisk.exports.in, apps/app_skel.c, main/config_options.c + (added), tests/test_config.c, makeopts.in, + configs/app_skel.conf.sample (added), + include/asterisk/stringfields.h: Add new config-parsing framework + This framework adds a way to register the various options in a + config file with Asterisk and to handle loading and reloading of + that config in a consistent and atomic manner. Review: + https://reviewboard.asterisk.org/r/1873/ + +2012-06-01 13:04 +0000 [r368143] Mark Michelson + + * channels/chan_sip.c, configs/sip.conf.sample, + channels/sip/include/sip.h: Help mitigate potential reinvite + glare scenarios. When Asterisk servers are set up back-to-back, + and direct media is to be used betweeen endpoints, it is fairly + common for the two Asterisk servers to send direct media + reinvites to each other simultaneously. This results in 491s and + ACKs being exchanged between the servers. While the media + eventually gets set up properly, the problem is that there can be + a noticeable delay for the streams to stabilize. This patch adds + a new directmedia option called "outgoing". With this set, an + immediate direct media reinvite will only be sent if the call + direction is outgoing. For incoming dialogs, an immediate direct + media reinvite will not be sent, but further "reactionary" direct + media reinvites may be sent. Review: + https://reviewboard.asterisk.org/r/1954 + +2012-06-01 03:30 +0000 [r368094] Michael L. Young + + * /, funcs/func_channel.c: Add documentation to function CHANNEL + for options echocan_mode and buffers The ability to set + "echocan_mode" and "buffers" through the dialplan was added to + chan_dahdi some time ago. This patch adds some documentation to + func_channel. (Closes issue ASTERISK-19911) Reported by: Dale + Noll Tested by: Michael L. Young Patches: + asterisk-19911-branch18.diff uploaded by Michael L. Young + (license 5026) Review: https://reviewboard.asterisk.org/r/1949/ + ........ Merged revisions 368092 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368093 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-31 18:39 +0000 [r368052] Richard Mudgett + + * res/ael/pval.c, main/tcptls.c, main/manager.c, + res/res_config_odbc.c, /, channels/chan_sip.c, + channels/chan_agent.c, funcs/func_math.c, main/features.c, + apps/app_queue.c, channels/chan_iax2.c, pbx/pbx_config.c: + Coverity Report: Fix issues for error type REVERSE_INULL (core + modules) * Fixes findings: 0-2,5,7-15,24-26,28-31 (issue + ASTERISK-19648) Reported by: Matt Jordan ........ Merged + revisions 368039 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 368042 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-30 18:08 +0000 [r367908-367982] Richard Mudgett + + * /: Use the DEADLOCK_AVOIDANCE() macro instead. (issue + ASTERISK-19854) ........ Merged revisions 367980 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367981 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/sig_pri.c, channels/sig_ss7.c: Fix deadlock when + executing CLI "pri show channels" and "ss7 show channels" + commands. * Fix sig_pri_lock_owner() to avoid deadlock properly. + * Code pri_grab() better. * Fix sig_ss7_lock_owner() to avoid + deadlock properly. * Code ss7_grab() better. (closes issue + ASTERISK-19854) Reported by: Jaxon Patches: + jira_asterisk_19854_v1.8.6.patch (license #5621) patch uploaded + by rmudgett (Modified to do the same thing to sig_ss7) Tested by: + Jaxon ........ Merged revisions 367976 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367978 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_meetme.c: Coverity Report: Fix issues for error type + REVERSE_INULL (deprecated modules) * Fix only issue pointed out + by deprecated_REVERSE_INULL.txt for app_meetme.c in find_user(). + * Change use of %i to %d in sscanf() in find_user(). The use of + %i gives unexpected parsing because it can accept hex, octal, and + decimal integer formats. * Changed other uses of %i in + app_meetme() to use %d for consistency. (issue ASTERISK-19648) + Reported by: Matt Jordan ........ Merged revisions 367906 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367907 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-29 18:40 +0000 [r367845] Matthew Jordan + + * /, channels/chan_skinny.c: AST-2012-008: Fix remote crash + vulnerability in chan_skinny When a skinny session is + unregistered, the corresponding device pointer is set to NULL in + the channel private data. If the client was not in the on-hook + state at the time the connection was closed, the device pointer + can later be dereferened if a message or channel event attempts + to use a line's pointer to said device. The patches prevent this + from occurring by checking the line's pointer in message handlers + and channel callbacks that can fire after an unregistration + attempt. (closes issue ASTERISK-19905) Reported by: Christoph + Hebeisen Tested by: mjordan, Damien Wedhorn Patches: + AST-2012-008-1.8.diff uploaded by mjordan (license 6283) + AST-2012-008-10.diff uploaded by mjordan (licesen 6283) ........ + Merged revisions 367844 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-25 16:33 +0000 [r367783] Richard Mudgett + + * /, channels/chan_iax2.c: AST-2012-007: Fix IAX receiving HOLD + without suggested MOH class crash. * Made schedule_delivery() set + the received frame f->data.ptr to NULL if the datalen is zero. * + Fix queue_signalling() memcpy() size error. * Made + queue_signalling() not use C++ keyword variable names. (closes + issue ASTERISK-19597) Reported by: mgrobecker Patches: + jira_asterisk_19597_v1.8.patch (license #5621) patch uploaded by + rmudgett Tested by: rmudgett, Michael L. Young ........ Merged + revisions 367781 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367782 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-25 02:31 +0000 [r367732] Michael L. Young + + * /, channels/chan_sip.c: Fix pvt_sip for inbound call to use + peer's allowtransfer setting The pvt_sip allowtransfer was not + being set to that of the peer's setting. Therefore, the global + allowtransfer setting was being used instead which would lead to + calls not being transfered if the global setting was set to 'no' + despite the setting on the peer being 'yes' and vice versa, calls + would be allowed to transfer even if the peer's setting was 'no' + but the global setting was 'yes'. (Closes issue ASTERISK-19856) + Reported by: Jacek Tested by: Michael L. Young, Jacek Patches: + issue-asterisk-19856-branch10-v3.diff uploaded by Michael L. + Young (license 5026) Review: + https://reviewboard.asterisk.org/r/1923/ ........ Merged + revisions 367730 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367731 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-24 23:52 +0000 [r367693] Richard Mudgett + + * apps/app_dial.c, /, apps/app_queue.c: Fix Dial I option ignored + if dial forked and one fork redirects. The Dial and Queue I + option is intended to block connected line updates and + redirecting updates. However, it is a feature that when a call is + locally redirected, the I option is disabled if the redirected + call runs as a local channel so the administrator can have an + opportunity to setup new connected line information. + Unfortunately, the Dial and Queue I option is disabled for *all* + forked calls if one of those calls is redirected. * Make the Dial + and Queue I option apply to each outgoing call leg independently. + Now if one outgoing call leg is locally redirected, the other + outgoing calls are not affected. * Made Dial not pass any + redirecting updates when forking calls. Redirecting updates do + not make sense for this scenario. * Made Queue not pass any + redirecting updates when using the ringall strategy. Redirecting + updates do not make sense for this scenario. * Fixed deadlock + potential with chan_local when Dial and Queue send redirecting + updates for a local redirect. * Converted the Queue stillgoing + flag to a boolean bitfield. (closes issue ASTERISK-19511) + Reported by: rmudgett Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/1920/ ........ Merged + revisions 367678 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367679 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-24 18:56 +0000 [r367640] Jonathan Rose + + * main/rtp_engine.c, channels/chan_sip.c, + include/asterisk/rtp_engine.h: chan_sip: fix problem + directmediapermit/deny uses the wrong address When remotely + bridging calls with directmedia, Asterisk would check the address + of the peers/users holding directmedia ACLs (set via + directmediapermit/directmediadeny) instead of the bridged peer. + This is similar to r366547, but trunk specific and involves + changes to the rtpengine instead of just chan_sip. (closes issue + AST-876) review: https://reviewboard.asterisk.org/r/1924/ + +2012-05-24 13:33 +0000 [r367563] Matthew Jordan + + * /, apps/app_confbridge.c: Fix crash in ConfBridge when user + announcement is played for more than 2 users A patch introduced + in r354938 made it so that ConfBridge would not attempt to play + sound files if those files did not exist. Unfortunately, + ConfBridge uses the same underlying function, play_sound_helper, + to playback both sound files and numbers to callers. When a + number is being played back, the name of the sound file is + expected to be NULL. This NULL value was passed into a function + that tested for the existance of a sound file and is not tolerant + to NULL file names, causing a crash. This patch fixes the + behavior, such that if a sound file does not exist we do not + attempt to play it, but we only attempt that check if the a sound + file was specified in the first place. If a sound file was not + specified, we use the 'play number' logic in the helper function. + (closes issue ASTERISK-19899) Reported by: Florian Gilcher Tested + by: Florian Gilcher patches: asterisk-19899.diff uploaded by + mjordan (license 6283) ........ Merged revisions 367562 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-24 00:36 +0000 [r367477-367520] Richard Mudgett + + * channels/iax2-parser.c: Made use IAX frame cache only for + cacheable frame types. + + * main/pbx.c, /: Fix WaitExten(x,m(musicclass)) string termination. + The AST_CONTROL_HOLD MOH class from the WaitExten application can + now be queued onto a channel, passed over local channels with the + /m option, and passed over IAX channels. ........ Merged + revisions 367469 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367470 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-23 20:39 +0000 [r367419] Jonathan Rose + + * main/pbx.c: logger: Fix a potential callid reference leak + discovered in development Uncovered a nasty reference leak while + I was writing some changes to chan_dahdi/sig_analog. Slapped + myself around a bit after seeing that I performed the unchecked + return causing this problem. + +2012-05-23 20:30 +0000 [r367418] Mark Michelson + + * main/tcptls.c, /: Only call SSL_CTX_free if DO_SSL is defined. + Thanks to Paul Belanger for pointing out this error. ........ + Merged revisions 367416 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367417 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-23 13:46 +0000 [r367376] Matthew Jordan + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Re-add + LastMsgsSent value for SIP peers Previously, MWI logic utilized a + counter called 'lastmsgssent' to know whether or not MWI NOTIFY + requests had been sent to a specific peer. When MWI notifications + were changed to use the internal event framework, this value was + no longer needed for its original purpose. Hence, it was no + longer updated with the new/old message counts for a peer. The + value was previously removed for Asterisk 10; however, since it + was still present in Asterisk 1.8 and still useful for reporting + purposes, it was decided to re-add the value. This patch re-adds + the 'LastMsgsSent' field in the response to an AMI/CLI 'sip show + peer [peer]' command, and makes it so that the value of + lastmsgssent is updated appropriately. The value should now + display the new/old message counts for a particular peer. (closes + issue ASTERISK-17866) Reported by: Steve Davies patches by: + ast-17866-rb1272.patch (License #5041 by irroot) Modified + slightly for this commit Review: + https://reviewboard.asterisk.org/r/1939 ........ Merged revisions + 367362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 367369 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-22 17:29 +0000 [r367274-367309] Terry Wilson + + * main/channel.c, /, include/asterisk/cel.h, + main/channel_internal_api.c, include/asterisk/channel.h, + main/cel.c, main/asterisk.c: Fix race condition for CEL + LINKEDID_END event This patch fixes to situations that could + cause the CEL LINKEDID_END event to be missed. 1) During a core + stop gracefully, modules are unloaded when ast_active_channels == + 0. The LINKDEDID_END event fires during the channel destructor. + This means that occasionally, the cel_* module will be unloaded + before the channel is destroyed. It seemed generally useful to + wait until the refcount of all channels == 0 before unloading, so + I added a channel counter and used it in the shutdown code. 2) + During a masquerade, ast_channel_change_linkedid is called. It + calls ast_cel_check_retire_linkedid which unrefs the linkedid in + the linkedids container in cel.c. It didn't ref the new linkedid. + Now it does. Review: https://reviewboard.asterisk.org/r/1900/ + ........ Merged revisions 367292 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367299 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Resolve crash in subscribing for MWI + notifications ASTOBJ_UNREF sets the variable to NULL after + unreffing it, so the variable should definitely not be used after + that. To solve this in the two cases that affect subscribing for + MWI notifications, we instead save the ref locally, and unref + them in the error conditions. (closes issue ASTERISK-19827) + Reported by: B. R Review: + https://reviewboard.asterisk.org/r/1940/ ........ Merged + revisions 367266 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367267 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-21 22:45 +0000 [r367227] Richard Mudgett + + * main/channel.c: Made ast_queue_hangup() and + ast_queue_hangup_with_cause() lock instead of trylock. It made no + sense to trylock the channel and then unconditionally lock the + channel right after. + +2012-05-21 20:35 +0000 [r367189] Kinsey Moore + + * channels/chan_iax2.c: Make chan_iax2 reject cause code + indications correctly If chan_iax2 does not reject the + PVT_CAUSE_CODE frames, the cause will not be stored properly. + +2012-05-21 20:31 +0000 [r367163-367183] Mark Michelson + + * include/asterisk/callerid.h, channels/chan_sip.c, + main/callerid.c: Revert revision 367163. This should have been + committed to my team trunk-digiumphones branch instead of trunk. + + * include/asterisk/callerid.h, channels/chan_sip.c, + main/callerid.c: Add "send to voicemail" Digium phone + functionality to Asterisk. This change accommodates two methods + by which calls can be directed to a user's voicemail. * Incoming + calls can be redirected to any user's voicemail. * Established + calls can be blind transferred to any user's voicemail. Digium + phones indicate the desire to direct a call to voicemail by using + a Diversion header with a reason parameter of "send_to_vm". This + patch adds the "send_to_vm" reason as a valid redirecting reason. + In addition, chan_sip.c has been modified to update redirecting + information on the transferred channel by reading a Diversion + header on a REFER request. (closes issue AST-871) Reported by + Malcolm Davenport Review: https://reviewboard.asterisk.org/r/1925 + +2012-05-21 17:39 +0000 [r367124] Terry Wilson + + * include/asterisk/astobj2.h: Minor documentation change + +2012-05-18 19:39 +0000 [r367080] Jonathan Rose + + * configs/queues.conf.sample, CHANGES, apps/app_queue.c: app_queue: + Per Member ringinuse option and deprecation of ignorebusy Adds a + number of methods for controlling the setting of 'ringinuse' + which is basically the same concept as the old ignorebusy + setting, only now the per member setting always controls whether + or not the member is actually ringed while in use. A CLI command + and a manager action have been added to change a given queue + member's ringinuse option while Asterisk is running and the an + argument has been added for adding members with deliberately set + ringinuse in queues.conf Some effort has been made to ensure + compatability with dialplans and databases still referring to + 'ignorebusy'. (issue ASTERISK-19536) reported by: Philippe + Lindheimer Review: https://reviewboard.asterisk.org/r/1919/ + +2012-05-18 17:54 +0000 [r367010-367029] Mark Michelson + + * channels/chan_dahdi.c, /, main/say.c: Address MISSING_BREAK + static analysis reports some more. This addresses core findings 4 + and 6. Moises Silva helped me by stating that a break could be + safely added to the case where it is added in chan_dahdi.c In + say.c, I have added a comment indicating that static analysis + complains but that it is currently unknown if this is correct. + This fixes all core findings of this type. (closes issue + ASTERISK-19662) reported by Matthew Jordan ........ Merged + revisions 367027 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367028 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h: + Fix memory leak of SSL_CTX structures in TLS core. SSL_CTX + structures were allocated but never freed. This was a bigger + issue for clients than servers since new SSL_CTX structures could + be allocated for each connection. Servers, on the other hand, + typically set up a single SSL_CTX for their lifetime. This is + solved in two ways: 1. In __ssl_setup(), if a tcptls_cfg has an + ssl_ctx on it, it is freed so that a new one can take its place. + 2. A companion to ast_ssl_setup() called ast_ssl_teardown() has + been added so that servers can properly free their SSL_CTXs. + (issue ASTERISK-19278) ........ Merged revisions 367002 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 367003 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-18 15:51 +0000 [r366917-366955] Matthew Jordan + + * channels/chan_dahdi.c, /, channels/chan_sip.c, funcs/func_odbc.c, + main/cli.c: Fix more memory leaks This patch adds to what was + fixed in r366880. Specifically, it addresses the following: * + chan_sip: dispose of an allocated frame in off nominal code paths + in sip_rtp_read * func_odbc: when disposing of an allocated + resultset, ensure that any rows that were appended to that + resultset are also disposed of * cli: free the created return + string buffer in another off nominal code path * chan_dahdi: free + a frame that was allocated by the dsp layer if we choose not to + process that frame (issue ASTERISK-19665) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/1922/ ........ + Merged revisions 366944 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 366948 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/netsock2.c, res/res_rtp_asterisk.c, main/pbx.c, + res/res_calendar_exchange.c, res/res_calendar_icalendar.c, + apps/app_page.c, /, funcs/func_dialgroup.c, channels/chan_sip.c, + apps/app_record.c, res/res_calendar_caldav.c, res/res_jabber.c, + apps/app_queue.c, channels/chan_iax2.c, main/enum.c, + main/editline/term.c, main/config.c, res/res_srtp.c, main/cli.c, + main/editline/tokenizer.c, main/data.c, channels/chan_dahdi.c, + funcs/func_odbc.c, main/features.c, apps/app_minivm.c, + main/editline/readline.c, channels/sip/config_parser.c, + main/xmldoc.c, res/res_calendar.c, apps/app_voicemail.c: Fix a + variety of memory leaks This patch addresses a number of memory + leaks in a variety of modules that were found by a static + analysis tool. A brief summary of the changes: * app_minivm: free + ast_str objects on off nominal paths * app_page: free the + ast_dial object if the requested channel technology cannot be + appended to the dialing structure * app_queue: if a penalty rule + failed to match any existing rule list names, the created rule + would not be inserted and its memory would be leaked * app_read: + dispose of the created silence detector in the presence of off + nominal circumstances * app_voicemail: dispose of an allocated + unique ID field for MWI event un-subscribe requests in off + nominal paths; dispose of configuration objects when using the + secret.conf option * chan_dahdi: dispose of the allocated frame + produced by ast_dsp_process * chan_iax2: properly unref peer in + CLI command "iax2 unregister" * chan_sip: dispose of the + allocated frame produced by sip_rtp_read's call of + ast_dsp_process; free memory in parse unit tests * + func_dialgroup: properly deref ao2 object grhead in nominal path + of dialgroup_read * func_odbc: free resultset in off nominal + paths of odbc_read * cli: free match_list in off nominal paths of + CLI match completion * config: free comment_buffer/list_buffer + when configuration file load is unchanged; free the same buffers + any time they were created and config files were processed * + data: free XML nodes in various places * enum: free context + buffer in off nominal paths * features: free ast_call_feature in + off nominal paths of applicationmap config processing * netsock2: + users of ast_sockaddr_resolve pass in an ast_sockaddr struct that + is allocated by the method. Failures in ast_sockaddr_resolve + could result in the users of the method not knowing whether or + not the buffer was allocated. The method will now not allocate + the ast_sockaddr struct if it will return failure. * pbx: cleanup + hash table traversals in off nominal paths; free ignore pattern + buffer if it already exists for the specified context * xmldoc: + cleanup various nodes when we no longer need them * + main/editline: various cleanup of pointers not being freed before + being assigned to other memory, cleanup along off nominal paths * + menuselect/mxml: cleanup of value buffer for an attribute when + that attribute did not specify a value * res_calendar*: responses + are allocated via the various *_request method returns and should + not be allocated in the various write_event methods; ensure + attendee buffer is freed if no data exists in the parsed node; + ensure that calendar objects are de-ref'd appropriately * + res_jabber: free buffer in off nominal path * res_musiconhold: + close the DIR* object in off nominal paths * res_rtp_asterisk: if + we run out of ports, close the rtp socket object and free the rtp + object * res_srtp: if we fail to create the session in libsrtp, + destroy the temporary ast_srtp object (issue ASTERISK-19665) + Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/1922 ........ Merged revisions + 366880 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 366881 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-18 14:27 +0000 [r366896] Jonathan Rose + + * channels/sip/dialplan_functions.c: chan_sip: Fix a small + TEST_FRAMEWORK related error that prevents compiling Introduced + with r366842, a function call made only with TEST_FRAMEWORK + enabled was missing an argument since the function arguments were + changed. + +2012-05-18 14:21 +0000 [r366843-366888] Kinsey Moore + + * /, channels/sip/config_parser.c: Reorder and renumber tests + appropriately It appears that a patch did not apply properly when + adding tests 12 and 13 and test 11 was duplicated. These tests + have been reordered and renumbered such that they make sense. + ........ Merged revisions 366882 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 366884 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/channel.c: Make the new SIP_CAUSE backend behave more like + the original SIP_CAUSE There was a slight discrepancy in the + behaviors of the old SIP_CAUSE and the new SIP_CAUSE/HANGUPCAUSE + when a channel had been originated and had not yet been answered. + This caused the noload_res_srtp_attempt_srtp test to fail since + the SIP_CAUSE variable was never actually set. This behavior has + been restored. + +2012-05-17 16:28 +0000 [r366842] Jonathan Rose + + * include/asterisk/logger.h, main/channel.c, + channels/sip/include/dialog.h, main/pbx.c, channels/chan_sip.c, + main/channel_internal_api.c, main/logger.c, + include/asterisk/channel.h, CHANGES, channels/sip/include/sip.h, + main/cli.c: logger: Adds additional support for call id logging + and chan_sip specific stuff This patch improves the handling of + call id logging significantly with regard to transfers and adding + APIs to better handle specific aspects of logging. Also, changes + have been made to chan_sip in order to better handle the creation + of callids and to enable the monitor thread to bind itself to a + particular call id when a dialog is determined to be related to a + callid. It then unbinds itself before returning to normal + monitoring. review: https://reviewboard.asterisk.org/r/1886/ + +2012-05-17 13:21 +0000 [r366746] Matthew Jordan + + * channels/chan_dahdi.c, /, res/res_calendar_ews.c: Fix checking + bounds of array index after using it; improper sizeof This patch + fixes two problems pointed out by a static analysis tool. * In + chan_dahdi, when an event is handled the index of the sub channel + is first obtained. In very off nominal cases, the method that + determines the index can return a negative value. In the event + handling code, whether or not the index returned is valid was + being checked after that value was used to index into an array. + This patch makes it so the value is checked before any indexing + is done. * In res_calendar_ews, sizeof was being passed a pointer + instead of the struct to determine the amount of memory to + allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes + issue ASTERISK-19671) Reported by: Matt Jordan ........ Merged + revisions 366740 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 366741 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-16 18:00 +0000 [r366663-366700] Richard Mudgett + + * include/asterisk/astobj2.h: Remove missed idx parameter to some + ao2 global holder macros. + + * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c: + Change ao2 global array to ao2 global object holder. Review: + https://reviewboard.asterisk.org/r/1921/ + +2012-05-15 23:41 +0000 [r366599] Mark Michelson + + * /, channels/chan_sip.c: Correct misuse of ast_strip_quoted() when + getting a Diversion header's reason parameter. The use here was + assuming that the pointer would be updated, but the updated + string is actually returned by ast_strip_quoted() instead. + ........ Merged revisions 366597 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 366598 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-15 19:36 +0000 [r366462-366546] Richard Mudgett + + * channels/chan_local.c: The predial routine must be run on the + local;1 channel. When ast_call() operates on a local channel, it + copies a lot of things from the local;1 channel to the local;2 + channel. This includes among other things, channel variables and + party id information. Other reasons it was a bad idea to run + predial on the local;2 channel: 1) The channel has not been + completely setup. The ast_call() completes the setup. 2) The + local;2 caller and connected line party information is opposite + to any other channels predial runs on. (And it hasn't been setup + yet.) * Partially back out -r366183 by removing the chan_local + implementation of the struct ast_channel_tech.pre_call callback. + + * CHANGES, apps/app_followme.c: Add predial support to FollowMe. + Like the new predial feature for Dial. This adds the same b/B + options to FollowMe. Review: + https://reviewboard.asterisk.org/r/1910/ + + * channels/chan_local.c: Make chan_local use the API call instead + of inlining its own version. + +2012-05-14 20:15 +0000 [r366413] Mark Michelson + + * /, pbx/dundi-parser.c: Fix two more coverity constant expression + result findings. These correspond to findings 0 and 1 in the core + findings of ASTERISK-19649. After contacting Mark Spencer, he was + unsure of what the intent behind these lines of code were, so + they are being axed. For Asterisk 1.8 and 10, the output of + debugging DUNDi frames will not be changed, but for trunk the + "Retry" portion will be omitted since it does not properly + distinguish retransmissions from initial frames. (closes issue + ASTERISK-19649) Reported by Matthew Jordan ........ Merged + revisions 366409 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 366412 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-14 19:44 +0000 [r366408] Kinsey Moore + + * channels/chan_unistim.c, apps/app_dial.c, main/rtp_engine.c, + channels/chan_vpb.cc, channels/chan_sip.c, UPGRADE.txt, + channels/chan_gtalk.c, channels/chan_console.c, + channels/chan_iax2.c, apps/app_queue.c, apps/app_followme.c, + channels/chan_oss.c, channels/chan_jingle.c, main/channel.c, + channels/chan_phone.c, main/dial.c, channels/chan_misdn.c, + channels/chan_skinny.c, funcs/func_frame_trace.c, + main/features.c, channels/chan_h323.c, main/file.c, + channels/chan_alsa.c, configs/sip.conf.sample, + include/asterisk/frame.h, channels/chan_mgcp.c: Commit framework + for HANGUPCAUSE (replacement for SIP_CAUSE) This is the starting + point for the Asterisk 11: Who Hung Up work and provides a + framework which will allow channel drivers to report the types of + hangup cause information available in SIP_CAUSE without incurring + the overhead of the MASTER_CHANNEL dialplan function. The initial + implementation only includes cause generation for chan_sip and + does not include cause code translation utilities. This change + deprecates SIP_CAUSE and replaces its method of reporting cause + codes with the new framework. This change also deprecates the + 'storesipcause' option in sip.conf. Review: + https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) + +2012-05-14 19:27 +0000 [r366401] Mark Michelson + + * /, channels/chan_sip.c: Fix broken reinvite glare scenario. To + make a long story short, reinvite glares were broken because + Asterisk would invert the To and From headers when ACKing a 491 + response. The reason was because the initreq of the dialog was + being changed to the incoming glared reinvite instead of being + set to the outgoing glared reinvite. This change has three parts + * In handle_incoming, we never will reject an ACK because it has + a to-tag present, even if we think the request may be out of + dialog. * In handle_request_invite, we do not change the initreq + when receiving a reinvite to which we will respond with a 491. * + In handle_request_invite, several superflous settings up + pendinginvite have been removed since this is dones automatically + by transmit_response_reliable Review: + https://reviewboard.asterisk.org/r/1911 ........ Merged revisions + 366389 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 366390 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-14 13:42 +0000 [r366351] Tzafrir Cohen + + * configure, configure.ac, autoconf/ast_pkgconfig.m4 (added): Macro + AST_PKG_CONFIG_CHECK to use chkconfig AST_PKG_CONFIG_CHECK: + Similar to AST_EXT_LIB_CHECK, but simply uses pkg-config data. + This simple version only uses pkg-config(1)'s tests. This commit + also uses the macro to test for GTK2 and GMIME (instead of the + current direct usage of pkg-config). Review: + https://reviewboard.asterisk.org/r/1906/ + +2012-05-12 00:03 +0000 [r366298] Russell Bryant + + * /, addons/format_mp3.c: format_mp3: Fix a possible crash in + mp3_read(). This patch fixes a potential crash in mp3_read() by + not assuming that dbuf has enough data to finish filling up the + output buffer. The patch also makes sure that the dbuf state gets + reset after we know we read everything out of it already. In + passing, this patch includes some other cleanups of this module, + including stripping trailing whitespace, formatting fixes based + on coding guidelines, and removing a number of unused members + from the private state struct. (closes issue ASTERISK-19761) + Reported by: Chris Maciejewsk Tested by: Chris Maciejewsk + ........ Merged revisions 366296 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 366297 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-10 23:49 +0000 [r366183-366242] Richard Mudgett + + * main/channel.c, /: * Made ast_change_name() hold the channels + container lock while changing the channel name. * Eliminate + redundant list not empty check in clone_variables(). ........ + Merged revisions 366240 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 366241 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_dial.c: Tweak app_dial predial documentation. + + * apps/app_dial.c, main/channel.c, channels/chan_local.c, + include/asterisk/channel.h: Run predial routine on local;2 + channel where you would expect. Before this patch, the predial + routine executes on the ;1 channel of a local channel pair. + Executing predial on the ;1 channel of a local channel pair is of + limited utility. Any channel variables set by the predial routine + executing on the ;1 channel will not be available when the local + channel executes dialplan on the ;2 channel. * Create + ast_pre_call() and an associated pre_call() technology callback + to handle running the predial routine. If a channel technology + does not provide the callback, the predial routine is simply run + on the channel. Review: https://reviewboard.asterisk.org/r/1903/ + +2012-05-10 20:56 +0000 [r366169] Kinsey Moore + + * funcs/func_speex.c, main/pbx.c, res/res_calendar_icalendar.c, /, + channels/chan_sip.c, funcs/func_lock.c, channels/chan_agent.c, + channels/sip/reqresp_parser.c, main/devicestate.c, + pbx/dundi-parser.c, channels/chan_iax2.c, channels/iax2-parser.c, + main/config.c, res/res_monitor.c, main/channel.c, main/cdr.c, + res/ael/pval.c, main/data.c, channels/chan_dahdi.c, + main/tcptls.c, main/manager.c, main/features.c, main/app.c, + main/event.c, pbx/pbx_dundi.c, res/res_odbc.c, main/xmldoc.c, + apps/app_voicemail.c: Resolve FORWARD_NULL static analysis + warnings This resolves core findings from ASTERISK-19650 numbers + 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, + 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers + 26, 33, and 29 were already resolved. Those skipped were either + extended/deprecated or in areas of code that shouldn't be + disturbed. (Closes issue ASTERISK-19650) ........ Merged + revisions 366167 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 366168 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-10 18:35 +0000 [r366126] Jonathan Rose + + * main/pbx.c, channels/sig_analog.c, /, channels/chan_sip.c, + funcs/func_lock.c, main/features.c, main/acl.c, + channels/iax2-provision.c, apps/app_queue.c, + channels/chan_iax2.c, res/ael/ael.flex, funcs/func_devstate.c, + main/asterisk.c, main/xmldoc.c, apps/app_voicemail.c: Coverity + Report: Fix issues for error type CHECKED_RETURN for core (issue + ASTERISK-19658) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/1905/ ........ Merged + revisions 366094 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 366106 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-10 16:22 +0000 [r366062] Mark Michelson + + * /, channels/chan_sip.c: Close the proper tcptls_session when + session creation fails. (issue AST-998) Reported by: Thomas + Arimont Tested by: Thomas Arimont ........ Merged revisions + 366052 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 366053 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-10 15:57 +0000 [r366007-366051] Jonathan Rose + + * /, funcs/func_cdr.c, main/features.c, apps/app_disa.c, + apps/app_chanspy.c: Coverity Report: Fix issues for error type + UNINIT in Core supported modules (issue ASTERISK-19652) Reported + by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1909/ + ........ Merged revisions 366048 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 366049 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, codecs/codec_dahdi.c: Block on frameout if the hardware has + enough samples to complete a frame. Fixes some problems with + skipping audio in elaborate scenarios involving multiple codecs + by making codec_dahdi operate in a more synchronous fashion + similar to codec_g729. This change also fixes the use of file + conversion tools from Asterisk's CLI. This change may cause the + thread responsible for transcoding audio to block briefly (Shaun + Ruffell describes this as 'several milliseconds') while waiting + for the hardware transcoder. (closes issue ASTERISK-19643) + reported by: Shaun Ruffell Patches: + 0001-codec_dahdi-Block-on-frameout-the-hardware-has-enoug.patch + uploaded by Shaun Ruffell (license 5417) ........ Merged + revisions 365989 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 365990 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-09 19:26 +0000 [r366002] Tzafrir Cohen + + * Makefile: pass BUILD_CFLGAS and BUILD_LDFLAGS to menuselect Allow + menuselect to get its set of CFLAGS and LDFLAGS through the + environment of Make: make BUILD_CFLAGS="whatever" + BUILD_LDFLAGS="whatever" Review: + https://reviewboard.asterisk.org/r/1907/ + +2012-05-09 17:58 +0000 [r365951] Richard Mudgett + + * configs/followme.conf.sample, apps/app_followme.c: Improve + FollowMe accept/decline DTMF string matching. If you hit the + wrong DTMF digit trying to accept/decline a FollowMe call, you + had to wait for the prompt to repeat to try again. * Make + FollowMe compare the last DTMF digits received to the + accept/decline matching strings. + +2012-05-09 16:36 +0000 [r365913] Mark Michelson + + * /, channels/chan_sip.c: Prevent sip_pvt refleak when an + ast_channel outlasts its corresponding sip_pvt. chan_sip was + coded under the assumption that a SIP dialog with an owner + channel will always be destroyed after the owner channel has been + hung up. However, there are situations where the SIP dialog can + time out and auto destruct before the corresponding channel has + hung up. A typical example of this would be if the 'h' extension + in the dialplan takes a long time to complete. In such cases, + __sip_autodestruct() would complain about the dialog being auto + destroyed with an owner channel still in place. The problem is + that even once the owner channel was hung up, the sip_pvt would + still be linked in its ao2_container because nothing would ever + unlink it. The fix for this is that if __sip_autodestruct() is + called for a sip_pvt that still has an owner channel in place, + the destruction is rescheduled for 10 seconds in the future. This + will continue until the owner channel is finally hung up. (closes + issue ASTERISK-19425) reported by David Cunningham Patches: + ASTERISK-19425.patch uploaded by Mark Michelson (License #5049) + (closes issue ASTERISK-19455) reported by Dean Vesvuio Tested by + Dean Vesvuio ........ Merged revisions 365896 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 365898 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-09 02:35 +0000 [r365766-365856] Richard Mudgett + + * configs/followme.conf.sample, UPGRADE.txt, apps/app_followme.c: + Keep answered FollowMe calls until call accepted or last step + times out. + + * apps/app_followme.c: Put winning FollowMe outgoing call on hold + if the caller put it on hold. The FollowMe caller call leg is + usually answered and listening to MOH. The caller could put the + call on hold while FollowMe is looking for a winner. The winning + outgoing call is now immediately placed on hold if the caller has + put the call on hold before the winning call was selected. + + * apps/app_followme.c: Restructure how the FollowMe outgoing + channel list is handled. + + * apps/app_followme.c: Addendum to -r365766. Since it is no longer + allocated. + + * apps/app_followme.c: Make FollowMe findmeexec() put the list head + on the stack instead of mallocing it. Why this tiny struct was + malloced instead of the 28k struct in the last change is beyond + me. Just doing my part to help stamp out sillyness. + +2012-05-08 21:46 +0000 [r365751] Sean Bright + + * apps/app_externalivr.c: Add interrupt ('I') command to + ExternalIVR. Sending the 'I' command from an external process + will cause the current playlist to be cleared, including stopping + any audio file that is currently playing. This is useful when you + want to interrupt audio playback only when specific DTMF is + entered by the caller. + +2012-05-08 21:41 +0000 [r365633-365749] Richard Mudgett + + * apps/app_followme.c: Make FollowMe app_exec() not declare a 28k + struct on the stack. Helping to stamp out stack abuse. + + * apps/app_followme.c: Simplify findmeexec() parameter passing. + + * /, apps/app_followme.c: * Fix FollowMe memory leak on error paths + in app_exec(). * Fix FollowMe leaving recorded caller name file + on error paths in app_exec(). * Use correct buffer dimension + define in struct fm_args.namerecloc[]. This fixes unexpected + namerecloc filename length restriction. ........ Merged revisions + 365692 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 365701 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_followme.c: * Fix accept/decline DTMF buffer + overwrite in FollowMe. * Made use MAX_YN_STRING define to make + all accept/decline DTMF buffers the same size. Just using 20 + isn't good enough when someone didn't get the memo. * Fix stupid + use of a global variable in FollowMe. (ynlongest) * Fix bit field + declarations in FollowMe. ........ Merged revisions 365631 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 365632 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-08 15:57 +0000 [r365576] Mark Michelson + + * /, channels/chan_sip.c: Send more accurate identification + information in dialog-info SIP NOTIFYs. This uses the calling + channel's caller ID and connected line information to populate + the remote and local identities in the dialog-info NOTIFY when an + extension is ringing. There is a bit of an oddity here, and that + is that we seed the remote target with the To header of the + outbound call rather than the from header. This is because it was + reported that seeding with the from header caused hints to be + broken with certain SNOM devices. A comment has been added to the + code to explain this. (closes issue ASTERISK-16735) reported by + Maciej Krajewski patches: local_remote_hint2.diff uploaded by + Mark Michelson (license #5049) 16735_tweak1.diff uploaded by Mark + Michelson (license #5049) Tested by Niccolo Belli ........ Merged + revisions 365574 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 365575 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-07 20:08 +0000 [r365532] Richard Mudgett + + * main/features.c: Change comment to use local channel name + designators in features.c + +2012-05-07 18:58 +0000 [r365480] Matthew Jordan + + * main/pbx.c, apps/app_voicemail.c: Fix channel opaquification + slip-up in r365477 Those channels are opaque now... + +2012-05-07 18:51 +0000 [r365479] Richard Mudgett + + * /, tests/test_config.c: Fix type punned compiler warning in + test_config.c ........ Merged revisions 365476 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 365478 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-07 18:42 +0000 [r365477] Matthew Jordan + + * main/pbx.c, /, apps/app_voicemail.c: Support VoiceMail d() option + when extension does not exist in channel's context The VoiceMail + d([c]) option is documented to accept digits for a new extension + in context , if played during the greeting. This option works + fine if the extension being redirected to has an extension with + the same initial digit in the channel's current context. If that + digit did not happen to exist in some extension, a dialplan match + would fail and the user would not be redirected. This patch fixes + it such that if the option is used, the extensions are + matched in that context as opposed to the caller's original + context. (closes issue ASTERISK-18243) Reported by: mjordan + Tested by: mjordan Review: + https://reviewboard.asterisk.org/r/1892 ........ Merged revisions + 365474 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 365475 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-04 22:17 +0000 [r365400] Kinsey Moore + + * /, channels/chan_sip.c, funcs/func_aes.c, main/features.c, + apps/app_followme.c, channels/chan_iax2.c, + channels/sip/config_parser.c, pbx/pbx_config.c, + apps/app_chanspy.c, apps/app_stack.c, main/config.c, + apps/app_voicemail.c: Fix many issues from the NULL_RETURNS + Coverity report Most of the changes here are trivial NULL checks. + There are a couple optimizations to remove the need to check for + NULL and outboundproxy parsing in chan_sip.c was rewritten to + avoid use of strtok. Additionally, a bug was found and fixed with + the parsing of outboundproxy when "outboundproxy=," was set. + (Closes issue ASTERISK-19654) ........ Merged revisions 365398 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 365399 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-04 17:38 +0000 [r365356] Richard Mudgett + + * channels/chan_local.c, /: Fix local channel chains optimizing + themselves out of a call. * Made chan_local.c:check_bridge() + check the return value of ast_channel_masquerade(). In long + chains of local channels, the masquerade occasionally fails to + get setup because there is another masquerade already setup on an + adjacent local channel in the chain. * Made the outgoing local + channel (the ;2 channel) flush one voice or video frame per + optimization attempt. * Made sure that the outgoing local channel + also does not have any frames in its queue before the masquerade. + * Made do the masquerade immediately to minimize the chance that + the outgoing channel queue does not get any new frames added and + thus unconditionally flushed. * Made block indication -1 (Stop + tones) event when the local channel is going to optimize itself + out. When the call is answered, a chain of local channels pass + down a -1 indication for each bridge. This blizzard of -1 events + really slows down the optimization process. (closes issue + ASTERISK-16711) Reported by: Alec Davis Tested by: rmudgett, Alec + Davis Review: https://reviewboard.asterisk.org/r/1894/ ........ + Merged revisions 365313 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 365320 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-04 15:52 +0000 [r365300] Mark Michelson + + * res/res_rtp_asterisk.c, /: Fix core FINDING 2, FINDING 3, and + FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report. + These three all are in RTP code that attempts to print the number + of sequence number cycles in an RTCP RR report. The code was + masking out the upper 16 bits and then shifting the number right + by 16 bits. This led to an all zero result in all cases. The fix + is to do the shift without the bit masking. (issue + ASTERISK-19649) ........ Merged revisions 365298 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 365299 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-03 19:36 +0000 [r365248] Michael L. Young + + * tests/test_security_events.c: Update security events unit tests + The security events framework API was changed in Asterisk 10 but + the unit tests were not updated at the same time. This patch does + the following: * Adds two more security events that were added to + the API * Add challenge, received_challenge and received_hash in + the inval_password security event unit test (Closes issue + ASTERISK-19760) Reported by: Michael L. Young Tested by: Michael + L. Young Patches: issue-asterisk-19760-trunk.diff uploaded by + Michael L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/1897/ + +2012-05-03 18:43 +0000 [r365213] Sean Bright + + * CHANGES: Update documentation references in CHANGES to reflect + the correct pages on the wiki. The current CHANGES file refers to + doc/ in many places and those files no longer exist. + +2012-05-03 15:05 +0000 [r365161] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, /, + addons/ooh323c/src/h323/H323-MESSAGES.h, + addons/ooh323c/src/h323/H323-MESSAGESEnc.c: Fix warning of + Coverity Static analysis, change H225ProtocolIdentifier from + value to pointer per functions that use this. (close issue + ASTERISK-19670) Reported by: Matt Jordan Patches: + ASTERISK-19670.patch (License #5415) ........ Merged revisions + 365159 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 365160 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-03 14:47 +0000 [r365158] Sean Bright + + * apps/app_externalivr.c, CHANGES: Add IPv6 support to ExternalIVR. + Review: https://reviewboard.asterisk.org/r/1896/ + +2012-05-03 14:35 +0000 [r365157] Alexandr Anikin + + * /, addons/ooh323c/src/ooq931.c: Fix coverity static analysis + warning, allocate full ie structure instead of without data + buffer (close issue ASTERISK-19674) Reported by: Matt Jordan + Patches: ASTERISK-19674.patch (License #5415) ........ Merged + revisions 365143 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 365155 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-02 17:43 +0000 [r365084] Terry Wilson + + * channels/chan_local.c, /, main/cel.c: Multiple revisions + 365006,365068 ........ r365006 | twilson | 2012-05-02 10:49:03 + -0500 (Wed, 02 May 2012) | 12 lines Fix a CEL LINKEDID_END race + and local channel linkedids This patch has the ;2 channel inherit + the linkedid of the ;1 channel and fixes the race condition by no + longer scanning the channel list for "other" channels with the + same linkedid. Instead, cel.c has an ao2 container of linkedid + strings and uses the refcount of the string as a counter of how + many channels with the linkedid exist. Not only does this + eliminate the race condition, but it also allows us to look up + the linkedid by the hashed key instead of traversing the entire + channel list. Review: https://reviewboard.asterisk.org/r/1895/ + ........ r365068 | twilson | 2012-05-02 12:02:39 -0500 (Wed, 02 + May 2012) | 11 lines Don't leak a ref if out of memory and can't + link the linkedid If the ao2_link fails, we are most likely out + of memory and bad things are going to happen. Before those bad + things happen, make sure to clean up the linkedid references. + This patch also adds a comment explaining why linkedid can't be + passed to both local channel allocations and combines two ao2_ref + calls into 1. Review: https://reviewboard.asterisk.org/r/1895/ + ........ Merged revisions 365006,365068 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 365083 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-02 15:59 +0000 [r365011] Jason Parker + + * channels/chan_sip.c: Save the address on which a MESSAGE was + received, so it can be used in MESSAGE() This is useful in cases + where chan_sip may be listening on multiple addresses. + +2012-05-02 02:51 +0000 [r364966] Matthew Jordan + + * /, main/audiohook.c: Only log a failure to get read/write samples + from factories if it didn't happen In audiohook_read_frame_both, + anytime samples are obtained from the read/write factories a + debug statement is logged stating that samples were not obtained + from the factories. This statement used to only occur if + option_debug was turned on and no samples were obtained; in some + refactoring when the option_debug statement was removed, the + "else" clause was removed as well. This patch makes it so that + those debug log statements only occur if the condition leading up + to them actually happened. ........ Merged revisions 364965 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-01 23:23 +0000 [r364915] Mark Michelson + + * channels/chan_sip.c: Remove a function that has been marked + unused since Asterisk 1.6.0. The reason I'm removing this is that + Coverity reported a STRAY_SEMICOLON issue here. Since the + function has been unused for so long, I just elected to remove it + altogether. (closes issue ASTERISK-19660) + +2012-05-01 23:21 +0000 [r364910] Richard Mudgett + + * /, main/astobj2.c: Fixed __ao2_ref() validating user_data twice. + (closes issue ASTERISK-19755) Reported by: Gunther Kelleter + Patches: ao2_ref.patch (license #6372) patch uploaded by Gunther + Kelleter ........ Merged revisions 364902 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364903 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-01 23:11 +0000 [r364901] Mark Michelson + + * /, funcs/func_volume.c: Fix Coverity-reported ARRAY_VS_SINGLETON + error. As it turned out, this wasn't a huge deal. We were calling + ast_app_parse_options() for a set of options of which none took + arguments. The proper thing to do for this case is to pass NULL + for the "args" parameter here. We were instead passing a + seemingly-randomly chosen char * from the function. While this + would never get written to, you can rest assured things would + have gotten bad had new options (which took arguments) been added + to func_volume. (closes issue ASTERISK-19656) ........ Merged + revisions 364899 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364900 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-01 22:00 +0000 [r364846] Richard Mudgett + + * channels/chan_local.c, /: * Fix error path resouce leak in + local_request(). * Restructure local_request() to reduce + indentation. ........ Merged revisions 364840 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364845 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-01 21:49 +0000 [r364844] Jason Parker + + * main/manager.c, /: Prevent a potential crash when using manager + hooks. Found by me while poking at DPMA-127. ........ Merged + revisions 364841 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364842 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-01 19:10 +0000 [r364788] Kinsey Moore + + * /, apps/app_confbridge.c: Play conf-placeintoconf message to the + correct channel Correct the code in app_confbridge to play the + conf-placeintoconf message to the marked user entering the bridge + instead of to the conference while the marked user hears silence. + (closes issue ASTERISK-19641) Reported-by: Mark A Walters + ........ Merged revisions 364786 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364787 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-01 18:29 +0000 [r364785] Jonathan Rose + + * /, main/app.c: Fix bad check in voicemail functions for + ast_inboxcount2_func Check looks for ast_inboxcount_func instead + of ast_inboxcount2_func on ast_inboxcount2_func calls. (closes + issue ASTERISK-19718) Reported by: Corey Farrell Patches: + ast_app_inboxcount2-null-refcheck.patch uploaded by Corey Farrell + (license 5909) ........ Merged revisions 364769 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364777 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-30 19:51 +0000 [r364708] Mark Michelson + + * /, channels/chan_sip.c: Revert revision 360862. Revision 360862 + was intended to improve identities sent in dialog-info NOTIFY + requests. Some users reported that hint became broken once this + was done. It's not clear exactly what part of the patch has + caused this regression, but broken hints are bad. For now, this + revision is being reverted so that the next releases of Asterisk + do not have bad behavior in them. The original reported issue + will have to be fixed differently in the next version of + Asterisk. (issue ASTERISK-16735) ........ Merged revisions 364706 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 364707 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-30 17:17 +0000 [r364654] Mark Murawki + + * /, main/logger.c: Merged revisions 364635 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r364635 | markm | 2012-04-30 11:51:12 -0400 (Mon, 30 Apr 2012) | + 10 lines Sanatize result from bfd_find_nearest_line + (BETTER_BACKTRACES) bfd_find_nearest_line can possibly set file + to null resulting in a crash when strrchr(file) runs (closes + issue ASTERISK-19815) Reported by Mark Murawski Tested by Mark + Murawski ........ ........ Merged revisions 364650 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-30 16:59 +0000 [r364652] Alexandr Anikin + + * /, addons/ooh323cDriver.c: Fix use freed pointer in return value + from call thread (issue ASTERISK-19663) Reported by: Matt Jordan + Patches: ASTERISK-19663-ooh323.patch (License #5415) ........ + Merged revisions 364649 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364651 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-29 19:50 +0000 [r364580] Matthew Jordan + + * formats/format_ilbc.c, /, formats/format_sln.c, + formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, + formats/format_g723.c, formats/format_h263.c, + formats/format_h264.c, formats/format_wav_gsm.c, + formats/format_siren14.c, formats/format_gsm.c, + formats/format_g719.c, formats/format_siren7.c, + formats/format_g729.c: Fix error that caused truncate operations + to fail Another very inappropriate placement of a ')' (again + introduced in r362151) caused the various truncate operations to + attempt to truncate the sound file at a position of '0'. (issue + ASTERISK-19655) Reported by: Matt Jordan (issue ASTERISK-19810) + Reported by: colbec ........ Merged revisions 364578 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364579 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-29 02:23 +0000 [r364537] Michael L. Young + + * /, apps/confbridge/conf_config_parser.c: Fix configuring custom + sound_leader_has_left in confbridge.conf The configuration option + to specify a custom sound_leader_has_left file for a conference + bridge was not being parsed. This patch fixes it so that a custom + sound file will now be used. (closes issue ASTERISK-19771) + Reported by: Pawel Kuzak Tested by: Pawel Kuzak, Michael L. Young + Patches: leaderhasleft_sound.dpatch uploaded by Pawel Kuzak + (license 6380) Review: https://reviewboard.asterisk.org/r/1884/ + ........ Merged revisions 364536 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-28 20:24 +0000 [r364500] Joshua Colp + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/sip/include/sip.h: Add support for lightweight NAT + keepalive. If enabled using the keepalive option in sip.conf a + small packet will be sent at a regular interval to keep the NAT + mapping open. This is lightweight as the remote side does not + need to parse and handle a SIP message. (closes issue AST-783) + Review: https://reviewboard.asterisk.org/r/1756/ + +2012-04-28 01:33 +0000 [r364437-364462] Russell Bryant + + * main/md5.c: md5: supress some compiler warnings. md5.c: In + function ‘MD5Final’: md5.c:154:2: error: dereferencing + type-punned pointer will break strict-aliasing rules + [-Werror=strict-aliasing] md5.c:155:2: error: dereferencing + type-punned pointer will break strict-aliasing rules + [-Werror=strict-aliasing] There is an md5 unit test and it still + passes. + + * configure, include/asterisk/autoconfig.h.in, res/res_corosync.c, + configure.ac: res_corosync: Fix build against corosync 2.0. + + * apps/app_minivm.c: app_minivm: Fix a couple compiler warnings. + The warnings were about argv[0] being used uninitialized, which + is correct. Just remove setting username to this value, since + username is set again before it actually gets used. + + * main/features.c, CHANGES: features: Add FEATURE() and + FEATUREMAP() functions. Add two new dialplan functions: FEATURE() + and FEATUREMAP(). FEATURE() lets you set some of the + configuration options from the [general] section of features.conf + on a per-channel basis. FEATUREMAP() lets you customize the key + sequence used to activate built-in features, such as blindxfer, + and automon. See the built-in documentation for details. Review: + https://reviewboard.asterisk.org/r/1871/ + +2012-04-28 00:31 +0000 [r364436] Richard Mudgett + + * apps/app_dial.c, CHANGES: PreDial - Ability to run dialplan on + callee and caller channels before Dial. Thanks to Mark Murawski + for the initial patch and feature definition. (closes issue + ASTERISK-19548) Reported by: Mark Murawski Review: + https://reviewboard.asterisk.org/r/1878/ Review: + https://reviewboard.asterisk.org/r/1229/ + +2012-04-27 22:54 +0000 [r364397] Terry Wilson + + * /, tests/test_config.c (added), main/config.c: Multiple revisions + 364365,364369 ........ r364365 | twilson | 2012-04-27 17:31:01 + -0500 (Fri, 27 Apr 2012) | 11 lines Fix ast_parse_arg numeric + type range checking and add tests ast_parse_arg wasn't checking + for strto* parse errors or limiting the results by the actual + range of the numeric types. This patch fixes that and adds unit + tests as well. Review: https://reviewboard.asterisk.org/r/1879/ + ........ Merged revisions 364340 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r364369 | twilson | 2012-04-27 17:33:10 -0500 (Fri, 27 Apr 2012) + | 2 lines Add missing test_config.c ........ Merged revisions + 364365,364369 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-27 22:11 +0000 [r364343] Mark Michelson + + * /, channels/chan_sip.c: Don't attempt to make use of the + dynamic_exclude_static ACL if DNS lookup fails. (closes issue + ASTERISK-18321) Reported by Dan Lukes Patches: + ASTERISK-18321.patch by Mark Michelson (license #5049) ........ + Merged revisions 364341 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364342 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-27 19:30 +0000 [r364287] Matthew Jordan + + * /, include/asterisk/time.h: Prevent overflow in calculation in + ast_tvdiff_ms on 32-bit machines The method ast_tvdiff_ms + attempts to calculate the difference, in milliseconds, between + two timeval structs, and return the difference in a 64-bit + integer. Unfortunately, it assumes that the long tv_sec/tv_usec + members in the timeval struct are large enough to hold the + calculated values before it returns. On 64-bit machines, this + might be the case, as a long may be 64-bits. On 32-bit machines, + however, a long may be less (32-bits), in which case, the + calculation can overflow. This overflow caused significant + problems in MixMonitor, which uses the method to determine if an + audio factory, which has not presented audio to an audiohook, is + merely late in providing said audio or will never provide audio. + In an overflow situation, the audiohook would incorrectly + determine that an audio factory that will never provide audio is + merely late instead. This led to situations where a MixMonitor + never recorded any audio. Note that this happened most frequently + when that MixMonitor was started by the ConfBridge application + itself, or when the MixMonitor was attached to a Local channel. + (issue ASTERISK-19497) Reported by: Ben Klang Tested by: Ben + Klang Patches: 32-bit-time-overflow-10-2012-04-26.diff (license + #6283) by mjordan (closes issue ASTERISK-19727) Reported by: Mark + Murawski Tested by: Michael L. Young Patches: + 32-bit-time-overflow-2012-04-27.diff (license #6283) by mjordan) + (closes issue ASTERISK-19471) Reported by: feyfre Tested by: + feyfre (issue ASTERISK-19426) Reported by: Johan Wilfer Review: + https://reviewboard.asterisk.org/r/1889/ ........ Merged + revisions 364277 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364285 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-27 18:59 +0000 [r364260] Kinsey Moore + + * /, channels/chan_sip.c: Allow SIP pvts involved in Replaces + transfers to fall out of reference sooner Unref the SIP pvt + stored in the refer structure as soon as it is no longer needed + so that the pvt and associated file descriptors can be freed + sooner. This change makes a reference decrement unnecessary in + code that handles SIP BYE/Also transfers which should not touch + the reference anyway. (Closes issue ASTERISK-19579) Reported by: + Maciej Krajewski Tested by: Maciej Krajewski ........ Merged + revisions 364258 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364259 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-27 14:45 +0000 [r364205] Matthew Jordan + + * /, channels/chan_sip.c: Allow for reloading SRTP crypto keys + within the same SIP dialog As a continuation of the patch in + r356604, which allowed for the reloading of SRTP keys in + re-INVITE transfer scenarios, this patch addresses the more + common case where a new key is requested within the context of a + current SIP dialog. This can occur, for example, when certain + phones request a SIP hold. Previously, once a dialog was + associated with an SRTP object, any subsequent attempt to process + crypto keys in any SDP offer - either the current one or a new + offer in a new SIP request - were ignored. This patch changes + this behavior to only ignore subsequent crypto keys within the + current SDP offer, but allows future SDP offers to change the + keys. (issue ASTERISK-19253) Reported by: Thomas Arimont Tested + by: Thomas Arimont Review: + https://reviewboard.asteriskorg/r/1885/ ........ Merged revisions + 364203 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 364204 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-27 12:58 +0000 [r364164] Stefan Schmidt + + * res/res_calendar_icalendar.c, /, res/res_calendar_caldav.c: fix a + wrong behavior of alarm timezones in caldav and icalendar when an + alarm doesnt use utc. This change uses the same timezone from the + start time. ........ Merged revisions 364163 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-26 21:11 +0000 [r364082-364110] Richard Mudgett + + * /, apps/app_directed_pickup.c: Update Pickup application + documentation. (With feeling this time.) ........ Merged + revisions 364108 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364109 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/features.c: Fix DTMF atxfer running h exten after the + wrong bridge ends. When party B does an attended transfer of + party A to party C, the attending bridge between party B and C + should not be running an h exten when the bridge ends. Running an + h exten now sets a softhangup flag to ensure that an AGI will run + in dead AGI mode. * Set the AST_FLAG_BRIDGE_HANGUP_DONT on the + party B channel for the attending bridge between party B and C. + (closes issue AST-870) (closes issue ASTERISK-19717) Reported by: + Mario (closes issue ASTERISK-19633) Reported by: Andrey Solovyev + Patches: jira_asterisk_19633_v1.8.patch (license #5621) patch + uploaded by rmudgett Tested by: rmudgett, Andrey Solovyev, Mario + ........ Merged revisions 364060 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364065 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-26 19:33 +0000 [r364048] Terry Wilson + + * /, main/asterisk.c: Add more constness to the end_buf pointer in + the netconsole issue ASTERISK-18308 Review: + https://reviewboard.asterisk.org/r/1876/ ........ Merged + revisions 364046 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 364047 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-26 13:59 +0000 [r363989] Olle Johansson + + * apps/app_queue.c: Code formatting fixes. + +2012-04-26 13:31 +0000 [r363988] Kinsey Moore + + * /, channels/chan_sip.c: Fix reference leaks involving SIP + Replaces transfers The reference held for SIP blind transfers + using the Replaces header in an INVITE was never freed on success + and also failed to be freed in some error conditions. This caused + a file descriptor leak since the RTP structures in use at the + time of the transfer were never freed. This reference leak and + another relating to subscriptions in the same code path have now + been corrected. (closes issue ASTERISK-19579) ........ Merged + revisions 363986 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 363987 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-26 09:48 +0000 [r363936] Alec L Davis + + * /, channels/chan_sip.c: chan_sip: [general] maxforwards, not + checked for a value greater than 255 The peer maxforwards is + checked for both '< 1' and '> 255', but the default 'maxforwards' + in the [general] section is only checked for '< 1' alecdavis + (license 585) Reported by: alecdavis Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/1888/ ........ Merged + revisions 363934 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 363935 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-26 03:12 +0000 [r363689-363877] Richard Mudgett + + * /, apps/app_directed_pickup.c: Update Pickup application + documentation. (Even better) ........ Merged revisions 363875 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 363876 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_directed_pickup.c: * Put more information in + pickup_exec() LOG_NOTICE. * Delay duplicating a string on the + stack in pickup_exec(). + + * /, apps/app_directed_pickup.c: Update Pickup application + documentation. ........ Merged revisions 363788 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 363789 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_dahdi.c, /, channels/sig_pri.c: Make + DAHDISendCallreroutingFacility wait 5 seconds for a reply before + disconnecting the call. Some switches may not handle the + call-deflection/call-rerouting message if the call is + disconnected too soon after being sent. Asteisk was not waiting + for any reply before disconnecting the call. * Added a 5 second + delay before disconnecting the call to wait for a potential + response if the peer does not disconnect first. (closes issue + ASTERISK-19708) Reported by: mehdi Shirazi Patches: + jira_asterisk_19708_v1.8.patch (license #5621) patch uploaded by + rmudgett Tested by: rmudgett ........ Merged revisions 363730 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 363734 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: + Clear ISDN channel resetting state if the peer continues to use + it. Some ISDN switches occasionally fail to send a RESTART + ACKNOWLEDGE in response to a RESTART request. * Made the second + SETUP received after sending a RESTART request clear the channel + resetting state as if the peer had sent the expected RESTART + ACKNOWLEDGE before continuing to process the SETUP. The peer may + not be sending the expected RESTART ACKNOWLEDGE. (issue + ASTERISK-19608) (issue AST-844) (issue AST-815) Patches: + jira_ast_815_v1.8.patch (license #5621) patch uploaded by + rmudgett (modified) ........ Merged revisions 363687 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 363688 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-25 13:57 +0000 [r363480-363637] Olle Johansson + + * apps/app_queue.c: Add documentation Thanks Tilghman! + + * apps/app_queue.c: Formatting changes only + + * apps/app_followme.c, apps/app_queue.c: Use the DEFINED value for + musicclass length. For some reason, features.c has it's own + definition. Should propably be fixed too. + + * main/channel.c, configs/asterisk.conf.sample, CHANGES, + include/asterisk/options.h, main/asterisk.c: Make it possible to + change the minimum DTMF duration in asterisk.conf Asterisk has a + setting for the minimum allowed DTMF. If we get shorter DTMF + tones, these will be changed to the minimum on the outbound call + leg. (closes issue ASTERISK-19772) Review: + https://reviewboard.asterisk.org/r/1882/ Reported by: oej Tested + by: oej Patches by: oej Thanks to the reviewers. 1.8 branch for + this patch: agave-dtmf-duration-asterisk-conf-1.8 + + * main/say.c: Formatting fixes Developer guidelines are important. + + * main/channel.c: Formatting fixes Found a small amount of curly + brackets in my hotel room here in Denmark. I hereby donate them + to the Asterisk project. + +2012-04-25 01:26 +0000 [r363377-363430] Richard Mudgett + + * /, main/features.c: Fix recalled party B feature flags for a + failed DTMF atxfer. 1) B calls A with Dial option T 2) B DTMF + atxfer to C 3) B hangs up 4) C does not answer 5) B is called + back 6) B answers 7) B cannot initiate transfers anymore * Add + dial features datastore to recalled party B channel that is a + copy of the original party B channel's dial features datastore. * + Extracted add_features_datastore() from + add_features_datastores(). * Renamed struct ast_dial_features + features_caller and features_callee members to my_features and + peer_features respectively. These better names eliminate the need + for some explanatory comments. * Simplified code accessing the + struct ast_dial_features datastore. (closes issue ASTERISK-19383) + Reported by: lgfsantos ........ Merged revisions 363428 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 363429 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/features.c: Hangup affected channel in error paths of + bridge_call_thread(). ........ Merged revisions 363375 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 363376 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-24 17:52 +0000 [r363335] Terry Wilson + + * /, main/asterisk.c: OpenBSD doesn't have rawmemchr, use strchr + (closes issue ASTERISK-19758) Reported by: Barry Miller Tested + by: Terry Wilson Patches: 362758-diff uploaded by Barry Miller + (license 5434) ........ Merged revisions 362868 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362869 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-23 17:05 +0000 [r363269] Richard Mudgett + + * apps/app_dial.c, apps/app_queue.c: Make app_dial and app_queue + use new macro and gosub calls. * Simplify some code in app_dial + and app_queue by calling ast_app_exec_macro() and + ast_app_exec_sub(). * Fix minor locking issue in app_dial for + post-answer macro/gosub MACRO/GOSUB_RESULT=GOTO: handling. + +2012-04-23 16:08 +0000 [r363215] Tilghman Lesher + + * /, main/astfd.c: On some platforms, O_RDONLY is not a flag to be + checked, but merely the absence of O_RDWR and O_WRONLY. The POSIX + specification does not mandate how these 3 flags must be + specified, only that one of the three must be specified in every + call. ........ Merged revisions 363209 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 363212 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-23 14:48 +0000 [r363159] Jonathan Rose + + * main/manager.c, /: AST-2012-004: Fix an error that allows AMI + users to run shell commands sans authorization. As detailed in + the advisory, AMI users without write authorization for SYSTEM + class AMI actions were able to run system commands by going + through other AMI commands which did not require that + authorization. Specifically, GetVar and Status allowed users to + do this by setting their variable/s options to the SHELL or EVAL + functions. Also, within 1.8, 10, and trunk there was a similar + flaw with the Originate action that allowed users with originate + permission to run MixMonitor and supply a shell command in the + Data argument. That flaw is fixed in those versions of this + patch. (closes issue ASTERISK-17465) Reported By: David Woolley + Patches: 162_ami_readfunc_security_r2.diff uploaded by jrose + (license 6182) 18_ami_readfunc_security_r2.diff uploaded by jrose + (license 6182) 10_ami_readfunc_security_r2.diff uploaded by jrose + (license 6182) ........ Merged revisions 363117 from + http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ + Merged revisions 363141 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 363156 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-23 14:10 +0000 [r363105-363108] Matthew Jordan + + * /, channels/chan_sip.c: AST-2012-006: Fix crash in UPDATE + handling when no channel owner exists If Asterisk receives a SIP + UPDATE request after a call has been terminated and the channel + has been destroyed but before the SIP dialog has been destroyed, + a condition exists where a connected line update would be + attempted on a non-existing channel. This would cause Asterisk to + crash. The patch resolves this by first ensuring that the SIP + dialog has an owning channel before attempting a connected line + update. If an UPDATE request is received and no channel is + associated with the dialog, a 481 response is sent. (closes issue + ASTERISK-19770) Reported by: Thomas Arimont Tested by: Matt + Jordan Patches: ASTERISK-19278-2012-04-16.diff uploaded by Matt + Jordan (license 6283) ........ Merged revisions 363106 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 363107 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_skinny.c: AST-2012-005: Fix remotely exploitable + heap overflow in keypad button handling When handling a keypad + button message event, the received digit is placed into a fixed + length buffer that acts as a queue. When a new message event is + received, the length of that buffer is not checked before placing + the new digit on the end of the queue. The situation exists where + sufficient keypad button message events would occur that would + cause the buffer to be overrun. This patch explicitly checks that + there is sufficient room in the buffer before appending a new + digit. (closes issue ASTERISK-19592) Reported by: Russell Bryant + ........ Merged revisions 363100 from + http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ + Merged revisions 363102 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 363103 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-21 11:45 +0000 [r363045-363046] Russell Bryant + + * res/res_corosync.c: res_corosync: Recover if corosync gets + restarted. If corosync gets restarted while Asterisk is running, + automatically recover. + + * res/res_corosync.c: res_corosync: reimplement "corosync show + members" command. Reimplement the "corosync show members" CLI + command using a CPG iterator instead of the cpg_membership_get + API call. This will also show all CPG members, including those in + groups other than 'asterisk', which may be useful at some point + for debugging purposes. + +2012-04-21 01:46 +0000 [r362920-362999] Richard Mudgett + + * apps/app_dial.c, /: Update app_dial M and U option GOTO return + value documentation. ........ Merged revisions 362997 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362998 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * include/asterisk/app.h, main/app.c, apps/app_stack.c: Fix + connected-line/redirecting interception gosubs executing more + than intended. * Redo ast_app_run_sub()/ast_app_exec_sub() to use + a known return point so execution will stop after the routine + returns there. (s@gosub_virtual_context:1) * Create + ast_app_exec_macro() and ast_app_exec_sub() to run the macro and + gosub application respectively with the parameter string already + created. + + * main/rtp_engine.c: Move debug message in + ast_rtp_instance_early_bridge_make_compatible(). Move debug + message in ast_rtp_instance_early_bridge_make_compatible() to be + output when what it states has actually happened. + +2012-04-20 16:50 +0000 [r362919] Michael L. Young + + * /, main/event.c: Add missing payload type to events API The + Security Events Framework API was changed while adding the + generation of security events in chan_sip. A payload type and + name was missed from being added to struct ie_maps. (closes issue + ASTERISK-19759) Reported by: Michael L. Young Patches: + issue-asterisk-19759.diff uploaded by Michael L. Young (license + 5026) ........ Merged revisions 362918 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-20 16:23 +0000 [r362867-362888] Richard Mudgett + + * apps/app_dial.c, channels/chan_dahdi.c, channels/chan_local.c, + channels/chan_misdn.c, main/rtp_engine.c: Use + ast_channel_lock_both() where it was inlined before. The + CHANNEL_DEADLOCK_AVOIDANCE() feature of preserving where the + channel lock was originally obtained is overkill where + ast_channel_lock_both() was inlined. + + * main/pbx.c: * Add more information to some messages in + __ast_pbx_run(). * Simplify some dialplan priority setting code + in ast_explicit_goto() because of opaquification. + +2012-04-20 14:50 +0000 [r362817] Terry Wilson + + * /, apps/app_speech_utils.c: Document Speech* apps hangup on + failure and suggest TryExec The Speech API apps return -1 on + failure, which will hang up the channel. This may not be + desirable behavior for some, but it isn't something that can be + changed without breaking people's dialplans or writing an option + to all of the Speech apps that does what TryExec already does. + This patch documents the hangup behavior of the apps, and + suggests TryExec as the solution. (closes issue AST-813) ........ + Merged revisions 362815 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362816 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-20 00:57 +0000 [r362779] Richard Mudgett + + * main/channel.c, UPGRADE.txt, include/asterisk/channel.h, CHANGES, + channels/sig_pri.c, funcs/func_callerid.c: Add original party id + and reason support. ISDN ETSI PTP and Q.SIG (And SS7 in future) + have support for reporting who was the original redirecting party + of a call. * Added support for the original redirecting party and + reason to the REDIRECTING function and the system core as well as + to the stubbed locations in sig_pri.c. Review: + https://reviewboard.asterisk.org/r/1829/ + +2012-04-19 22:01 +0000 [r362731] Walter Doekes + + * funcs/func_version.c, /: Fix documentation for + ${VERSION(ASTERISK_VERSION_NUM)}. ........ Merged revisions + 362729 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 362730 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-19 21:14 +0000 [r362682] Michael L. Young + + * /, tests/test_linkedlists.c, tests/test_poll.c: Add leading and + trailing backslashes A couple of unit tests did not have have + leading or trailing backslashes when setting their test category + resulting in a warning message being displayed. Added the + backslash where needed. ........ Merged revisions 362680 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362681 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-19 21:01 +0000 [r362679] Richard Mudgett + + * /, configs/queues.conf.sample: Update membermacro and membergosub + documentation in queues.conf.sample. ........ Merged revisions + 362677 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 362678 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-19 19:05 +0000 [r362635] Terry Wilson + + * addons/chan_ooh323.c, apps/app_alarmreceiver.c, + channels/iax2-provision.c, res/snmp/agent.c: Convert some + strncpys to ast_copy_string Review: + https://reviewboard.asterisk.org/r/1732/ + +2012-04-19 16:10 +0000 [r362588] Sean Bright + + * /, apps/app_externalivr.c: Prevent a crash in ExternalIVR when + the 'S' command is sent first. If the first command sent from an + ExternalIVR client is an 'S' command, we were blindly removing + the first element from the play list and deferencing it, even if + it was NULL. This corrects that and also locks appropriately in + one place. (issue ASTERISK-17889) Reported by: Chris Maciejewski + ........ Merged revisions 362586 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362587 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-19 14:35 +0000 [r362538] Terry Wilson + + * /, main/asterisk.c: Handle multiple commands per connection via + netconsole Asterisk would accept multiple NULL-delimited CLI + commands via the netconsole socket, but would occasionally miss a + command due to the command not being completely read into the + buffer. This patch ensures that any partial commands get moved to + the front of the read buffer, appended to, and properly sent. + (closes issue ASTERISK-18308) Review: + https://reviewboard.asterisk.org/r/1876/ ........ Merged + revisions 362536 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362537 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-19 02:40 +0000 [r362497] Matthew Jordan + + * channels/chan_unistim.c, /, main/tdd.c, main/jitterbuf.c, + apps/app_sms.c, main/stdtime/localtime.c, utils/extconf.c, + addons/chan_mobile.c, main/format_pref.c, main/asterisk.c: Fix a + variety of potential buffer overflows * chan_mobile: Fixed an + overrun where the cind_state buffer (an integer array of size 16) + would be overrun due to improper bounds checking. At worst, the + buffer can be overrun by a total of 48 bytes (assuming 4-byte + integers), which would still leave it within the allocated memory + of struct hfp. This would corrupt other elements in that struct + but not necessarily cause any further issues. * app_sms: The + array imsg is of size 250, while the array (ud) that the data is + copied into is of size 160. If the size of the inbound message is + greater then 160, up to 90 bytes could be overrun in ud. This + would corrupt the user data header (array udh) adjacent to ud. * + chan_unistim: A number of invalid memmoves are corrected. These + would move data (which may or may not be valid) into the ends of + these buffers. * asterisk: ast_console_toggle_loglevel does not + check that the console log level being set is less then or equal + to the allowed log levels of 32. * format_pref: In + ast_codec_pref_prepend, if any occurrence of the specified codec + is not found, the value used to index into the array pref->order + would be one greater then the maximum size of the array. * + jitterbuf: If the element being placed into the jitter buffer + lands in the last available slot in the jitter history buffer, + the insertion sort attempts to move the last entry in the buffer + into one slot past the maximum length of the buffer. Note that + this occurred for both the min and max jitter history buffers. * + tdd: If a read from fsk_serial returns a character that is + greater then 32, an attempt to read past one of the statically + defined arrays containing the values that character maps to would + occur. * localtime: struct ast_time and tm are not the same size + - ast_time is larger, although it contains the elements of tm + within it in the same layout. Hence, when using memcpy to copy + the contents of tm into ast_time, the size of tm should be used, + as opposed to the size of ast_time. * extconf: this treats + ast_timing's minmask array as if it had a length of 48, when it + has defined the size of the array as 24. pbx.h defines minmask as + having a size of 48. (issue ASTERISK-19668) Reported by: Matt + Jordan ........ Merged revisions 362485 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362496 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-18 17:03 +0000 [r362432] Michael L. Young + + * tests/test_security_events.c: Fix building security events test + The Security Events Framework API changed in trunk to support + IPv6. This broke the building of the security events test which + was based around IPv4. This patches fixes the build by changing + the test to conform to the new changes. (related to issue + ASTERISK-19447) Review: https://reviewboard.asterisk.org/r/1874/ + +2012-04-18 16:41 +0000 [r362430] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, /, channels/sig_pri.c: Add + ability to ignore layer 1 alarms for BRI PTMP lines. Several + telcos bring the BRI PTMP layer 1 down when the line is idle. + When layer 1 goes down, Asterisk cannot make outgoing calls. + Incoming calls could fail as well because the alarm processing is + handled by a different code path than the Q.931 messages. * Add + the layer1_presence configuration option to ignore layer 1 alarms + when the telco brings layer 1 down. This option can be configured + by span while the similar DAHDI driver teignorered=1 option is + system wide. This option unlike layer2_persistence does not + require libpri v1.4.13 or newer. Related to JIRA AST-598 JIRA + ABE-2845 ........ Merged revisions 362428 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362429 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-17 21:23 +0000 [r362365-362380] Matthew Jordan + + * /, main/format_pref.c: Handle case where an unknown format is + used to get the preferred codec size In ast_codec_pref_getsize, + if an unknown format is passed to the method, no preferred codec + will be selected and a negative number will be used to index into + the format list. The method now logs an unknown format as a + warning, and returns an empty format list. (issue ASTERISK-19655) + Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/1863/ ........ Merged + revisions 362377 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * res/res_rtp_asterisk.c, /, res/res_agi.c, res/res_musiconhold.c: + Fix places in resources where a negative return value could + impact execution This patch addresses a number of modules in + resources that did not handle the negative return value from + function calls adequately. This includes: * res_agi.c: if the + result of the read function is a negative number, indicating some + failure, the result would instead be treated as the number of + bytes read. This patch now treats negative results in the same + manner as an end of file condition, with the exception that it + also logs the error code indicated by the return. * + res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor + to srcfd, and instead assigns a negative value, that file + descriptor could later be passed to functions that require a + valid file descriptor. If spawn_mp3 fails, we now immediately + retry instead of continuing in the logic. * res_rtp_asterisk.c: + if no codec can be matched between two RTP instances in a peer to + peer bridge, we immediately return instead of attempting to use + the codec payload type as an index to determine the appropriate + negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/1863/ ........ Merged + revisions 362362 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362364 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-17 21:10 +0000 [r362363] Jonathan Rose + + * res/res_config_curl.c, res/res_config_pgsql.c, + res/res_config_odbc.c, /: Make use of va_args more appropriate to + form in various res_config modules plus utils. A number of + va_copy operations weren't matched with a corresponding va_end in + res_config_odbc. Also, there was a potential for va_end to be + invoked twice on the same va_arg in utils, which would mean + invoking va_end on an undefined variable... which is bad. va_end + is removed from various functions in config_pgsql and config_curl + since they aren't making their own copy. The invokers of those + functions are responsible for calling va_end on them. (issue + ASTERISK-19451) Reported by: Walter Doekes Review: + https://reviewboard.asterisk.org/r/1848/ ........ Merged + revisions 362354 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362357 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-17 21:08 +0000 [r362358-362361] Matthew Jordan + + * main/manager.c, /, main/asterisk.c: Fix places in main where a + negative return value could impact execution This patch addresses + a number of modules in main that did not handle the negative + return value from function calls adequately, or were not + sufficiently clear that the conditions leading to improper + handling of the return values could not occur. This includes: * + asterisk.c: A negative return value from the read function would + be used directly as an index into a buffer. We now check for + success of the read function prior to using its result as an + index. * manager.c: Check for failures in mkstemp and lseek when + handling the temporary file created for processing data returned + from a CLI command in action_command. Also check that the result + of an lseek is sanitized prior to using it as the size of a + memory map to allocate. (issue ASTERISK-19655) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ + Merged revisions 362359 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362360 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, funcs/func_env.c: Fix places where a negative return from + ftello could be used as invalid input In a variety of locations + in both reading and writing a file, the result from the C library + function ftello is used as input to other functions. For the + parameters and functions in question, a negative value is invalid + input. This patch checks the return value from the ftello + function to determine if we were able to determine the current + position in the file stream and, if not, fail gracefully. (issue + ASTERISK-19655) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/1863/ ........ Merged + revisions 362355 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362356 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-17 18:57 +0000 [r362307] Walter Doekes + + * channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c, + funcs/func_env.c, res/res_phoneprov.c, channels/chan_gtalk.c, + cdr/cdr_pgsql.c, res/res_http_post.c, res/res_musiconhold.c, + res/res_jabber.c, res/res_format_attr_celt.c, + channels/chan_dahdi.c, funcs/func_groupcount.c, + apps/app_osplookup.c, funcs/func_odbc.c, main/ast_expr2f.c, + apps/app_minivm.c, channels/chan_alsa.c, codecs/codec_resample.c, + formats/format_h264.c, res/res_format_attr_silk.c, + res/res_config_ldap.c, main/ast_expr2.fl, + res/res_config_sqlite3.c, channels/chan_sip.c, + channels/vcodecs.c, codecs/codec_g726.c, main/data.c, + res/res_corosync.c, channels/chan_h323.c, codecs/codec_dahdi.c, + funcs/func_callerid.c, main/asterisk.c, res/res_odbc.c: Avoid + cppcheck warnings; removing unused vars and a bit of cleanup. + Patch by: junky Review: https://reviewboard.asterisk.org/r/1743/ + +2012-04-17 18:29 +0000 [r362306] Matthew Jordan + + * /, formats/format_sln.c, formats/format_vox.c, + formats/format_wav.c, formats/format_pcm.c, + formats/format_wav_gsm.c, formats/format_siren14.c, + formats/format_gsm.c, formats/format_g719.c, + formats/format_siren7.c: Fix error that caused seek format + operations to set max file size to '1' or '0' A very + inappropriate placement of a ')' (introduced in r362151) caused + the maximum size of a file to be set as the result of a + comparison operation, as opposed to the result of the ftello + operation. This resulted in seeking being restricted to the + beginning of the file, or 1 byte into the file. Thanks to the + Asterisk Test Suite for properly freaking out about this on at + least one test. (issue ASTERISK-19655) Reported by: Matt Jordan + ........ Merged revisions 362304 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362305 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-17 15:00 +0000 [r362266] Michael L. Young + + * /, channels/chan_sip.c: Turn off warning message when bind + address is set to any. When a bind address is set to an ANY + address (udpbindport=::), a warning message is displayed stating + that "Address remapping activated in sip.conf but we're using + IPv6, which doesn't need it. Please remove 'localnet' and/or + 'externaddr' settings." But if one is running dual stack, we + shouldn't be told to turn those settings off. This patch checks + if the bind address is an ANY address or not. The warning message + will now only be displayed if the bind address is NOT an ANY + address and IPv6 is being used. Also, updated the copyright year. + (closes issue ASTERISK-19456) Reported by: Michael L. Young + Tested by: Michael L. Young Patches: chan_sip_ipv6_message.diff + uploaded by Michael L. Young (license 5026) ........ Merged + revisions 362253 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362264 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-16 21:58 +0000 [r362203-362206] Matthew Jordan + + * channels/chan_dahdi.c, /, channels/chan_agent.c: Fix negative + return handling in channel drivers In chan_agent, while handling + a channel indicate, the agent channel driver must obtain a lock + on both the agent channel, as well as the channel the agent + channel is using. To do so, it attempts to lock the other channel + first, then unlock the agent channel which is locked prior to + entry into the indicate handler. If this unlock fails with a + negative return value, which can occur if the object passed to + agent_indicate is an invalid ao2 object or is NULL, the return + value is passed directly to strerror, which can only accept + positive integer values. In chan_dahdi, the return value of + dahdi_get_index is used to directly index into the sub-channel + array. If dahd_get_index returns a negative value, it would use + that value to index into the array, which could cause an invalid + memory access. If dahdi_get_index returns a negative number, we + now default to SUB_REAL. (issue ASTERISK-19655) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/1863/ ........ + Merged revisions 362204 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362205 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_voicemail.c: Fix handling of negative return code + when storing voicemails in ODBC storage When storing a voicemail + message using an ODBC connection to a database, the voicemail + message is first stored on disk. The sound file associated with + the message is read into memory before being transmitted to the + database. When this occurs, a failure in the C library's lseek + function would cause a negative value to be passed to the mmap as + the size of the memory map to create. This would almost certainly + cause the creation of the memory map to fail, resulting in the + message being lost. (issue ASTERISK-19655) Reported by: Matt + Jordan Review: https://reviewboard.asterisk.org/r/1863 ........ + Merged revisions 362201 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362202 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-16 21:20 +0000 [r362200] Michael L. Young + + * main/manager.c, main/security_events.c, + channels/sip/security_events.c, CHANGES, + include/asterisk/security_events_defs.h: Add IPv6 address support + to security events framework. The current Security Events + Framework API only supports IPv4 when it comes to generating + security events. This patch does the following: * Changes the + Security Events Framework API to support IPV6 and updates the + components that use this API. * Eliminates an error message that + was being generated since the current implementation was treating + an IPv6 socket address as if it was IPv4. * Some copyright dates + were updated on files touched by this patch. (closes issue + ASTERISK-19447) Reported by: Michael L. Young Tested by: Michael + L. Young Patches: security_events_ipv6v3.diff uploaded by Michael + L. Young (license 5026) Review: + https://reviewboard.asterisk.org/r/1777/ + +2012-04-16 20:17 +0000 [r362153] Matthew Jordan + + * formats/format_ilbc.c, /, formats/format_sln.c, + formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, + formats/format_g723.c, formats/format_h263.c, + formats/format_h264.c, formats/format_wav_gsm.c, + formats/format_siren14.c, formats/format_gsm.c, + formats/format_g719.c, formats/format_siren7.c, + formats/format_g729.c: Check for IO stream failures in various + format's truncate/seek operations For the formats that support + seek and/or truncate operations, many of the C library calls used + to determine or set the current position indicator in the file + stream were not being checked. In some situations, if an error + occurred, a negative value would be returned from the library + call. This could then be interpreted inappropriately as + positional data. This patch checks the return values from these + library calls before using them in subsequent operations. (issue + ASTERISK-19655) Reported by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/1863/ ........ Merged + revisions 362151 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362152 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-13 16:12 +0000 [r362081-362085] Jonathan Rose + + * apps/app_forkcdr.c, /: Make ForkCDR e option not set end time of + the newly forked CDR log Prior to this patch, ForkCDR's e option + would immediately set the end time of the forked CDR to that of + the CDR that is being terminated. This resulted in the new CDR's + end time being roughly the same as it's beginning time (which is + in turn roughly the same as the original's end time). (closes + issue ASTERISK-19164) Reported by: Steve Davies Patches: + cdr_fork_end.v10.patch uploaded by Steve Davies (license 5012) + ........ Merged revisions 362082 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362084 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_meetme.c: Send relative path named recordings to the + meetme directory instead of sounds Prior to this patch, no effort + was made to parse the path name to determine a proper destination + for recordings of MeetMe's r option. This fixes that. Review: + https://reviewboard.asterisk.org/r/1846/ ........ Merged + revisions 362079 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 362080 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-12 20:08 +0000 [r362043] Paul Belanger + + * main/srv.c: Convert SRV lookup message to debug level This helps + clean up the Asterisk CLI by converting the log message from + verbose to debug + +2012-04-12 16:29 +0000 [r361998] Richard Mudgett + + * configs/asterisk.conf.sample, UPGRADE.txt, pbx/pbx_config.c, + include/asterisk/options.h, main/asterisk.c: Add option to invoke + the extensions.conf stdexten using the legacy macro method. + ASTERISK-18809 eliminated the legacy macro invocation of the + stdexten in favor of the Gosub method without a means of + backwards compatibility. (issue ASTERISK-18809) (closes issue + ASTERISK-19457) Reported by: Matt Jordan Tested by: rmudgett + Review: https://reviewboard.asterisk.org/r/1855/ + +2012-04-12 16:25 +0000 [r361968-361987] Kinsey Moore + + * /, channels/chan_iax2.c: Make trunkfreq take effect when set + Previously, setting trunkfreq had no effect on initial load or on + reload and only ever used the default value. This causes + trunkfreq to be used appropriately on initial load and reload. + (closes issue ASTERISK-19521) Patch-by: Jaco Kroon ........ + Merged revisions 361972 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361981 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * Makefile, build_tools/cflags.xml, /, + build_tools/menuselect-deps.in, codecs/gsm/src/k6opt.s, + configure, codecs/gsm/Makefile, configure.ac, Makefile.rules, + makeopts.in, codecs/lpc10/Makefile: Simplify build system + architecture optimization This change to the build system rips + out any usage of PROC along with architecture-specific + optimizations in favor of using -march=native where it is + supported. This fixes broken builds on 64bit Intel systems and + results in better optimized code on systems running GCC 4.2+. + Review: https://reviewboard.asterisk.org/r/1852/ (closes issue + ASTERISK-19462) ........ Merged revisions 361955 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361956 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-11 17:20 +0000 [r361909] Jonathan Rose + + * /, configs/queues.conf.sample, apps/app_queue.c: Change default + value of 'ignorebusy' on Queue members so that behavior is more + like 1.8 Prior to this patch, in order to restore that behavior, + a function would have to be used on the QueueMember to make the + ringinuse option do anything, which is pretty unreasonable. + (closes issue ASTERISK-19536) reported by: Philippe Lindheimer + Review: https://reviewboard.asterisk.org/r/1860/ ........ Merged + revisions 361907 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-10 21:50 +0000 [r361856] Richard Mudgett + + * channels/chan_dahdi.c, /: Prevent invalid access of free'd memory + if DAHDI channel during an MWI event In the MWI processing loop, + when a valid event occurs the temporary caller ID information is + deallocated. If a new DAHDI channel is successfully created, the + event is passed up to the analog_ss_thread without error and the + loop exits. If, however, the DAHDI channel is not created, then + the caller ID struct has been free'd, and the gains reset to + their previous level. This will almost certainly cause an invalid + access to the free'd memory, either in subsequent calls to + callerid_free or calls to callerid_feed. * Rework the -r361705 + patch to better manage the cs and mtd allocated resources. * + Fixed use of mwimonitoractive flag to be correct if the + mwi_thread() fails to start. ........ Merged revisions 361854 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 361855 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-10 19:58 +0000 [r361659-361805] Matthew Jordan + + * /, main/http.c: Fix crash caused by unloading or reloading of + res_http_post When unlinking itself from the registered HTTP + URIs, res_http_post could inadvertently free all URIs registered + with the HTTP server. This patch modifies the unregister method + to only free the URI that is actually being unregistered, as + opposed to all of them. ........ Merged revisions 361803 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361804 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * funcs/func_curl.c, /: Allow func_curl to exit gracefully if list + allocation fails during write If the global_curl_info data + structure could not be allocated, the datastore associated with + the operation would be free'd, but the function would not return. + This would later dereference the datastore, almost certainly + causing Asterisk to crash. With this patch, if the data structure + is not allocated the method will return an error code, and not + attempt any further operation. ........ Merged revisions 361753 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 361754 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_dahdi.c, /: Prevent invalid access of free'd memory + if DAHDI channel during an MWI event In the MWI processing loop, + when a valid event occurs the temporary caller ID information is + deallocated. If a new DAHDI channel is successfully created, the + event is passed up to the analog_ss_thread without error and the + loop exits. If, however, the DAHDI channel is not created, then + the caller ID struct has been free'd, and the gains reset to + their previous level. This will almost certainly cause an invalid + access to the free'd memory, either in subsequent calls to + callerid_free or calls to callerid_feed. This patch makes it so + that we only free the caller ID structure if a DAHDI channel is + successfully created, and we bump the gains back up if we fail to + make a DAHDI channel. ........ Merged revisions 361705 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361706 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, funcs/func_global.c: Change SHARED function to use a safe + traversal when modifying a variable When the SHARED function + modifies a variable, it removes it from its list of variables and + reinserts the new value at the head of the list of variables. + Doing this inside a standard list traversal can be dangerous, as + the standard list traversal does not account for the list being + changed. While the code in question should not cause a use after + free violation due to its breaking out of the loop after freeing + the variable, it could lead to a maintenance issue if the loop + was modified. This also fixes a violation reported by a static + analysis tool, which also makes this code easier to maintain in + the future. ........ Merged revisions 361657 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361658 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-06 22:00 +0000 [r361561-361608] Matthew Jordan + + * /, res/res_calendar_ews.c: Fix memory leak in res_calendar_ews + when event email address node is empty If the XML calendar data + returned by a Microsoft Exchange Web Service specifies an XML + Event E-Mail Address ("EmailAddress"), and no e-mail address is + provided, a condition existed where an ast_calendar_attendee + struct would be allocated but not appended to the list of + attendees. Because of that, the memory associated with the + attendee would never be freed. This patch frees the memory if no + e-mail address is provided. ........ Merged revisions 361606 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361607 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_meetme.c: Fix memory leak when using MeetMeAdmin 'e' + option with user specified A memory leak/reference counting leak + occurs if the MeetMeAdmin 'e' command (eject last user that + joined) is used in conjunction with a specified user. Regardless + of the command being executed, if a user is specified for the + command, MeetMeAdmin will look up that user. Because the 'e' + option kicks the last user that joined, as opposed to the one + specified, the reference to the user specified by the command + would be leaked when the user variable was assigned to the last + user that joined. ........ Merged revisions 361558 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361560 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-06 19:58 +0000 [r361523] Richard Mudgett + + * /, main/message.c: Don't add an empty MESSAGE_DATA(key) header if + it doesn't already exist. Doing Set(MESSAGE_DATA(key)=) would add + an empty key header if the key header did not already exist. If + it already existed it would delete it. * Made msg_set_var_full() + exit early if the named variable did not already exist and the + value to set is empty. ........ Merged revisions 361522 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-06 18:19 +0000 [r361476] Kinsey Moore + + * channels/chan_unistim.c, main/pbx.c, /, channels/chan_sip.c, + funcs/func_strings.c, formats/format_ogg_vorbis.c, + channels/console_video.c, apps/app_ices.c, channels/chan_gtalk.c, + channels/chan_iax2.c, res/res_config_sqlite.c, res/res_srtp.c, + main/cdr.c, main/tcptls.c, channels/console_gui.c, + funcs/func_channel.c, apps/app_sms.c, addons/chan_mobile.c, + apps/app_chanspy.c, main/xmldoc.c, channels/chan_mgcp.c, + res/res_config_sqlite3.c, res/res_clioriginate.c, + apps/app_voicemail.c: Add missing newlines to CLI logging + ........ Merged revisions 361471 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361472 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-06 16:33 +0000 [r361429] Paul Belanger + + * bridges/bridge_builtin_features.c, /, funcs/func_sysinfo.c, + bridges/bridge_multiplexed.c: Multiple revisions 361403,361412 + ........ r361403 | pabelanger | 2012-04-06 12:24:36 -0400 (Fri, + 06 Apr 2012) | 2 lines Fix typo in svn:keywords ........ r361412 + | pabelanger | 2012-04-06 12:27:30 -0400 (Fri, 06 Apr 2012) | 2 + lines Fix typo in svn:keywords ........ Merged revisions + 361403,361412 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361422 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-06 15:50 +0000 [r361382] Russell Bryant + + * /, configs/rpt.conf.sample (removed), + configs/usbradio.conf.sample (removed), apps/rpt_flow.pdf + (removed): Remove a few more files related to chan_usbradio and + app_rpt. ........ Merged revisions 361380 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361381 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-06 14:02 +0000 [r361334] Matthew Jordan + + * /, channels/chan_sip.c: Fix a typo in the warning messages for an + ignored media stream Added a '\n' to the warning messages when we + ignore a media stream due to the port number being '0'. (closes + issue ASTERISK-19646) Reported by: Badalian Vyacheslav ........ + Merged revisions 361332 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361333 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-06 13:32 +0000 [r361331] Kinsey Moore + + * apps/app_dial.c, /: Remove unnecessary error message in + app_dial.c The error message for failure to stop autoservice + after a gosub or macro call during a dial was removed for macro + while Asterisk 1.4 was still being actively developed. The + corresponding gosub error message was never removed. (closes + issue ASTERISK-19551) ........ Merged revisions 361329 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361330 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-05 17:22 +0000 [r361092-361279] Jonathan Rose + + * /, apps/app_meetme.c: Fix MusicOnHold in MeetMe so that it always + uses the class if it's been defined There were a few instances of + restarting music on hold in meetme that would cause Asterisk to + revert to the default class of music on hold for no adequate + reason. Review: https://reviewboard.asterisk.org/r/1844/ ........ + Merged revisions 361269 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361270 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, addons/ooh323cDriver.c: Fix some stuff involving calls to + memcpy and memset The important parts of the patch were already + applied through other updates. (closes issue ASTERISK-19445) + Reported by: Makoto Dei Patches: memset-memcpy-length.patch + uploaded by Makoto Dei (license 5027) ........ Merged revisions + 361210 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 361211 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, funcs/func_devstate.c: Make 'help devstate change' display + properly (get rid of excess comma) (closes issue ASTERISK-19444) + Reported by: Makoto Dei Patches: + devstate-change-usage-truncate.patch uploaded by Makoto Dei + (license 5027) ........ Merged revisions 361201 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361208 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/channel.c, pbx/pbx_loopback.c, addons/chan_ooh323.c, /, + channels/chan_sip.c, main/app.c, pbx/pbx_realtime.c, + apps/app_externalivr.c, channels/chan_iax2.c, + res/res_fax_spandsp.c, apps/app_milliwatt.c: Replace GNU + old-style field designator extensions to fix clang warnings + (issue ASTERISK-19540) Reported by: Makoto Dei Patches: + clang-gnu-designator.patch uploaded by Makoto Dei (license 5027) + ........ Also add from the patch the portion in res_fax_spandsp + that didn't apply to 1.8 Merged revisions 361142 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 (closes issue + ASTERISK-19540) ........ Merged revisions 361143 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_meetme.c: Make the MeetMeAdmin N command (mute all + nonadmins) not mute admins (Closes Issue ASTERISK-19335) Reported + by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1843/ + ........ Merged revisions 361090 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361091 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-03 20:14 +0000 [r361042] Kinsey Moore + + * /, apps/app_transfer.c: Fix the display of documentation for + Transfer This came up while fixing documentation generation for + many other cases where the argument separator was not being + displayed properly. Now that it is displayed properly, it shows + up in the wrong place for Transfer since the '/' is only required + if Tech is present. (related to issue ASTERISK-18168) ........ + Merged revisions 361040 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 361041 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-04-03 20:03 +0000 [r361038-361039] Mark Murawki + + * include/asterisk/manager.h: Fix dev-mode compiler warning about + gnu_printf (related to ASTERISK-19575) + + * main/channel.c, main/manager.c, main/utils.c, + include/asterisk/channel.h, include/asterisk/strings.h, CHANGES, + include/asterisk/manager.h: Allow the Hangup manager action to + match channels by regex * Hangup now can take a regular + expression as the Channel option. If you want to hangup multiple + channels, use /regex/ as the Channel option. Existing behavior to + hanging up a single channel is unchanged, but if you pass a + regex, the manager will send you a list of channels back that + were hung up. (closes issue ASTERISK-19575) Reported by: Mark + Murawski Tested by: Mark Murawski + +2012-04-02 22:27 +0000 [r360994] Kinsey Moore + + * /, channels/chan_sip.c: Stop sending out RTCP if RTP is inactive + This change prevents Asterisk from sending RTCP receiver reports + during a remote bridge since it is no longer receiving media and + should not be reporting anything. (related to ASTERISK-19366) + ........ Merged revisions 360987 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360993 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-30 21:38 +0000 [r360935] Richard Mudgett + + * /, main/logger.c: Fix logger deadlock on Asterisk shutdown. The + logger_thread() had an exit path that failed to release the + logmsgs list lock. * Make logger_thread() exit path unlock the + logmsgs list lock. * Made ast_log() not queue any messages to the + logmsgs list if the close_logger_thread flag is set. (issue + ASTERISK-19463) Reported by: Matt Jordan ........ Merged + revisions 360933 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360934 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-29 23:36 +0000 [r360872-360886] Mark Michelson + + * /, main/features.c: Fix potential race condition during call + pickup. Prior to this patch, a connected line update was queued + during call pickup and then an answer frame was queued. The + original caller would presumably then have his connected line + updated and then the call would be answered. In actuality, the + answer frame was not how the call ended up being answered. + Rather, an odd section in app_dial that checks if the called + channel's state is up. The result is that the order of the + connected line update and the answer were variable. In most + cases, this wasn't actually a bad thing. However, if the 'I' + option was passed to dial, the connected line update would be + inhibited. The fix is to queued the connected line after the + answer frame is queued. This way the race in app_dial is between + two conditions resulting in an answer. This way the connected + line update occurs after the answer every time. (closes issue + ASTERISK-19183) Reported by: Thomas Arimont Tested by: Thomas + Arimont Mark Michelson Patches: ASTERISK-19183.patch uploaded by + Mark Michelson (license 5049) ........ Merged revisions 360884 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 360885 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Improve accuracy of identifying + information sent in dialog-info SIP NOTIFY requests. This change + makes use of connected party information in addition to caller ID + in order to populate local and remote XML elements in the + dialog-info NOTIFYs. (closes issue ASTERISK-16735) Reported by: + Maciej Krajewski Tested by: Maciej Krajewski Patches: + local_remote_hint2.diff uploaded by Mark Michelson (license 5049) + ........ Merged revisions 360862 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360863 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-29 21:57 +0000 [r360827] Richard Mudgett + + * include/asterisk/astobj2.h, main/astobj2.c: Misc changes to make + astobj2 enhancement diffs easier to follow. * Rename astobj2 API + parameter funcname to func. * Rename astobj2 API iterator + parameter to iter. * Update some documentation for OBJ_MULTIPLE. + +2012-03-29 20:01 +0000 [r360785-360787] Jonathan Rose + + * include/asterisk/logger.h, main/dial.c, main/pbx.c, + include/asterisk/bridging.h, main/features.c, main/logger.c, + CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample: + Introducing the log message unique call identifiers feature Log + messages will now display a call number that they are tied to + (ordered for calls based on when they started). This feature is + made to be minimally invasive without requiring changes to many + of the existing log messages. These IDs won't show up for verbose + messages on CLI (but they will in log files) This is currently in + phase II of production, see more about this feature on the wiki + -- + https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging + Review: https://reviewboard.asterisk.org/r/1823/ + + * include/asterisk/logger.h, main/dial.c, main/pbx.c, /, + include/asterisk/bridging.h, main/features.c, main/logger.c, + CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample: + undoing 360785 due to merging mistake + + * include/asterisk/logger.h, main/dial.c, main/pbx.c, /, + include/asterisk/bridging.h, main/features.c, main/logger.c, + CHANGES, apps/app_mixmonitor.c, configs/logger.conf.sample: + Introducing the log message unique call identifiers feature Log + messages will now display a call number that they are tied to + (ordered for calls based on when they started). This feature is + made to be minimally invasive without requiring changes to many + of the existing log messages. These IDs won't show up for verbose + messages on CLI (but they will in log files) This is currently in + phase II of production, see more about this feature on the wiki + -- + https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging + Review: https://reviewboard.asterisk.org/r/1823/ + +2012-03-28 19:39 +0000 [r360724] Terry Wilson + + * channels/chan_jingle.c, addons/chan_ooh323.c, + cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, + channels/chan_gtalk.c, apps/confbridge/conf_config_parser.c: Fix + setting CDR variables in the hangup extension A previous CDR fix + for setting CDR variables during a bridge via custom dialplan + features broke setting CDR variables in the hangup extension. + This patch fixes the issue. Review: + https://reviewboard.asterisk.org/r/1794/ ........ Merged + revisions 358978 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358989 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-27 18:44 +0000 [r360673] Mark Michelson + + * /, channels/chan_sip.c: Make a debug message regarding + subscription changes more accurate. I was getting confused during + some testing why Asterisk was saying that a subscription was + being added when it was clearly being removed. This fixes that + confusion. ........ Merged revisions 360625 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360672 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-27 17:13 +0000 [r360626-360627] Richard Mudgett + + * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c: + Add global ao2 array container. Global ao2 objects must always + exist after initialization because there is no access control to + obtain another reference to the global object. It is expected + that module configuration could use these new API calls to + replace an active configuration parameter object with an updated + configuration parameter object. With these new API calls, the + global object could be replaced, removed, or referenced without + the risk of someone using a stale global object pointer. Review: + https://reviewboard.asterisk.org/r/1824/ + + * main/astobj2.c: Attempt to be more helpful when using a bad ao2 + object pointer. + +2012-03-27 14:43 +0000 [r360576] Jonathan Rose + + * /, configure: Updates config with bootstrap where I changed + configure.ac in r360488 (issue ASTERISK-17842) Reported by: Bryon + Clark ........ Merged revisions 360574 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360575 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-26 21:22 +0000 [r360536] Paul Belanger + + * main/dnsmgr.c, /: Convert ast_verb() to ast_debug() and increase + log level Rather then flood the CLI with verbose messages, we've + changed the level to debug. This will help keep the CLI clean. + +2012-03-26 19:49 +0000 [r360490] Jonathan Rose + + * /, configure.ac: Fix BETTER_BACKTRACES library detection for + Fedora/RedHat/CentOS (closes ASTERISK-17842) Reported by: Bryon + Clark Patches: 20110512__issue19278.diff.txt uploaded by Tilghman + Lesher (license 5003) configure_bfd_with_dl_and_iberty.patch + uploaded by Bryon Clark (license 6157) ........ Merged revisions + 360488 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 360489 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-24 23:49 +0000 [r360359-360415] Russell Bryant + + * funcs/func_curl.c, /: func_curl: Fix leak of an ast_str in error + handling code path. ........ Merged revisions 360413 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360414 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_iax2.c: chan_iax2: Use OBJ_NODATA to be a bit more + explicit. This is just a minor code cleanup change. These uses of + ao2_callback() would never return anything since the callbacks + always returned 0. However, be more explicit that no returned + results are wanted by specifying OBJ_NODATA. + + * /, apps/app_page.c: app_page: Fix a memory leak on every Page(). + dial_list is a dynamically allocated array that is allocated at + the beginning of Page() based on how many devices will be dialed. + This was never being freed. ........ Merged revisions 360363 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360364 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_jack.c: app_jack: fix datastore memory leak in error + handling path. ........ Merged revisions 360360 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360361 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/ast_expr2.h, res/ael/ael.tab.c, main/ast_expr2.y, + main/ast_expr2f.c, res/ael/ael_lex.c, res/ael/ael.tab.h, + main/ast_expr2.c: Multiple revisions 360356-360357 ........ + r360356 | russell | 2012-03-23 22:33:36 -0400 (Fri, 23 Mar 2012) + | 6 lines expression parser: Fix (theoretical) memory leak. Fix a + memory leak that is very unlikely to actually happen. If a + malloc() succeeded, but the following strdup() failed, the memory + from the original malloc() would be leaked. ........ r360357 | + russell | 2012-03-23 22:34:39 -0400 (Fri, 23 Mar 2012) | 6 lines + Rebuild parsers. This is needed to include the last fix to + main/ast_expr2.y. The changes look much bigger as this + regeneration of the code was done with newer versions of flex and + bison. ........ Merged revisions 360356-360357 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360358 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-24 00:40 +0000 [r360264-360311] Richard Mudgett + + * main/channel.c, /, channels/sig_pri.c: Make number not available + presentation also set screening to network provided. Q.951 + indicates that when the presentation indicator is "Number not + available due to interworking" for a number then the screening + indicator field should be "Network provided". * Made + ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE + when the presentation is "Number not available due to + interworking". This fix makes Asterisk consistent and it also + makes it consistent with earlier branches as far as this + presentation value is concerned. * Made pri_to_ast_presentation() + and ast_to_pri_presentation() conversions handle the "Number not + available due to interworking" case better in sig_pri.c. This + change is possible because the minimum required libpri version + (v1.4.11) has the necessary defines in libpri.h. ........ Merged + revisions 360309 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360310 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Add missing initialization of + update_redirecting in chan_sip.c ........ Merged revisions 360262 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 360263 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-22 21:25 +0000 [r360227] Jonathan Rose + + * apps/app_dial.c, include/asterisk/utils.h, main/features.c, + main/utils.c, CHANGES, apps/app_queue.c: Adds F option to Bridge + application Similar to dial and queue F option. (Closes issue + ASTERISK-19282) Reported by: To Patches: bridge_f-v3.diff + uploaded by To (license 6347) Review: + https://reviewboard.asterisk.org/r/1825/ + +2012-03-22 19:51 +0000 [r360190] Kinsey Moore + + * main/udptl.c, main/stdtime/test.c, main/autoservice.c, + main/rtp_engine.c, main/frame.c, main/fskmodem_float.c, + main/sha1.c, main/say.c, main/ecdisa.h, main/utils.c, + main/devicestate.c, main/taskprocessor.c, main/indications.c, + main/enum.c, main/config.c, main/loader.c, main/term.c, + main/cli.c, main/io.c, main/ulaw.c, main/channel.c, main/dial.c, + main/manager.c, main/tdd.c, main/strcompat.c, main/plc.c, + main/features.c, main/logger.c, main/fskmodem_int.c, main/app.c, + main/stdtime/localtime.c, main/image.c, main/dns.c, + main/message.c, main/md5.c, main/sched.c, main/lock.c, + main/pbx.c, main/dnsmgr.c, main/slinfactory.c, main/translate.c, + main/jitterbuf.c, main/cel.c, main/chanvars.c, main/netsock.c, + main/srv.c, main/privacy.c, main/fixedjitterbuf.c, main/file.c, + main/callerid.c, main/event.c, main/astmm.c, main/audiohook.c, + main/cygload.c, main/fixedjitterbuf.h, main/asterisk.c, + main/xmldoc.c, main/dsp.c, main/timing.c: Kill off red blobs in + most of main/* Everything still compiled after making these + changes, so I assume these whitespace-only changes didn't break + anything (and shouldn't have). + +2012-03-21 14:55 +0000 [r360140] Jonathan Rose + + * /, contrib/scripts/install_prereq: Update install_prereq script + to include missing GSM library for debian amd move SQLite3. + (closes issue ASTERISK-19367) Reported by: Andrew Latham Patches: + debian_install_prereq.diff uploaded by Andrew Latham (license + 5985) ........ Merged revisions 360138 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360139 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-21 14:47 +0000 [r360137] Tzafrir Cohen + + * /, configure, configure.ac: Also detect gmime 2.6 Also detect + gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen + (License #5035) ........ Merged + revisions 360087 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360098 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-21 13:31 +0000 [r360089] Matthew Jordan + + * /, channels/chan_sip.c: Ensure Asterisk sends a BYE when pending + on the final response to a re-INVITE When Asterisk detects a + hangup and cannot send a BYE due to a pending INVITE, it sets the + pendingbye flag and waits for the final response to that INVITE. + When the response is received, it transmits the BYE. If, however, + that INVITE request is a pending re-INVITE, it needs to first + send a CANCEL request to terminate the pending re-INVITE. In that + circumstance, Asterisk was, in some scenarios, clearing the + pendingbye flag after processing the CANCEL request and not + checking for a pending BYE when receiving the final 487 response + to the INVITE. This patch ensures that if the pendingbye flag is + set, it is honored regardless of the nature of the INVITE request + currently in flight. (closes issue ASTERISK-19365) Reported by: + Thomas Arimont Tested by: Thomas Arimont Patches: + bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license + 6283) Review: https://reviewboard.asterisk.org/r/1807 ........ + Merged revisions 360086 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360088 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-20 20:42 +0000 [r360036] Kinsey Moore + + * /, apps/app_echo.c: Prevent Echo() from relaying control, null, + and modem frames Echo()'s description states that it echoes + audio, video, and DTMF except for # while it actually echoes any + frame that it receives other than DTMF #. This was causing frame + storms in the test suite in some circumstances where Echo() was + attached to both ends of a pair of local channels and control + frames were being periodically generated. Echo()'s behavior and + description have been modifed so that it only echoes media and + non-# DTMF frames. ........ Merged revisions 360033 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 360034 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-20 18:17 +0000 [r359983] Sean Bright + + * /, UPGRADE.txt, channels/chan_iax2.c, include/asterisk/manager.h: + chan_iax2: Correct spelling of 'Port' header in IAX2 PeerStatus + AMI Events The PeerStatus event for IAX2 channels currently + includes a header named Post which should have been Port. Post + was removed and the AMI version has been updated to 1.3. ........ + Merged revisions 359982 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-20 17:31 +0000 [r359942-359981] Richard Mudgett + + * main/data.c, main/pbx.c, main/manager.c, /, main/features.c, + include/asterisk/manager.h, main/db.c: Allow AMI action callback + to be reentrant. Fix AMI module reload deadlock regression from + ASTERISK-18479 when it tried to fix the race between calling an + AMI action callback and unregistering that action. Refixes + ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2 + object guaranteed that there were no active callbacks that + mattered when ast_manager_unregister() was called. Unfortunately, + this causes the deadlock situation. The patch stops locking the + ao2 object to allow multiple threads to invoke the callback + re-entrantly. There is no way to guarantee a module unload will + not crash because of an active callback. The code attempts to + minimize the chance with the registered flag and the maximum 5 + second delay before ast_manager_unregister() returns. The trunk + version of the patch changes the API to fix the race condition + correctly to prevent the module code from unloading from memory + while an action callback is active. * Don't hold the lock while + calling the AMI action callback. (closes issue ASTERISK-19487) + Reported by: Philippe Lindheimer Review: + https://reviewboard.asterisk.org/r/1818/ Review: + https://reviewboard.asterisk.org/r/1820/ ........ Merged + revisions 359979 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359980 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * res/res_mutestream.c: Convert MuteAudio documentation to XML. * + Added missing error exits with cause in manager_mutestream(). * + Cleaned up manager_mutestream() and func_mute_write(). * Some + whitespace and comment cleanup. + +2012-03-16 21:00 +0000 [r359905] Jonathan Rose + + * /, apps/app_chanspy.c: Prevent chanspy from binding to zombie + channels This patch addresses a bug with chanspy on local + channels which roughly 50% of the time would create a situation + where chanspy can latch onto a zombie channel, keeping the zombie + alive forever and causing the channel doing the spying to never + be able to hang up. (closes issue ASTERISK-19493) Reported by: + lvl Review: https://reviewboard.asterisk.org/r/1819/ ........ + Merged revisions 359892 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359898 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-16 20:37 +0000 [r359904] Richard Mudgett + + * include/asterisk/app.h, main/app.c: Simplify some code in + ast_app_run_sub(). * Remove unnnecessary const from const char * + const var declaration in the ast_app_run_macro() and + ast_app_run_sub() prototypes. The second const is unnecessary. + +2012-03-16 15:38 +0000 [r359857] Mark Michelson + + * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES: + Revert the pre-dial addition. The code may be just fine, but it + had not received a "ship it!" on review board yet. + +2012-03-16 08:27 +0000 [r359811] Alec L Davis + + * /, channels/sip/include/sip.h: Missed lastinvite CSeq int to + uint32_t change from Review: + https://reviewboard.asterisk.org/r/1699/ ........ Merged + revisions 359809 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359810 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-15 20:11 +0000 [r359772] Mark Murawki + + * main/pbx.c: Fix warning from commit r359705 (predial options for + app_dial) + +2012-03-15 19:11 +0000 [r359708] Matthew Jordan + + * /, main/utils.c: Fix remotely exploitable stack overflow in HTTP + manager There exists a remotely exploitable stack buffer overflow + in HTTP digest authentication handling in Asterisk. The + particular method in question is only utilized by HTTP AMI. When + parsing the digest information, the length of the string is not + checked when it is copied into temporary buffers allocated on the + stack. This patch fixes this behavior by parsing out pre-defined + key/value pairs and avoiding unnecessary copies to the stack. + (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested + by: Matt Jordan ........ Merged revisions 359706 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359707 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-15 18:58 +0000 [r359705] Mark Murawki + + * apps/app_dial.c, main/pbx.c, include/asterisk/pbx.h, CHANGES: Add + options PreDial options 'b' and 'B' to app_dial * Added 'b' and + 'B' options to Dial. These options will allow you to run + last-minute dialplan on the caller and callee channels while the + Dial application is executing, but before the call is started. + For example you can use the 'b' option to run dialplan on the + callee channel to get the name of the newly created channel right + away. Review: https://reviewboard.asterisk.org/r/1229/ (closes + issue: ASTERISK-19548) Reported by: Mark Murawski Tested by: Mark + Murawski, Stefan Schmidt + +2012-03-15 18:55 +0000 [r359704] Matthew Jordan + + * /, apps/app_milliwatt.c: Fix remotely exploitable stack overrun + in Milliwatt Milliwatt is vulnerable to a remotely exploitable + stack overrun when using the 'o' option. This occurs due to the + milliwatt_generate function not accounting for + AST_FRIENDLY_OFFSET when calculating the maximum number of + samples it can put in the output buffer. This patch resolves this + issue by taking into account AST_FRIENDLY_OFFSET when determining + the maximum number of samples allowed. Note that at no point is + remote code execution possible. The data that is written into the + buffer is the pre-defined Milliwatt data, and not custom data. + (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested + by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by + Russell Bryant (license 6283) Note that this patch was written by + Russell, even though Matt uploaded it ........ Merged revisions + 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 + ........ Merged revisions 359656 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359694 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-15 18:34 +0000 [r359651] Paul Belanger + + * channels/chan_sip.c: Remove unused variable ‘srch’ Missed on the + previous commit + +2012-03-15 18:32 +0000 [r359644] Richard Mudgett + + * apps/app_dial.c, /, apps/app_queue.c: Add missing connected line + macro calls to initial dial for Dial and Queue apps. The + connected line interception macros do not get executed when the + outgoing channel is initially created and that channel's + caller-id is implicitly imported into the incoming channel's + connected line data. If you are using the interception macros, + you would expect that they get run for every change to a + channel's connected line information outside of normal dialplan + execution. Review: https://reviewboard.asterisk.org/r/1817/ + ........ Merged revisions 359609 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359620 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-15 17:36 +0000 [r359607] Paul Belanger + + * channels/chan_sip.c: Remove some dead code found in + _sip_show_peers() Review: + https://reviewboard.asterisk.org/r/1696/ + +2012-03-15 00:54 +0000 [r359456-359560] Russell Bryant + + * /, channels/chan_iax2.c: chan_iax2: Fix use of uninitialized + sockaddr_in in try_transfer(). Initialize a struct sockaddr_in in + try_transfer() so that the code isn't (potentially) trying to + read from it while uninitialized. ........ Merged revisions + 359558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 359559 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_gtalk.c: chan_gtalk: Fix potential use of + uninitialized variable. Avoid potential use of idroster in + gtalk_alloc() before it has been initialized. ........ Merged + revisions 359508 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359509 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_chanisavail.c: app_chanisavail: Fix use of + uninitialized variable. Ensure that status is set before it is + used by resetting it during each loop iteration. This could have + resulted in incorrect results from this app. ........ Merged + revisions 359486 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359491 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/udptl.c, /: udptl: Ensure fec[] in udptl_build_packet() is + initialized. Scan results indicated that this array could be used + uninitialized. At a quick look, it looks correct. In any case, + initializing it is a Good Thing (tm). ........ Merged revisions + 359457 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 359458 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * include/asterisk/app.h, /: app.h: Always initialize + AST_DECLARE_APP_ARGS(). This patch ensures that the struct + defined by AST_DECLARE_APP_ARGS() is always fully initialized. + I'm not sure if this fixes any real bugs, but it silences a bunch + of warnings from coverity, and is generally a good thing to do + anyway. ........ Merged revisions 359452 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359454 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-14 22:38 +0000 [r359455] Richard Mudgett + + * main/channel.c, /, channels/chan_agent.c, + include/asterisk/channel.h: Fix deadlock potential with some + ast_indicate/ast_indicate_data calls. Calling + ast_indicate()/ast_indicate_data() with the channel lock held can + result in a deadlock with a local channel because of how local + channels need to avoid deadlock. ........ Merged revisions 359451 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 359453 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-14 18:56 +0000 [r359406] Matthew Jordan + + * tests/test_jitterbuf.c (added): Add tests for main/jitterbuf.c + This patch adds unit tests for main/jitterbuf.c. This includes + checking for the following: * Nominal insertion and retrieval of + frames * Insertion and retrieval of frames where the frames are + inserted out of order with respect to the previous frame * + Insertion and retrieval of frames where some number of frames + that would occur in the expected sequence are instead dropped * + Insertion and retrieval of frames with an arrival time that does + not occur at the same rate as the surrounding frames * + Resynchronization of the jitter buffer when an inserted frame + breaks the resynchronization threshold * Overfilling of the + jitter buffer For each of the tests, both JB_TYPE_VOICE and + JB_TYPE_CONTROL permutations exist. Review: + https://reviewboard.asterisk.org/r/1815 (issue: ASTERISK-18964) + Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan + +2012-03-14 18:12 +0000 [r359360] Richard Mudgett + + * include/asterisk/channel_internal.h: Three copies of the file + contents in channel_internal.h are a bit excessive. + +2012-03-14 17:48 +0000 [r359359] Matthew Jordan + + * /, main/jitterbuf.c: Fix incorrect jitter buffer overflow due to + missed resynchronizations When a change in time occurs, such that + the timestamps associated with frames being placed into an + adaptive jitter buffer (implemented in jitterbuf.c) are + significantly different then the previously inserted frames, the + jitter buffer checks to see if it needs to be resynched to the + new time frame. If three consecutive packets break the threshold, + the jitter buffer resynchs itself to the new timestamps. This + currently only occurs when history is calculated, and hence only + on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other + hand, are never passed to the history calculations. Because of + this, if the jump in time is greater then the maximum allowed + length of the jitter buffer, the JB_TYPE_CONTROL frames are + dropped and no resynchronization occurs. Alterntively, if the + overfill logic is not triggered, the JB_TYPE_CONTROL frame will + be placed into the buffer, but with a time reference that is not + applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger + the overflow logic until reads from the jitter buffer reach the + errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL + frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames + are unlikely to occur in multiples, it perform the + resynchronization on any JB_TYPE_CONTROL frame that breaks the + resynch threshold. Note that this only impacts chan_iax2, as + other consumers of the adaptive jitter buffer use the abstract + jitter buffer API, which does not use JB_TYPE_CONTROL frames. + Review: https://reviewboard.asterisk.org/r/1814/ (closes issue + ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt + Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw + (license 5722) ........ Merged revisions 359356 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359358 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-14 17:39 +0000 [r359357] Richard Mudgett + + * apps/app_dial.c, main/channel.c, /: Fix Dial m and r options and + forked calls generating warnings for voice frames. When connected + line support was added, the wait_for_answer() variable single + changed its meaning slightly. Unfortunately, the places where + single was used did not necessarily get updated to reflect that + change. Also audio/video frames were sent to all forked calls + when the endpoints were never made compatible. * Don't pass + audio/video media frames when the channels have not been made + compatible. * Added handling of AST_CONTROL_SRCCHANGE to + app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD + because that frame can also pass a requested MOH class. (closes + issue ASTERISK-16901) Reported by: Chris Gentle (closes issue + ASTERISK-17541) Reported by: clint Review: + https://reviewboard.asterisk.org/r/1805/ ........ Merged + revisions 359344 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359355 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-14 14:40 +0000 [r359306] Matthew Jordan + + * include/asterisk/astobj2.h: Force non-inlining of + ao2_iterator_destroy when TEST_FRAMEWORK is enabled In r357272, + astobj2 was changed to automatically enable REF_DEBUG when the + TEST_FRAMEWORK flag was enabled. Unfortunately, some compilers + (gcc 4.5.1 at least) will attempt to inline ao2_iterator_destroy + in handle_astobj2_test. This by itself is not a problem; + unfortunately, the compiler believes that there is a code path + wherein an object allocated on the stack will be free'd. As + warnings are treated as errors, this prevents compilation of + astobj2. This patch works around that by adding the noinline + attribue to ao2_iterator_destroy, but only if the TEST_FRAMEWORK + flag is enabled. Preventing inlining is only needed for the test + method defined in astobj2, which is also only enabled if + TEST_FRAMEWORK is enabled. + +2012-03-14 10:56 +0000 [r359052-359261] Russell Bryant + + * include/asterisk/logger.h, /, main/logger.c: Fix bogus + reads/writes of console log levels in asterisk.c This patch + updates the NUMLOGLEVELS define in logger.h to 32, to match the + fact that logger.c implements 32 log levels (because of the + custom log level stuff). asterisk.c uses this define to size an + array of levels per remote console. This array is modified in + ast_console_toggle_loglevel(), which is called by the "logger set + level" CLI command. While the documentation for the CLI command + doesn't make it terribly obvious, you can use this CLI command to + toggle a custom log level on a remote console, as well. However, + doing so led to an invalid array index in asterisk.c. This array + is read from any time a log message is written to a console. So, + all custom log level messages resulted in a bogus read if a + remote console was connected. ........ Merged revisions 359259 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 359260 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_externalivr.c, channels/chan_iax2.c: Fix invalid + reads/writes due to incorrect sizeof(). These few places in the + code used sizeof() on h_addr in struct hostent. This is + sizeof(char *). The correct way to get the size of this address + is to use h_length. This error would result in reads/writes of 8 + bytes instead of 4 on 64-bit machines. ........ Merged revisions + 359211 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 359212 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/sched.c: Fix inaccurate sizeof() in sched.c. This code + just needed sizeof(int), not sizeof(int *). ........ Merged + revisions 359157 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359162 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, utils/astman.c: Fix incorrect sizeof() in astman. ........ + Merged revisions 359116 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359117 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, res/res_crypto.c: Fix incorrect usage of sizeof() in + res_crypto. In this case, just remove the memset(). There was a + redundant memset that is done correctly just 2 lines later. + ........ Merged revisions 359110 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359114 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, res/res_adsi.c: Fix broken usage of sizeof() in res_adsi. + ........ Merged revisions 359088 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359091 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/features.c: Fix incorrect sizeof() usage in features.c. + This didn't actually result in a bug anywhere, luckily. The only + place where the result of these memcpys was used is in app_dial, + and the only field that it read out of ast_call_feature was the + first one, which is an int, so these memcpys always copied just + enough to avoid a problem. ........ Merged revisions 359069 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359072 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final(). + ........ Merged revisions 359059 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359060 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/pbx.c, /: Don't use a buffer after it goes out of scope. 's' + is set to 'workspace'. Make sure 'workspace' doesn't go out of + scope while the reference to it via 's' is still used. ........ + Merged revisions 359056 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359057 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_usbradio.c (removed), /, channels/xpmr (removed), + build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, configure.ac, makeopts.in, + apps/app_rpt.c (removed): Remove chan_usbradio and app_rpt. These + modules are being maintained outside of the tree and have been + for a long time now, so it doesn't make sense to keep them here. + Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged + revisions 359050 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 359051 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-13 21:24 +0000 [r359011] Terry Wilson + + * include/asterisk/channel_internal.h (added): Add missing + channel_internal.h ...again. + +2012-03-13 21:18 +0000 [r358997] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, channels/sig_pri.c: Add ability + for chan_dahdi ISDN to block connected line updates per span. + Added new chan_dahdi.conf colp_send option parameter to block + connected line updates per span. (closes issue ASTERISK-17025) + Reported by: Michael Smith + +2012-03-13 20:43 +0000 [r358907-358993] Terry Wilson + + * /, main/features.c: Fix setting CDR variables in the hangup + extension A previous CDR fix for setting CDR variables during a + bridge via custom dialplan features broke setting CDR variables + in the hangup extension. This patch fixes the issue. Review: + https://reviewboard.asterisk.org/r/1794/ ........ Merged + revisions 358978 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358989 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * include/asterisk/devicestate.h, /, channels/chan_sip.c, + tests/test_devicestate.c, main/devicestate.c: Make hints for + invalid SIP devices return Unavail, not idle This patch + drastically simplifies the device state aggegation code. The old + method was not only overly complex, but also made it impossible + to return AST_DEVICE_INVALID from the aggregation code. The unit + test update is as a result of fixing that bug. The SIP change + stems from a bug introduced by removing a DNS lookup for + hostname-based SIP channels. (closes issue ASTERISK-16702) + Review: https://reviewboard.asterisk.org/r/1808/ ........ Merged + revisions 358943 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358944 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_voicemail.c: Fix IMAP storage compilation after + opaquification changes (closes issue ASTERISK-19513) + + * channels/chan_unistim.c, main/autoservice.c, + channels/chan_vpb.cc, channels/chan_local.c, main/rtp_engine.c, + res/res_musiconhold.c, bridges/bridge_multiplexed.c, + apps/app_followme.c, main/indications.c, main/cli.c, + main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c, + channels/sig_analog.c, main/manager.c, main/features.c, + apps/app_dumpchan.c, res/res_agi.c, main/app.c, + apps/app_confbridge.c, apps/app_externalivr.c, main/bridging.c, + apps/app_parkandannounce.c, apps/app_dial.c, main/pbx.c, + channels/chan_sip.c, channels/chan_bridge.c, + main/channel_internal_api.c, channels/chan_agent.c, + apps/app_disa.c, include/asterisk/channel.h, + apps/app_talkdetect.c, apps/app_queue.c, apps/app_speech_utils.c, + apps/app_channelredirect.c, main/file.c, res/snmp/agent.c, + apps/app_macro.c, apps/app_stack.c, apps/app_chanspy.c, + apps/app_mixmonitor.c: Finalize ast_channel opaquification + Review: https://reviewboard.asterisk.org/r/1786/ + +2012-03-13 17:01 +0000 [r358858-358861] Richard Mudgett + + * main/channel.c: Fix crash caused by opaquification change + -r356042. The set_format() function was more subtle in how it + modified the struct ast_channel readtrans/writetrans values. * + Fixed ast_activate_generator() conversion correctly. (closes + issue ASTERISK-19434) Reported by: Birger Harzenetter Tested by: + rmudgett + + * main/format.c: Use struct copy instead of memcpy(). + +2012-03-13 08:06 +0000 [r358812] Tilghman Lesher + + * res/ael/pval.c, funcs/func_dialplan.c, /, tests/test_gosub.c, + utils/ael_main.c, apps/app_stack.c, utils/conf2ael.c: Enable + macros in 1.8 to find the next highest "h" extension in a + context, like in 1.4. This change restores functionality that was + present in 1.4, when AEL macros were implemented with the Macro + dialplan application. Macros are fraught with functionality + issues, because they consume a large portion of the underlying + application stack. This limits the ability of AEL users to call + many layers of subroutines, an issue which Gosub does not have + (originally tested to 100,000 levels deep). Therefore, starting + in 1.6.0, AEL macros were implemented with Gosub. However, there + were some implicit behaviors of Macro, which were not replicated + at the same time as with the transition to Gosub, one of which is + documented in the related issue. In particular, the "h" extension + is designed to execute not in the Macro context, but in the + topmost calling context. Due to legacy issues with a misapplied + bugfix many years ago, when a macro exited in 1.4, it looks in + all calling contexts, bubbling up from the deepest level until it + finds an "h" extension. Since AEL hides the complexity of the + underlying dialplan logic from the AEL programmer, it's + reasonable to assume that this behavior should not change in the + transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we + break working AEL configurations in the transition to Asterisk + 1.8 LTS. This fix is the result, which implements a search for + the "h" extension in all calling Gosub contexts. Fixes + ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff + (License #5003) by Tilghman Lesher (with slight modifications for + 1.8) Tested by: Johan Wilfer Review: + https://reviewboard.asterisk.org/r/1776/ ........ Merged + revisions 358810 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358811 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-12 17:01 +0000 [r358766] Igor Goncharovskiy + + * channels/chan_unistim.c, contrib/unistimLang/ru.po (added), + contrib/unistimLang/ru.po.utf8 (added), + configs/unistim.conf.sample, UPGRADE.txt, CHANGES, + contrib/unistimLang/en.po (added), contrib/unistimLang (added): + Massive changes in chan_unistim channel driver. Include many + fixes in channel driver operation and add additional + functionality: * Added ability to use multiple lines on phone, so + for one device in configuration multiple lines can be defined, it + allows to have multiple calls on one phone, callwaiting and + switching between calls. * Added ability for translation + on-screen menu to multiple languages. Tested on Russian + languages. Supported encodings: ISO 8859-1, ISO 8859-2, ISO + 8859-4, ISO 8859-5, ISO 2022-JP. Language controlled by + 'language' and on-screen menu of phone * Other described in + CHANGES file Testing done by issue tracker users: ibercom, + scsiborg, idarwin, TeknoJuce, c0rnoTa. Tested on production + system by Jonn Taylor (jonnt) using phone models: Nortel i2004, + 1120E and 1140E. (closes issue ASTERISK-16890) Review: + https://reviewboard.asterisk.org/r/1243/ + +2012-03-10 20:06 +0000 [r358730] Joshua Colp + + * configs/confbridge.conf.sample, main/dial.c, apps/app_page.c, + apps/confbridge/include/confbridge.h, apps/app_confbridge.c, + include/asterisk/dial.h, CHANGES, + apps/confbridge/conf_config_parser.c: Transition app_page to + using app_confbridge internally for the conference bridge portion + of paging. This also adds a new 'announcement' option to + ConfBridge user profiles. Review: + https://reviewboard.asterisk.org/r/1754/ + +2012-03-08 17:48 +0000 [r358646-358691] Sean Bright + + * apps/app_dial.c, apps/app_directory.c, apps/app_queue.c: Resolve + a few more cases of variable shadowing. + + * channels/chan_phone.c, channels/chan_skinny.c, + channels/chan_agent.c, pbx/pbx_lua.c, pbx/pbx_dundi.c, + channels/chan_gtalk.c, pbx/pbx_config.c, channels/chan_oss.c, + apps/confbridge/conf_config_parser.c: Eliminate a bunch of shadow + warnings. + + * include/asterisk/linkedlists.h: Add some underscores in a few of + our llist macros to reduce name collisions. + +2012-03-08 16:59 +0000 [r358645] Jonathan Rose + + * /, channels/chan_sip.c: Make transfer not ignore port information + with SIP. Attempting to transfer with SIP to an address like + 1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from + the host string and ignored. This simply keeps chan_sip from + cutting off the port number during these kinds of transfers. + (closes issue ASTERISK-19321) Reported by: Federico Alves Review: + https://reviewboard.asterisk.org/r/1790/diff/#index_header + ........ Merged revisions 358643 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358644 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-08 16:21 +0000 [r358609-358622] Sean Bright + + * Makefile, configure, configure.ac, makeopts.in: Add + --enable-dev-mode=strict to configure. Passing -Wshadow to gcc + enables shadow warnings. From the gcc manual: Warn whenever a + local variable or type declaration shadows another variable, + parameter, type, or class member (in C++), or whenever a built-in + function is shadowed. Asterisk will not currently compile with + this option set, but a number of bugs have been discovered by + enabling this flag on specific files. The long-term goal is to + eliminate all of the suspect code that causes this warning to be + emitted. + + * Makefile: Whitespace only change to the Makefile + +2012-03-07 21:28 +0000 [r358576] Terry Wilson + + * cel/cel_odbc.c, configs/cel_odbc.conf.sample: Handle numeric + columns for eventtype properly in cel_odbc Patch also implements + correct handling of datetime2 and datetimeoffset new datatypes in + SQL Server 2008 and 2008 R2. (closes issue ASTERISK-17548) + Review: https://reviewboard.asterisk.org/r/1160/ Review: + https://reviewboard.asterisk.org/r/1804/ + +2012-03-07 18:33 +0000 [r358532] Richard Mudgett + + * /, channels/sig_ss7.c: Change directly setting _softhangup in + sig_ss7.c to use ast_softhangup_nolock(). Update to: (issue + ASTERISK-19372) ........ Merged revisions 358530 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358531 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-07 16:16 +0000 [r358486] Sean Bright + + * /, codecs/codec_dahdi.c: Return g729 and g723.1 frames with the + number of samples set properly. If the wctc4xxp returns more than + a single packet, we need to update the number of samples in the + returned frame accordingly. Acked-by: Shaun Ruffell + ........ Merged revisions 358484 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358485 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-07 15:19 +0000 [r358437-358444] Terry Wilson + + * /, configs/cdr_adaptive_odbc.conf.sample: Set snarkiness = 0 in + cdr_adaptive_odbc.conf.sample ........ Merged revisions 358438 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 358441 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * cel/cel_odbc.c, /, cdr/cdr_adaptive_odbc.c: Add detection for + ODBC WCHAR fields Without detecting these types, cel_odbc blows + up when the character set for the table is utf8. This also wraps + cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR + #ifdef seen in other parts of the code. ........ Merged revisions + 358435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 358436 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-06 17:47 +0000 [r358262-358379] Richard Mudgett + + * channels/chan_dahdi.c, /: Fix ring cadance setup for outgoing + calls on FXS ports. * Fix referencing the wrong variable in + chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for + compiling with -Wshadow and finding this bug. ........ Merged + revisions 358377 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358378 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, CHANGES: + Add dialtone_detect option for analog incoming calls. For analog + lines, enables Asterisk to use dialtone detection per channel if + an incoming call was hung up before it was answered. If dialtone + is detected, the call is hung up. no: Disabled. (Default) yes: + Look for dialtone for 10000 ms after answer. : Look for + dialtone for the specified number of ms after answer. always: + Look for dialtone for the entire call. Dialtone may return if the + far end hangs up first. dialtone_detect=yes dialtone_detect=5000 + dialtone_detect=always (closes issue ASTERISK-19316) Reported by: + Jeremy Pepper Patch by: Jeremy Pepper Tested by: rmudgett,Jeremy + Pepper Review: https://reviewboard.asterisk.org/r/1737/ + + * /, channels/sig_ss7.c: Drop SS7 call if not connected yet when + INCOMPLETE/BUSY/CONGESTION. SS7 is a trunk protocol and should + clear a failed call as soon as possible. * Made SS7 hangup a call + immediately if it has not connected yet for + INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate + inband tone. (closes issue ASTERISK-19372) Reported by: Igor + Nikolaev ........ Merged revisions 358278 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358284 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * include/asterisk/channel.h: Make usage of + DECLARE_STRINGFIELD_SETTERS_FOR() not look so odd. + + * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c: + Setup DSP when SS7 call is connected or early media is available. + Outgoing SS7 calls fail to detect incoming DTMF so any bridged + channel that requires out-of-band DTMF will not work. * Added + sig_ss7_open_media() calls at appropriate places in sig_ss7.c. + The new call converts conditionaled out unconverted code and + shows that the code really did something useful. * Improved some + chan_dahdi DTMF debug messages to help track DTMF handling. + (closes issue ASTERISK-19312) Reported by: Igor Nikolaev ........ + Merged revisions 358260 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358261 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-05 19:06 +0000 [r358216] Jonathan Rose + + * main/manager.c, /: Eliminate double close of file descriptor in + manager.c The process_output function in manager.c attempted to + call fclose and close immediately afterwards. Since fclose + implies close, this resulted in a potential double free on file + descriptors. This patch changes that behavior and also adds error + checking to fclose and close depending on which was deemed + necessary. Also error messages. Thanks to Rosen Iliev for + pointing out the location of the problem. (closes issue + ASTERISK-18453) Reported By: Jaco Kroon Review: + https://reviewboard.asterisk.org/r/1793/ ........ Merged + revisions 358214 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358215 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-05 16:44 +0000 [r358164] Joshua Colp + + * /, channels/chan_sip.c: Defer sending the connected line reinvite + if a reinvite is already in progress. (issue ASTERISK-19355) + Reported by: tomaso (closes issue AST-825) ........ Merged + revisions 358162 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358163 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-05 16:00 +0000 [r358117] Kinsey Moore + + * /, channels/chan_sip.c: Ensure Asterisk acknowledges ACKs to 4xx + on Replaces errors Asterisk was not setting pendinginvite in the + upper half of handle_request_invite such that the 4xx was + retransmitted repeatedly even though an ack was received for + every retransmission. (closes issue ASTERISK-19303) Reported by: + Jon Tsiros Patches: fix-19303.patch uploaded by Jeremiah Gowdy + (license 6358) ........ Merged revisions 358115 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358116 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-05 11:20 +0000 [r358082] Sean Bright + + * configs/iax.conf.sample: Tab to spaces and text change. + +2012-03-02 23:29 +0000 [r357999-358038] Terry Wilson + + * channels/chan_usbradio.c, /, channels/xpmr/xpmr.c: Fix + unused-but-set-variable warnings All of these were pretty + obviously unused. Some were unused because the code that used + them was #if 0'd. In those cases, I just commented out the + unused-but-set variables. ........ Merged revisions 358029 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358033 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /: Correct some set-but-unused variable warnings in the mISDN + library. (from kpfleming's commit to trunk r356292) ........ + Merged revisions 358011 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 358017 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/xpmr/xpmr.c: Make chan_usbradio compile under dev + mode x=++x and x=x=1? Really? ........ Merged revisions 357986 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 357987 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-02 21:06 +0000 [r357942] Kinsey Moore + + * /, main/ccss.c, tests/test_event.c, main/event.c, + include/asterisk/strings.h: Fix case-sensitivity for + device-specific event subscriptions and CCSS This change fixes + case-sensitivity for device-specific subscriptions such that the + technology identifier is case-insensitive while the remainder of + the device string is still case-sensitive. This should also + preserve the original case of the device string as passed in to + the event system. CCSS is the only feature affected as it is the + only consumer of device-specific event subscriptions. The second + part of this patch addresses similar case-sensitivity issues + within CCSS itself that prevented it from functioning correctly + after the fix to the events system. This adds a unit test to + verify that the event system works as expected. (closes issue + ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/ + ........ Merged revisions 357940 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357941 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-02 18:38 +0000 [r357896] Richard Mudgett + + * main/channel.c, /, channels/sig_pri.c: Remove ISDN hold + restriction for non-bridged calls. The check if an ISDN call is + bridged before it could be placed on hold is not necessary and is + overly restrictive. The check was originally done to prevent + problems with call transfers in case a user tried to transfer a + call connected to an application to another call connected to an + application. The ISDN transfer code has not required this + restriction for quite some time because ECT could transfer any + two active calls to each other. * Remove ISDN hold restriction + for calls connected to applications. * Made + ast_waitfordigit_full() ignore AST_CONTROL_HOLD and + AST_CONTROL_UNHOLD instead of generating a warning message. + (closes issue ASTERISK-19388) Reported by: Birger Harzenetter + Tested by: rmudgett ........ Merged revisions 357894 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357895 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-02 16:57 +0000 [r357861] Jonathan Rose + + * apps/app_queue.c: Adds a transfer callee on hangup option (like + with Dial option F) to queues. This should (and does in my + testing) act just like the Dial option of the same name. This + allows a queue member to be transfered to the next priority (no + args), or to a context/extension/priority similar to goto (with + args context^extension^priority) when a caller hangs up on them. + (closes issue ASTERISK-19283) Reported by: To Patches: + queue_f-v3.diff uploaded by To (license 6347) Review: + https://reviewboard.asterisk.org/r/1785/ + +2012-03-02 16:26 +0000 [r357834] Richard Mudgett + + * apps/app_chanspy.c: Remove bad usage of goto in ChanSpy + next_channel(). + +2012-03-02 16:19 +0000 [r357821] Sean Bright + + * configs/iax.conf.sample: Beef up the IAX2 sample configuration a + bit and fix some formatting issues. + +2012-03-02 16:03 +0000 [r357814-357815] Richard Mudgett + + * /, apps/app_chanspy.c: Fix channel reference leak in ChanSpy. * + Fix next_channel() channel reference leak in ChanSpy. (closes + issue ASTERISK-19461) Reported by: Irontec Patches: + app_chanspy_iteartor_next_unref.patch (license #6213) patch + uploaded by Irontec (issue ASTERISK-17515) ........ Merged + revisions 357809 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357810 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_usbradio.c: Fix compile error from latest channel + opaquification change. + +2012-03-02 16:00 +0000 [r357813] Sean Bright + + * /, channels/chan_iax2.c: The default value for mohinterpret is + the empty string, so when resetting to default values don't + explicitly set the value to "default." ........ Merged revisions + 357811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 357812 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-03-02 01:33 +0000 [r357774-357775] Mark Michelson + + * main/channel.c, /: Fix race condition that can cause important + control frames (such as a hangup) to be missed. This takes two + actions. 1. Move the reading of the alertpipe in __ast_read() to + immediately before the removal of frames from the readq. This + means we won't do something silly like read from the alertpipe, + then ignore the fact that there's a frame to get from the readq + since channel's fdno is the AST_TIMING_FD. 2. When + ast_settimeout() sets the rate to 0 and the timingfunc to NULL, + if the channel's fdno is the AST_TIMING_FD, then set the fdno to + -1. This is because if the rate is 0 and the timingfunc is NULL, + it means that the channel's timing fd is being invalidated, so + any pending reads should not occur. This may actually solve more + issues than the referenced one below, but it's not known at this + time for sure. (closes issue ASTERISK-19223) reported by + Frank-Michael Wittig Review: + https://reviewboard.asterisk.org/r/1779 ........ Merged revisions + 357761 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 357762 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_dahdi.c: Fix compilation error due to typo during + channel opaquification. + s/ast_channel_fd_set/ast_channel_internal_fd_set/g + +2012-03-01 22:09 +0000 [r357721] Terry Wilson + + * channels/chan_unistim.c, apps/app_dahdibarge.c, + main/autoservice.c, addons/chan_ooh323.c, channels/chan_vpb.cc, + apps/app_meetme.c, channels/console_video.c, + channels/chan_gtalk.c, channels/chan_iax2.c, main/cli.c, + main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c, + channels/sig_analog.c, channels/chan_skinny.c, main/features.c, + apps/app_dumpchan.c, channels/sig_ss7.c, channels/chan_mgcp.c, + main/pbx.c, channels/chan_sip.c, main/channel_internal_api.c, + channels/chan_agent.c, apps/app_dahdiras.c, + include/asterisk/channel.h, apps/app_queue.c, channels/sig_pri.c, + channels/chan_jingle.c, channels/chan_misdn.c, apps/app_flash.c, + funcs/func_channel.c, apps/app_directed_pickup.c, main/file.c, + channels/chan_h323.c, res/snmp/agent.c, main/dsp.c: Opaquify + ast_channel typedefs, fd arrays, and softhangup flag Review: + https://reviewboard.asterisk.org/r/1784/ + +2012-03-01 14:22 +0000 [r357673] Kinsey Moore + + * /, main/acl.c: Prevent outbound SIP NOTIFY packets from + displaying a port of 0 In the change from 1.6.2 to 1.8, + ast_sockaddr was introduced which changed the behavior of + ast_find_ourip such that port number was wiped out. This caused + the port in internip (which is used for Contact and Call-ID on + NOTIFYs) to be 0. This change causes ast_find_ourip to be + port-preserving again. (closes issue ASTERISK-19430) ........ + Merged revisions 357665 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357667 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-29 20:41 +0000 [r357621] Walter Doekes + + * /, main/utils.c, include/asterisk/stringfields.h: Update + stringfield documentation for removed second va_list in favor of + va_copy. In r320946, the second va_list that was passed to + ast_string_field_build_va and friends, was removed. This patch + updates the documentation to reflect that. ........ Merged + revisions 357620 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-29 20:31 +0000 [r357610] Sean Bright + + * res/res_agi.c, CHANGES: Add IPv6 support to FastAGI. Review: + https://reviewboard.asterisk.org/r/1774/ Reviewed by: Simon + Perreault, Mark Michelson + +2012-02-29 19:48 +0000 [r357577] Walter Doekes + + * apps/app_dial.c, /: Fix copying of CDR(accountcode) to local + channels. In r203638, during the addition of the Channel Event + Logging, in mid-2009, this got broken in trunk and ended up in + asterisk 1.8 and higher. This fixes so the CDR(accountcode) from + the calling channel is available to dialed channels again as well + as showing up properly in the CDR's. (closes issue + ASTERISK-19384) Reported by: jamicque Patches: accountcode.patch + (License #6033) by jamicque Review: + https://reviewboard.asterisk.org/r/1775/ Reviewed by: Richard + Mudgett ........ Merged revisions 357575 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357576 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-29 16:52 +0000 [r357542] Terry Wilson + + * channels/chan_local.c, addons/chan_ooh323.c, + funcs/func_strings.c, channels/console_video.c, + apps/app_alarmreceiver.c, channels/chan_iax2.c, main/cli.c, + channels/chan_dahdi.c, channels/sig_analog.c, + channels/chan_skinny.c, apps/app_dumpchan.c, main/features.c, + apps/app_amd.c, channels/sig_ss7.c, apps/app_dial.c, main/pbx.c, + include/asterisk/utils.h, funcs/func_timeout.c, + apps/app_privacy.c, apps/app_fax.c, channels/chan_agent.c, + apps/app_disa.c, include/asterisk/channel.h, + apps/app_talkdetect.c, main/cel.c, channels/chan_misdn.c, + apps/app_macro.c, apps/app_zapateller.c, apps/app_mixmonitor.c, + apps/app_voicemail.c, channels/chan_unistim.c, + tests/test_substitution.c, channels/chan_vpb.cc, + apps/app_meetme.c, main/ccss.c, apps/app_readexten.c, + channels/chan_gtalk.c, main/autochan.c, apps/app_followme.c, + main/cdr.c, main/channel.c, main/dial.c, channels/chan_phone.c, + apps/app_osplookup.c, apps/app_setcallerid.c, main/manager.c, + bridges/bridge_builtin_features.c, apps/app_minivm.c, + res/res_agi.c, main/app.c, apps/app_confbridge.c, apps/app_rpt.c, + main/message.c, channels/chan_mgcp.c, apps/app_parkandannounce.c, + apps/app_while.c, funcs/func_dialplan.c, channels/chan_sip.c, + res/res_fax.c, main/channel_internal_api.c, pbx/pbx_lua.c, + channels/chan_console.c, channels/sig_pri.c, apps/app_queue.c, + channels/chan_oss.c, channels/chan_jingle.c, + channels/chan_usbradio.c, funcs/func_blacklist.c, + main/abstract_jb.c, channels/chan_h323.c, main/file.c, + res/snmp/agent.c, apps/app_sms.c, apps/app_stack.c, + funcs/func_callerid.c: Opaquify ast_channel structs and lists + Review: https://reviewboard.asterisk.org/r/1773/ + +2012-02-28 22:31 +0000 [r357460-357503] Jonathan Rose + + * /, configs/sip.conf.sample, UPGRADE-1.8.txt: Adding transport=udp + to sample sip.conf - Also changes version of Asterisk 1.8 in + UPGRADE (issue ASTERISK-19352) Reported by: jamicque Patches: + asterisk-19352-transport-warning-message-v1.patch uploaded by + Michael L. Young (license 5026) ........ Merged revisions 357490 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 357497 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, cdr/cdr_adaptive_odbc.c: Add additional character type types + to supported data types for cdr_adaptive_odbc The reporter was + uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so + this patch adds those along with some other character types to + the list of types cdr_adaptive_odbc will work using the varchar + conditions. The problem wasn't really UTF8 characters as much as + it was a failure to respond to the exact type that was + declared/in use on that database. (closes issue ASTERISK-19334) + Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch + uploaded by Igor Nikolaev (license 6236) ........ Merged + revisions 357455 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357458 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-28 21:26 +0000 [r357436] Tilghman Lesher + + * /, apps/app_stack.c: Correctly reset the dialplan priority. When + the stack frame is allocated, we save the address to which we + should return, when the Gosub returns. However, if we just want + to restore the priority, then we need to subtract 1 before + setting it. Otherwise, when a Gosub goes to a nonexistent + address, it will skip a priority in the dialplan. This is because + when we return from an application, the PBX increments the + priority for us. ........ Merged revisions 357416 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357421 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-28 21:01 +0000 [r357409] Richard Mudgett + + * /, channels/sig_pri.c: Use more reasonable cause code when + rejecting incoming call waiting calls. (closes issue + ASTERISK-19397) Reported by: Birger Harzenetter Patches: + nochannel-cause.patch (license #5870) patch uploaded by Birger + Harzenetter ........ Merged revisions 357407 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357408 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-28 20:43 +0000 [r357406] Jonathan Rose + + * /, UPGRADE-10.txt: revision 357386 -- oops, accidentally made it + 10.3 to 10.4 instead of 10.2 to 10.3 (issue ASTERISK-19352) + reported by: jamicque ........ Merged revisions 357405 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-28 20:34 +0000 [r357404] Richard Mudgett + + * main/channel.c, res/res_musiconhold.c, apps/app_queue.c: Fix + REF_DEBUG compile errors. + +2012-02-28 20:33 +0000 [r357358-357403] Jonathan Rose + + * /, UPGRADE-10.txt, UPGRADE-1.8.txt: Moves UPGRADE.txt notes from + r357356 to a new section specific to 1.8.12 (issue + ASTERISK-19352) reported by: jamicque ........ Merged revisions + 357386 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 357400 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, UPGRADE-1.8.txt: Adds UPGRADE.txt notes to r357266 indicating + changes to transport option (issue ASTERISK-19352) Reported by: + jamicque ........ Merged revisions 357356 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357357 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-28 19:55 +0000 [r357355] Sean Bright + + * include/asterisk/netsock2.h: Documentation update. There is no + AST_SOCKADDR_UNSPEC. + +2012-02-28 19:37 +0000 [r357354] Richard Mudgett + + * /, apps/app_page.c: Remove dupliate 'i' option table entry in + app_page.c. (closes issue ASTERISK-19310) Reported by: Makoto Dei + Patches: app_page-duplicate-i-option.patch (license #5027) patch + uploaded by Makoto Dei ........ Merged revisions 357352 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357353 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-28 18:52 +0000 [r357319] Mark Michelson + + * /, channels/sip/security_events.c: Add a security event for the + case where fake authentication challenge is sent. ........ Merged + revisions 357318 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-28 18:46 +0000 [r357317] Richard Mudgett + + * main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h: + Convert struct ast_tcptls_session_instance to finally use the ao2 + object lock. + +2012-02-28 18:23 +0000 [r357288] Jonathan Rose + + * /, channels/chan_sip.c: Changes transport option in sip.conf so + that using multiple instances doesn't stack. Prior to this patch, + Using "transport=" multiple times would cause them to add to one + another like allow/deny. This patch changes that behavior to + simply use the transport option specified last. Also, if no + transport option is applied now, the default will automatically + be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches: + asterisk-19352-transport-warning-message-v1.patch uploaded by + Michael L. Young (license 5026) + issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes + (license 5674) Review: + https://reviewboard.asterisk.org/r/1745/diff/#index_header + ........ Merged revisions 357266 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357271 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-28 18:15 +0000 [r357272] Richard Mudgett + + * main/format.c, main/format_cap.c, include/asterisk/astobj2.h, + include/asterisk/lock.h, main/astobj2.c: Astobj2 locking + enhancement. Add the ability to specify what kind of locking an + ao2 object has when it is allocated. The locking could be one of: + MUTEX, RWLOCK, or none. New API: ao2_t_alloc_options() + ao2_alloc_options() ao2_t_container_alloc_options() + ao2_container_alloc_options() ao2_rdlock() ao2_wrlock() + ao2_tryrdlock() ao2_trywrlock() The OBJ_NOLOCK and + AO2_ITERATOR_DONTLOCK flags have a slight meaning change. They no + longer mean that the object is protected by an external + mechanism. They mean the lock associated with the object has + already been manually obtained by one of the ao2_lock calls. This + change is necessary for RWLOCK support since they are not + reentrant. Also an operation on an ao2 container may require + promoting a read lock to a write lock by releasing the already + held read lock to re-acquire as a write lock. Replaced API calls: + ao2_t_link_nolock() ao2_link_nolock() ao2_t_unlink_nolock() + ao2_unlink_nolock() with the respective ao2_t_link_flags() + ao2_link_flags() ao2_t_unlink_flags() ao2_unlink_flags() API + calls to be more flexible and to allow an anticipated enhancement + to control linking duplicate objects into a container. The + changes to format.c and format_cap.c are taking advantange of the + new ao2 locking options to simplify the use of the format + capabilities containers. Review: + https://reviewboard.asterisk.org/r/1554/ + +2012-02-28 14:47 +0000 [r357178-357214] Kevin P. Fleming + + * /, Makefile.rules: Make COMPILE_DOUBLE magic actually work. The + build system has some special magic to ensure that if Asterisk is + built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the + source is still compiled with the optimizer enabled (even though + the result will be thrown away), because the compiler is able to + find a great deal of coding errors and bugs as a result of + running its optimizers. Unfortunately at some point this mode got + broken, and the 'throwaway' compile of the code was no longer + done with the optimizer enabled. This patch corrects that + problem. ........ Merged revisions 357212 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 357213 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/astobj2.c: Trailing whitespace cleanup. + +2012-02-28 00:42 +0000 [r357096-357145] Richard Mudgett + + * include/asterisk/astobj2.h, tests/test_astobj2.c, main/astobj2.c: + Add ability to clone ao2 containers. Occasionally there is a need + to put all objects in one container also into another container. + Some reasons you might need to do this: 1) You need to + reconfigure a container. You would do this by creating a new + container with the new configuration and ao2_container_dup the + old container into it. Then replace the old container with the + new. Then destroy the old container. 2) You need the contents of + a container to remain stable while operating on all of the + objects. You would do this by creating a cloned container of the + original with ao2_container_clone. The cloned container is a + snapshot of the objects at the time of the cloning. When done, + just destroy the cloned container. Review: + https://reviewboard.asterisk.org/r/1746/ + + * main/channel.c: Fix ast_channel allocation init setting priority + to -1 instead of 1. * Fix opaquification conversion error. + (closes issue ASTERISK-19424) Reported by: Jeremy Pepper Patches: + asterisk-19424-initialize_priority_regression.diff (license + #5026) patch uploaded by Michael L. Young + + * main/channel.c, /: Fix callerid of Originated calls. Thanks to + Matt Riddell for tracking this down. (closes issue + ASTERISK-19385) Reported by: ornix ........ Merged revisions + 357093 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 357095 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-27 19:55 +0000 [r357051] Jonathan Rose + + * include/asterisk/res_odbc.h, res/res_odbc.c: Converts locking for + odbc containers from ast_mutex_lock to ao2_locks. + +2012-02-27 17:03 +0000 [r357014] Sean Bright + + * channels/chan_iax2.c, main/netsock.c: Address comments from Mark + Michelson + +2012-02-27 16:50 +0000 [r357013] Kinsey Moore + + * apps/app_dial.c, main/channel.c, include/asterisk/app.h, + main/dial.c, main/rtp_engine.c, main/ccss.c, main/features.c, + UPGRADE.txt, main/app.c, include/asterisk/channel.h, + configs/ccss.conf.sample, apps/app_followme.c, apps/app_queue.c, + include/asterisk/ccss.h: Deprecated macro usage for connected + line, redirecting, and CCSS This commit adds GoSub alternatives + to connected line, redirecting, and CCSS macro hooks so that + macro can finally be deprecated. This also adds deprecation + warnings for those features when used and in documentation. + Review: https://reviewboard.asterisk.org/r/1760/ (closes issue + SWP-4256) + +2012-02-27 16:31 +0000 [r357005] Sean Bright + + * include/asterisk/netsock.h, channels/chan_iax2.c, main/netsock.c: + Convert netsock.h over to use ast_sockaddrs rather than + sockaddr_in and update chan_iax2 to pass in the correct types. + chan_iax2 is the only consumer for the various ast_netsock_* + functions in trunk at this point, so this feels like a safe + change to make. + +2012-02-27 16:24 +0000 [r356987] Jonathan Rose + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/sip/include/sip.h: Adds an option to sip.conf that + prevents diversion headers from being added. send_diversion=no + will prevent Diversion headers from being added to SIP requests. + This doesn't prevent Diversion from being added with dialplan + such as with SIPAddHeader. (closes issue ASTERISK-16862) Reported + by: rsw686 Review: https://reviewboard.asterisk.org/r/1769/ + +2012-02-27 16:12 +0000 [r356966] Sean Bright + + * channels/chan_iax2.c: There isn't much point in saving off and + restoring a value that we never use again. + +2012-02-27 16:08 +0000 [r356965] Terry Wilson + + * /, main/features.c: Copy CDR variables when set during a bridge + This patch makes sure amaflags, accountcode, and userfield get + copied to the bridge CDR when set during a bridge (like via a + custom feature). (closes issue ASTERISK-16990) Review: + https://reviewboard.asterisk.org/r/1721/ ........ Merged + revisions 356963 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 356964 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-27 15:35 +0000 [r356962] Jonathan Rose + + * /, res/res_odbc.c: Remove possible segfaults from res_odbc by + adding locks around usage of odbc handle (closes issue + ASTERISK-19011) Reported by: Walter Doekes Patches: + issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch + uploaded by Walter Doekes (license 5674) review: + https://reviewboard.asterisk.org/r/1719/ review: + https://reviewboard.asterisk.org/r/1622/ ........ Merged + revisions 356917 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 356961 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-27 14:57 +0000 [r356881-356916] Sean Bright + + * include/asterisk/netsock.h, main/netsock.c: Make + ast_netsock_set_qos() delegate to ast_set_qos(). + + * include/asterisk/netsock.h: Correct typo in deprecation comment. + + * channels/chan_unistim.c, main/udptl.c, channels/chan_skinny.c, + include/asterisk/netsock.h, pbx/pbx_dundi.c, + channels/chan_mgcp.c: Prefer ast_set_qos() over + ast_netsock_set_qos() + + * main/netsock.c: Remove trailing whitespace + +2012-02-26 18:25 +0000 [r356848] Alexandr Anikin + + * addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c: Add + support change gatekeeper mode or ip per ooh323 reload command + (issue ASTERISK-19298) Reported by: Dmitry Melekhov Patches: + change_gk_on_reload-1.patch (License #5415) + +2012-02-25 17:22 +0000 [r356799] Matthew Jordan + + * /, apps/app_voicemail.c: Fix crash in app_voicemail during + close_mailbox In r354890, a memory leak in app_voicemail was + fixed by properly disposing of the allocated heard/deleted + pointers. However, there are situations, particularly when no + messages are found in a folder, where these pointers are not + allocated and not NULL. In that case, an invalid free would be + attempted, which could crash app_voicemail. As there are a number + of code paths where this could occur, this patch uses the number + of messages detected in the folder before it attempts to free the + pointers. This resolves the crash detected in the Asterisk Test + Suite's check_voicemail_nominal test. ........ Merged revisions + 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 356798 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-24 23:40 +0000 [r356697-356765] Richard Mudgett + + * include/asterisk/astobj2.h: astobj2.h comment tweaks. + + * include/asterisk/astobj2.h, main/astobj2.c: astobj2.h + documentation updates. + + * /, channels/chan_sip.c, include/asterisk/tcptls.h, + channels/sip/include/sip.h: Fix worker thread resource leak in + SIP TCP/TLS. The SIP TCP/TLS worker threads were created joinable + but noone could join them if they died on their own. * Fix the + SIP TCP/TLS worker threads to not be created joinable. * + _sip_tcp_helper_thread() only needs one parameter since the pvt + parameter is only passed in as NULL and never used. (closes issue + ASTERISK-19203) Reported by: Steve Davies Review: + https://reviewboard.asterisk.org/r/1714/ ........ Merged + revisions 356677 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 356690 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-24 17:43 +0000 [r356606-356652] Matthew Jordan + + * /, res/res_srtp.c: Remove srtp_shutdown from res_srtp The patch + for ASTERISK-19253 included properly shutting down the libsrtp + library in the case of module unload. Unfortunately, not all + distributions have the srtp_shutdown call. As such, this patch + removes calling srtp_shutdown. ........ Merged revisions 356650 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 356651 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/sip/sdp_crypto.c, include/asterisk/res_srtp.h, + main/rtp_engine.c, /, include/asterisk/rtp_engine.h, + res/res_srtp.c: Allow SRTP policies to be reloaded Currently, + when using res_srtp, once the SRTP policy has been added to the + current session the policy is locked into place. Any attempt to + replace an existing policy, which would be needed if the remote + endpoint negotiated a new cryptographic key, is instead rejected + in res_srtp. This happens in particular in transfer scenarios, + where the endpoint that Asterisk is communicating with changes + but uses the same RTP session. This patch modifies res_srtp to + allow remote and local policies to be reloaded in the underlying + SRTP library. From the perspective of users of the SRTP API, the + only change is that the adding of remote and local policies are + now added in a single method call, whereas they previously were + added separately. This was changed to account for the differences + in handling remote and local policies in libsrtp. Review: + https://reviewboard.asterisk.org/r/1741/ (closes issue + ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas + Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt + Jordan (license 6283) (with some small modifications for this + check-in) ........ Merged revisions 356604 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 356605 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-24 00:32 +0000 [r356573] Terry Wilson + + * channels/chan_unistim.c, channels/chan_local.c, + addons/chan_ooh323.c, channels/chan_multicast_rtp.c, + channels/chan_vpb.cc, main/rtp_engine.c, apps/app_meetme.c, + apps/app_dictate.c, apps/app_record.c, apps/app_test.c, + bridges/bridge_softmix.c, channels/chan_gtalk.c, apps/app_ices.c, + res/res_musiconhold.c, channels/chan_iax2.c, + bridges/bridge_multiplexed.c, main/indications.c, main/cli.c, + main/channel.c, channels/chan_phone.c, channels/chan_dahdi.c, + channels/chan_skinny.c, res/res_agi.c, main/features.c, + apps/app_mp3.c, apps/app_dumpchan.c, main/app.c, apps/app_amd.c, + channels/chan_alsa.c, apps/app_confbridge.c, + addons/chan_mobile.c, main/bridging.c, channels/chan_mgcp.c, + apps/app_nbscat.c, main/pbx.c, channels/chan_sip.c, + res/res_fax.c, apps/app_festival.c, channels/chan_bridge.c, + main/channel_internal_api.c, apps/app_fax.c, + apps/app_waitforsilence.c, res/res_adsi.c, channels/chan_agent.c, + bridges/bridge_simple.c, include/asterisk/channel.h, + channels/chan_console.c, apps/app_talkdetect.c, + channels/chan_oss.c, apps/app_speech_utils.c, + channels/chan_usbradio.c, channels/chan_jingle.c, + channels/chan_misdn.c, funcs/func_channel.c, main/file.c, + channels/chan_nbs.c, apps/app_chanspy.c, apps/app_voicemail.c, + res/res_calendar.c: Opaquification for ast_format structs in + struct ast_channel Review: + https://reviewboard.asterisk.org/r/1770/ + +2012-02-23 20:14 +0000 [r356523] Richard Mudgett + + * /, channels/chan_sip.c, main/features.c: Fix blind transfer + parking issues if the dialed extension is not recognized as a + parking extension. Custom parking extensions may not be coded + such that the first and only extension priority is the Park + application. These custom parking extensions will not be + recognized as parking extensions. When a call is blind + transferred to an extension that is not recognized as a parking + extension, the normal blind transfer code causes the transferred + channel to start executing dialplan. Calls that get parked in + this manner do not know the original channel name that parked the + call so the original parker could never be called back if the + parked call is not retrieved before the timeout time. The parking + space is also announced to the call being parked as a side effect + of not knowing the original parking channel. * Fix handling of + BLINDTRANSFER channel variable for call parking. * Fixed SIP + blind transfer using the wrong dialplan context variable to check + for the parking extension. (closes issue ASTERISK-19322) Reported + by: aragon Tested by: rmudgett, jparker Review: + https://reviewboard.asterisk.org/r/1730/ JIRA AST-766 ........ + Merged revisions 356521 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 356522 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-23 15:49 +0000 [r356477] Mark Michelson + + * /, channels/chan_sip.c: Fix ACK routing for non-2xx responses. + When we send an ACK for a 2xx response to an INVITE, we are + supposed to use the learned route set. However, when we receive a + non-2xx final response to an INVITE, we are supposed to send the + ACK to the same place we initially sent the INVITE. We had been + doing this up until the changes went in that would build a route + set from provisional responses. That introduced a regression + where we would use the learned route set under all circumstances. + With this change, we now will set the destination of our ACK + based on the invitestate. If it is INV_COMPLETED then that means + that we have received a non-2xx final response (INV_TERMINATED + indicates a 2xx response was received). If it is INV_CANCELLED, + then that means the call is being canceled, which means that we + should be ACKing a 487 response. The other change introduced here + is setting the invitestate to INV_CONFIRMED when we send an ACK + *after* the reqprep instead of before. This way, we can tell in + reqprep more easily what the invitestate is prior to sending the + ACK. (closes issue ASTERISK-19389) reported by Karsten Wemheuer + patches: ASTERISK-19389v2.patch uploaded by Mark Michelson + (license #5049) (with some slight modifications prior to commit) + ........ Merged revisions 356475 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 356476 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-23 03:27 +0000 [r356429] Paul Belanger + + * /, apps/app_rpt.c: Multiple revisions 356290,356335,356337 + ........ r356290 | pabelanger | 2012-02-22 15:20:29 -0500 (Wed, + 22 Feb 2012) | 4 lines Fix -Werror=unused-but-set-variable + compiler error (gcc 4.6.2) Review: + https://reviewboard.asterisk.org/r/1763/ ........ r356335 | + pabelanger | 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) | 2 + lines Add back strsep() function for previous commit ........ + r356337 | pabelanger | 2012-02-22 16:36:37 -0500 (Wed, 22 Feb + 2012) | 2 lines Missed one strsep() function ........ Merged + revisions 356290,356335,356337 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 356428 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-23 01:53 +0000 [r356397] Terry Wilson + + * tests/test_substitution.c, tests/test_utils.c: Fix some tests + that didn't get opaquification changes Review: + https://reviewboard.asterisk.org/r/1766/ + +2012-02-23 00:56 +0000 [r356366] Richard Mudgett + + * main/channel_internal_api.c: Revert some apparently accidental + spacing changes. + +2012-02-22 21:22 +0000 [r356314] Terry Wilson + + * /, include/asterisk/calendar.h, main/loader.c, + res/res_calendar.c: Track module use count for res_calendar If + the res_calendar module was followed immediately by one of the + calendar tech modules and "core stop gracefully" was run, + Asterisk would crash. This patch adds use count tracking for + res_calendar so that it is unloaded after the tech modules when + shutting down gracefully. It is now not possible to unload all + the of the calendar modules via "module unload res_calednar.so", + but it is still possible to unload them all via "module unload -h + res_calendar.so". Review: + https://reviewboard.asterisk.org/r/1752/ ........ Merged + revisions 356291 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 356297 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-22 21:10 +0000 [r356292] Kevin P. Fleming + + * channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: + Correct some set-but-unused variable warnings in the mISDN + library. + +2012-02-22 17:34 +0000 [r356259] Terry Wilson + + * channels/chan_misdn.c: Fix chan_misdn after the lastest + opaquification changes It now compiles, but there are some + unrelated warnings for set but unused variables. + +2012-02-22 14:54 +0000 [r356216] Matthew Jordan + + * /, channels/chan_sip.c: Merged revisions 356215 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r356215 | mjordan | 2012-02-22 08:53:53 -0600 + (Wed, 22 Feb 2012) | 32 lines Merged revisions 356214 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012) + | 27 lines Fix potential buffer overrun and memory leak when + executing "sip show peers" The "sip show peers" command uses a + fix sized array to sort the current peers in the peers + ao2_container. The size of the array is based on the current + number of peers in the container. However, once the size of the + array is determined, the number of peers in the container can + change, as the peers container is not locked. This could cause a + buffer overrun when populating the array, if peers were added to + the container after the array was created. Additionally, a memory + leak of the allocated array would occur if a user caused the + _show_peers method to return CLI_SHOWUSAGE. We now create a + snapshot of the current peers using an ao2_callback with the + OBJ_MULTIPLE flag. This size of the array is set to the number of + peers that the iterator will iterate over; hence, if peers are + added or removed from the peers container it will not affect the + execution of the "sip show peers" command. Review: + https://reviewboard.asterisk.org/r/1738/ (closes issue + ASTERISK-19231) (closes issue ASTERISK-19361) Reported by: Thomas + Arimont, Jamuel Starkey Tested by: Thomas Arimont, Jamuel Starkey + Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan + (license 6283) ........ ................ + +2012-02-22 00:35 +0000 [r356152-356183] Terry Wilson + + * main/channel.c, main/channel_internal_api.c, + include/asterisk/channel.h: Rename + ast_channel_emulate_dtmf_digit* funcs The accessors names for the + "emulate_dtmf_digit" field on the ast_channel are misleading. + Change them to ast_channel_dtmf_digit_to_emulate*. + + * main/channel.c, main/framehook.c, res/res_monitor.c: Fix some + opaquification-related compiler warnings (closes issue + ASTERISK-19419) PseudoReview - seanbright on IRC + +2012-02-21 11:17 +0000 [r356111] Sean Bright + + * /, channels/chan_iax2.c: Make 'iax2 show callnumber usage' output + make sense when an IP is passed in. ........ Merged revisions + 356107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 356108 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-21 04:31 +0000 [r356075] Kinsey Moore + + * /, main/ccss.c: Add missing newline to ccss state change + notification Move along, nothing to see here... ........ Merged + revisions 356074 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-20 23:43 +0000 [r356042] Terry Wilson + + * main/udptl.c, apps/app_dahdibarge.c, addons/chan_ooh323.c, + cdr/cdr_sqlite3_custom.c, channels/chan_local.c, + main/rtp_engine.c, apps/app_playtones.c, apps/app_record.c, + apps/app_sayunixtime.c, apps/app_test.c, main/devicestate.c, + apps/app_alarmreceiver.c, apps/app_chanisavail.c, + apps/app_ices.c, channels/chan_iax2.c, + bridges/bridge_multiplexed.c, main/cli.c, channels/chan_dahdi.c, + channels/sig_analog.c, main/framehook.c, channels/chan_skinny.c, + main/features.c, apps/app_dumpchan.c, pbx/pbx_realtime.c, + channels/chan_alsa.c, apps/app_externalivr.c, main/bridging.c, + channels/sig_ss7.c, apps/app_milliwatt.c, cdr/cdr_manager.c, + apps/app_dial.c, main/pbx.c, funcs/func_timeout.c, + apps/app_privacy.c, channels/chan_bridge.c, apps/app_echo.c, + apps/app_softhangup.c, apps/app_fax.c, apps/app_dahdiras.c, + channels/chan_agent.c, apps/app_disa.c, bridges/bridge_simple.c, + include/asterisk/channel.h, apps/app_talkdetect.c, + apps/app_transfer.c, main/cel.c, res/res_monitor.c, + apps/app_playback.c, apps/app_speech_utils.c, + channels/chan_misdn.c, apps/app_sendtext.c, funcs/func_channel.c, + funcs/func_cdr.c, channels/sip/dialplan_functions.c, + apps/app_macro.c, apps/app_zapateller.c, main/audiohook.c, + apps/app_chanspy.c, apps/app_voicemail.c, apps/app_cdr.c, + res/res_calendar.c, channels/chan_unistim.c, + channels/chan_multicast_rtp.c, channels/chan_vpb.cc, + apps/app_meetme.c, main/ccss.c, apps/app_dictate.c, + apps/app_authenticate.c, apps/app_readexten.c, + channels/chan_gtalk.c, res/res_musiconhold.c, + apps/app_followme.c, main/channel.c, main/cdr.c, + channels/chan_phone.c, main/dial.c, main/manager.c, + apps/app_osplookup.c, bridges/bridge_builtin_features.c, + res/res_agi.c, apps/app_minivm.c, main/app.c, + apps/app_confbridge.c, main/image.c, apps/app_directory.c, + main/message.c, apps/app_ivrdemo.c, addons/chan_mobile.c, + apps/app_rpt.c, cdr/cdr_custom.c, apps/app_parkandannounce.c, + channels/chan_mgcp.c, apps/app_while.c, res/res_rtp_asterisk.c, + apps/app_read.c, channels/chan_sip.c, apps/app_festival.c, + res/res_fax.c, cdr/cdr_syslog.c, apps/app_waitforsilence.c, + main/channel_internal_api.c, res/res_adsi.c, pbx/pbx_lua.c, + funcs/func_jitterbuffer.c, channels/chan_console.c, + apps/app_queue.c, channels/sig_pri.c, channels/chan_oss.c, + channels/chan_jingle.c, channels/chan_usbradio.c, + apps/app_channelredirect.c, apps/app_forkcdr.c, apps/app_flash.c, + main/abstract_jb.c, main/file.c, channels/chan_h323.c, + include/asterisk/sched.h, res/snmp/agent.c, apps/app_sms.c, + channels/chan_nbs.c, funcs/func_callerid.c, apps/app_verbose.c, + apps/app_stack.c: ast_channel opaquification of pointers and + integral types Review: https://reviewboard.asterisk.org/r/1753/ + +2012-02-20 18:40 +0000 [r355903-355999] Sean Bright + + * /, channels/chan_iax2.c: Remove spurious warning when + 'qualifyfreqnotok' is set successfully. (closes issue + ASTERISK-17176) Reported by: John Covert Tested by: Sean Bright + Patches: chan_iax2.c.qualifyfreqnotok.patch uploaded by John + Covert (license 5512) ........ Merged revisions 355997 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355998 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_dahdi.c, /: This was a LOG_NOTICE, so roll it back. + ........ Merged revisions 355952 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355953 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_dahdi.c, /: Change some debug messages from + LOG_DEBUG to ast_debug. ........ Merged revisions 355949 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355950 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_iax2.c: Add some boilerplate documentation for + IAXVAR and IAXPEER. ........ Merged revisions 355904 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355905 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_iax2.c: Set the length of the ast_sockaddr, so + that we can set it's port later. Without this, the call to + ast_sockaddr_set_port a few lines later is a noop. ........ + Merged revisions 355901 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355902 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-18 08:02 +0000 [r355852] Alec L Davis + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_ss7.h, /, channels/sig_analog.h, channels/sig_pri.c, + channels/sig_ss7.c: push 'outgoing' flag from sig_XXX up to + chan_dahdi 'p->outgoing' in chan_dahdi and sig_analog wern't kept + in sync, particulary FXS ast_hangup didn't clear the 'outgoing' + flag. sig_pri and sig_ss7 were keeping 'outgoing' flag insync. + Now provides a callback for all the low level sig_XXX modules. + (issue ASTERISK-19316) alecdavis (license 585) Reported by: + Jeremy Pepper Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/1747/ ........ Merged + revisions 355850 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355851 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-17 22:03 +0000 [r355795] Sean Bright + + * configs/iax.conf.sample, /, channels/chan_iax2.c: Don't allow + trunkfreq to be greater than 1000ms. ........ Merged revisions + 355793 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 355794 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-17 19:56 +0000 [r355749] Tilghman Lesher + + * main/asterisk.c: Non-verbose output should always go to the + remote console, regardless of the previous level. + +2012-02-17 19:35 +0000 [r355748] Sean Bright + + * /, channels/chan_iax2.c: Pass the correct value to + ast_timer_set_rate() for IAX2 trunking. IAX2 uses the trunkfreq + variable to determine how often to send trunk packets, but this + value is in milliseconds while ast_timer_set_rate() expects the + rate argument to be ticks per second. So we divide 1000 by + trunkfreq and pass that in instead. With a default of 20ms, this + change makes IAX2 send trunk packets every 20ms instead of every + 50ms. Tracked down by myself and Bob Wienholt. ........ Merged + revisions 355746 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355747 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-17 19:22 +0000 [r355745] Mark Michelson + + * /, channels/chan_sip.c: Fix regressions with regards to route-set + creation on early dialogs. This fixes two main issues: 1. + Asterisk would send a CANCEL to the route created by the + provisional response instead of using the same destination it did + in the initial INVITE. 2. If a new route set arrives in a 200 OK + than was in the 1XX response (perfectly possible if our outbound + INVITE gets forked), then the route set in the 200 OK needs to + overwrite the route set in the 1XX response. (closes issue + ASTERISK-19358) Reported by: Karsten Wemheuer Tested by: Karsten + Wemheuer patches: ASTERISK-19358.patch uploaded by Mark Michelson + (license 5049) ASTERISK-19358.patch uploaded by Stefan Schmidt + (license 6034) Review: https://reviewboard.asterisk.org/r/1749 + ........ Merged revisions 355732 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355733 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-16 22:00 +0000 [r355667] Paul Belanger + + * apps/app_rpt.c: Fix channel opaquification for app_rpt + +2012-02-16 20:03 +0000 [r355624] Sean Bright + + * /, main/audiohook.c: Revert a change to + audio_audiohook_write_list that had no affect. When I made this + change initially, I was under the false impression that the + audiohooks structure remained on the channel after all of the + hooks had been detached. This is not the case, ast ast_read takes + care of removing the audiohooks structure if the lists are empty. + ........ Merged revisions 355622 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355623 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-16 19:51 +0000 [r355576-355621] Richard Mudgett + + * /, configure, include/asterisk/autoconfig.h.in, + autoconf/ast_c_declare_check.m4 (added), configure.ac, + formats/format_ogg_vorbis.c: Fix compile problem when old version + of libvorbisfile v1.1.2 is used. The principle difference between + libvorbisfile v1.1.2 and newer (at least v1.2.0) is the addition + of the predefined callbacks OV_CALLBACKS_xxx in + vorbis/vorbisfile.h used for ov_open_callbacks(). * Updated the + configure script to detect if libvorbisfile.h declares + OV_CALLBACKS_NOCLOSE. * Copied the declaration of + OV_CALLBACKS_NOCLOSE from v1.2.0 to allow v1.1.2 to compile. + (closes issue ASTERISK-19370) Reported by: Jonn Taylor ........ + Merged revisions 355608 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355620 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, res/res_monitor.c: Fix AMI Monitor action without File header + converting channel name into filename. * Fix potential Solaris + crash if Monitor application has a urlbase and no fname_base + option. ........ Merged revisions 355574 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355575 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-15 19:29 +0000 [r355450-355531] Sean Bright + + * /, channels/chan_iax2.c: When IAX2 debugging is enabled, make + sure to log 'apathetic' messages too. ........ Merged revisions + 355529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 355530 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * build_tools/cflags.xml, channels/chan_iax2.c: Remove IAX_OLD_FIND + from chan_iax2. + + * /, channels/chan_iax2.c: Use TRUNK_CALL_START as originally + intended. Back in r646, TRUNK_CALL_START was added and defined as + 0x4000. That same value was also hard-coded in one part of the + IAX2 code instead of using the #define. TRUNK_CALL_START has + changed over the years (for dealing with LOW_MEMORY), but the + hard-coded usage was never updated to match. This patch fixes + that. ........ Merged revisions 355448 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355449 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-14 20:27 +0000 [r355413] Tilghman Lesher + + * utils/refcounter.c, main/pbx.c, funcs/func_timeout.c, + include/asterisk/autoconfig.h.in, utils/hashtest.c, UPGRADE.txt, + CHANGES, main/config.c, configs/logger.conf.sample, + main/loader.c, include/asterisk/logger.h, main/manager.c, + main/logger.c, utils/ael_main.c, utils/hashtest2.c, + codecs/codec_dahdi.c, main/stdtime/localtime.c, main/asterisk.c, + addons/res_config_mysql.c: Re-commit the verbose branch. This + change permits each verbose destination (consoles, logger) to + have its own concept of what the verbosity level is. The big + feature here is that the logger will now be able to capture a + particular verbosity level without condemning each console to + need to suffer that level of verbosity. Additionally, a stray + 'core set verbose' will no longer change what will go to the log. + Review: https://reviewboard.asterisk.org/r/1599/ + +2012-02-14 19:29 +0000 [r355321-355376] Richard Mudgett + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + formats/format_ogg_vorbis.c: Fix voicemail problems when using + ogg/vorbis. Ogg/vorbis was fairly useless as a voicemail file + format because it did not implement the seek and tell format + callbacks among other problems. Since we were already using the + libvorbis and libvorbisenc libraries we can use libvorbisfile as + it is also part of the vorbis library package. * Made use the + libvorbisfile to handle the ogg/vorbis file stream. The + format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile. + (closes issue ASTERISK-16926) Reported by: sque Patches: + ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded + by sque ........ Merged revisions 355365 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355375 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, cel/cel_sqlite3_custom.c: Fix lock typo that should be unlock + in cel_sqlite_custom reload. (closes issue ASTERISK-19356) + Reported by: Alex Villacis Lasso Patches: + asterisk-1.8.9.2-cel_sqlite3_custom-fix-reload-locking-typo.patch + (license #5617) patch uploaded by Alex Villacis Lasso Review: + https://reviewboard.asterisk.org/r/1740/ ........ Merged + revisions 355319 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355320 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-14 16:28 +0000 [r355274] Mark Michelson + + * /, channels/chan_sip.c: Properly invert the return of a strncmp + call. This was causing identification that should have been made + private to be public. (closes issue AST-814) reported by Patrick + Anderson Patches: chan_sip.c.diff uploaded by Patrick Anderson + (license 5430) ........ Merged revisions 355268 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355271 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-14 15:58 +0000 [r355230] Jason Parker + + * /, configs/cdr_sqlite3_custom.conf.sample: Don't enable sqlite3 + CDRs by default in sample configs. ........ Merged revisions + 355228 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 355229 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-14 13:35 +0000 [r355184] Sean Bright + + * /, channels/chan_iax2.c: Clear the high order bit from the + destination call number before sending. send_apathetic_reply + takes the incoming frame's source call number as the destination + call number for the outgoing frame. If the incoming frame was a + full frame, then the high order bit of the source call number is + set and will be interpreted as a retransmit when sent back out as + the destination call number. ........ Merged revisions 355182 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 355183 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-14 09:58 +0000 [r355138] Alexandr Anikin + + * addons/chan_ooh323.c, /: call manager_event only if there is not + null channel structure (Closes issue ASTERISK-19298) Reported by: + robinfood Patches: issue19298.patch uploaded by may213 (License + #5415) ........ Merged revisions 355136 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355137 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-14 00:43 +0000 [r355102] Russell Bryant + + * res/res_agi.c, CHANGES: res_agi: Add AGIEXITONHANGUP variable. + This patch adds a variable AGIEXITONHANGUP for res_agi. If this + variable is set to "yes" on a channel, AGI() will exit + immediately once a channel hangup has been detected. This was the + behavior of AGI() in Asterisk 1.4 and earlier and is still + desired by some people. Review: + https://reviewboard.asterisk.org/r/1734/ + +2012-02-13 22:04 +0000 [r355055-355058] Richard Mudgett + + * pbx/pbx_spool.c, /: Fix occasional incorrectly delayed call-file + execution. Since the dir timestamp is available at one second + resolution, we cannot know if it was updated within the same + second after we scanned it. Therefore, we will force another scan + if the dir was just modified. * Changed to force another scan if + the directory was just modified. (closes issue ASTERISK-19081) + Reported by: Knut Bakke Review: + https://reviewboard.asterisk.org/r/1688/ ........ Merged + revisions 355056 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 355057 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/chan_misdn.c: Fix compile error from most recent + ast_channel opaquification installment. + +2012-02-13 19:56 +0000 [r355011] Joshua Colp + + * /, pbx/pbx_config.c: Only allow one 'dialplan reload' to execute + at a time as otherwise they would share the same common local + context list. (closes issue AST-758) ........ Merged revisions + 355009 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 355010 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-13 17:27 +0000 [r354968] Terry Wilson + + * channels/chan_local.c, addons/chan_ooh323.c, + channels/chan_iax2.c, main/cli.c, channels/chan_dahdi.c, + channels/sig_analog.c, channels/chan_skinny.c, main/features.c, + apps/app_dumpchan.c, pbx/pbx_realtime.c, channels/chan_alsa.c, + apps/app_dial.c, main/pbx.c, apps/app_fax.c, + channels/chan_agent.c, include/asterisk/channel.h, + apps/app_talkdetect.c, main/cel.c, channels/chan_misdn.c, + funcs/func_channel.c, apps/app_macro.c, apps/app_chanspy.c, + res/res_calendar.c, apps/app_voicemail.c, + channels/chan_unistim.c, tests/test_substitution.c, + channels/chan_vpb.cc, apps/app_meetme.c, main/ccss.c, + apps/app_readexten.c, channels/chan_gtalk.c, main/cdr.c, + main/channel.c, main/dial.c, channels/chan_phone.c, + main/manager.c, apps/app_osplookup.c, + bridges/bridge_builtin_features.c, res/res_agi.c, + apps/app_minivm.c, apps/app_confbridge.c, apps/app_directory.c, + addons/chan_mobile.c, apps/app_rpt.c, apps/app_parkandannounce.c, + channels/chan_mgcp.c, apps/app_while.c, funcs/func_dialplan.c, + channels/chan_sip.c, res/res_fax.c, main/channel_internal_api.c, + pbx/pbx_lua.c, channels/sig_pri.c, apps/app_queue.c, + channels/chan_oss.c, channels/chan_jingle.c, + apps/app_directed_pickup.c, main/file.c, channels/chan_h323.c, + res/snmp/agent.c, pbx/pbx_dundi.c, channels/chan_nbs.c, + apps/app_stack.c, apps/app_verbose.c: Opaquify char * and char[] + in ast_channel Review: https://reviewboard.asterisk.org/r/1733/ + +2012-02-13 17:25 +0000 [r354964] Richard Mudgett + + * res/res_config_pgsql.c, /, configs/extconfig.conf.sample: Fix + reconnecting to pgsql database after connection loss. There can + only be one database connection in res_config_pgsql just like + res_config_sqlite. If the connection is lost, the connection may + not get reestablished to the same database if the res_pgsql.conf + and extconfig.conf files are inconsistent. * Made only use the + configured database from res_pgsql.conf. * Fixed potential buffer + overwrite of last[] in config_pgsql(). (closes issue + ASTERISK-16982) Reported by: german aracil boned Review: + https://reviewboard.asterisk.org/r/1731/ ........ Merged + revisions 354953 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354959 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-13 16:42 +0000 [r354939] Joshua Colp + + * /, apps/app_confbridge.c: Don't try to play sound files that do + not exist. (closes issue ASTERISK-19188) Reported by: slesru + ........ Merged revisions 354938 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-10 22:44 +0000 [r354903] Jason Parker + + * /, apps/app_voicemail.c: Fix a voicemail memory leak with + heard/deleted messages. open_mailbox() was changed quite a long + time ago to allocate this memory. close_mailbox() should have + been changed to be responsible for freeing it. ........ Merged + revisions 354889 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354890 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-10 18:08 +0000 [r354837] Richard Mudgett + + * main/manager.c, /: Fix AMI Redirect ExtraChannel not redirecting + to the same exten and context. The astman_get_header() never + returns NULL so the check by the code for NULL would never fail. + (closes issue ASTERISK-16974) Reported by: Nuno Borges Patches: + 0018325.patch (license #6116) patch uploaded by Nuno Borges + (modified) ........ Merged revisions 354835 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354836 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-10 14:51 +0000 [r354799] Matthew Jordan + + * apps/app_voicemail.c: Fix IMAP app_voicemail compilation issue + introduced in r354429 This simply fixes the compilation issue + introduced in r354429 by re-adding the 'quote' variable. (closes + issue ASTERISK-19337) Reported by: John Taylor + +2012-02-09 22:06 +0000 [r354751] Terry Wilson + + * /, funcs/func_cdr.c: Note that CDRs are immutable once a bridge + is torn down CDRs cannot be modified after a bridge is torn down, + (e.g. after Dial() returns) even though the CDR() function may be + called. Since modifying the CDR code to change this behavior + could very easily break all kinds of things, this patch just + documents this limitation. (closes issues ASTERISK-16923) Review: + https://reviewboard.asterisk.org/r/1720/ ........ Merged + revisions 354749 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354750 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-09 20:52 +0000 [r354657-354704] Kinsey Moore + + * /, channels/chan_sip.c: Fix parsing of SIP headers where compact + and non-compact headers are mixed Change parsing of SIP headers + so that compactness of the header no longer influences which + header will be chosen. Previously, a non-compact header would be + chosen instead of a preceeding compact-form header. (closes issue + ASTERISK-17192) Review: https://reviewboard.asterisk.org/r/1728/ + ........ Merged revisions 354702 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354703 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/config.c: Make the config parser remove escaping + backslashes The config parser in Asterisk does not currently + remove a backslash that is used to escape a semicolon which would + otherwise be interpreted as the start of a comment. The change + here causes that backslash to be removed, but does not create a + real escape system in the config parser. The biggest complication + with a real escape system would be breaking existing configs + everywhere (parsing \\ as \ and breaking on escaped non-semicolon + characters) even though it would be the "right" way to do things. + (closes issue ASTERISK-17121) Review: + https://reviewboard.asterisk.org/r/1724/ ........ Merged + revisions 354655 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354656 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-09 18:14 +0000 [r354597] Terry Wilson + + * channels/chan_sip.c, channels/sip/include/config_parser.h, + channels/sip/utils.c (added), configs/sip.conf.sample, CHANGES, + channels/sip/config_parser.c, channels/sip/include/sip.h, + channels/sip/include/sip_utils.h: Add auto_force_rport and + auto_comedia NAT options This patch adds the auto_force_rport and + auto_comedia NAT options. It also converts the nat= setting to a + list of comma-separated combinable options: no, force_rport, + comedia, auto_force_rport, and auto_comedia. nat=yes remains as + an undocumented option equal to "force_rport,comedia". The first + instance of 'yes' or 'no' in the list stops parsing and overrides + any previously set options. If an auto_* option is specified with + its non-auto_ counterpart, the auto setting takes precedence. + This patch builds upon the patch posted to ASTERISK-17860 by JIRA + user pedro-garcia. (closes issue ASTERISK-17860) Review: + https://reviewboard.asterisk.org/r/1698/ + +2012-02-09 17:17 +0000 [r354552] Mark Michelson + + * /, res/res_fax.c: Adding reload support to res_fax.so (closes + issue ASTERISK-16712) reported by Frank DiGennaro Review: + https://reviewboard.asterisk.org/r/1713 ........ Merged revisions + 354545 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 354546 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-09 17:09 +0000 [r354544-354549] Matthew Jordan + + * /, channels/chan_sip.c: Clean-up of minor formatting issues in + r354542/3/4 rmudgett pointed out some formatting issues in the + check-in for ASTERISK-19290. This cleans those up. Review: + https://reviewboards.asterisk.org/r/1722/ ........ Merged + revisions 354547 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354548 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Fix SIP INFO DTMF handling for + non-numeric codes In ASTERISK-18924, SIP INFO DTMF handlingw as + changed to account for both lowercase alphatbetic DTMF events, as + well as uppercase alphabetic DTMF events. When this occurred, the + comparison of the character buffer containing the event code was + changed such that the buffer was first compared again '0' and '9' + to determine if it was numeric. Unfortunately, since the first + character in the buffer will typically be '1' in the case of + non-numeric event codes (10-16), this caused those codes to be + converted to a DTMF event of '1'. This patch fixes that, and + cleans up handling of both application/dtmf-relay and + application/dtmf content types. Review: + https://reviewboard.asterisk.org/r/1722/ (closes issue + ASTERISK-19290) Reported by: Ira Emus Tested by: mjordan ........ + Merged revisions 354542 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354543 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-09 03:09 +0000 [r354497-354498] Richard Mudgett + + * channels/chan_dahdi.c, channels/chan_misdn.c: Fix some compile + problems from the 'cppcheck' patch. + + * /, apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce. + Well, thats embarrasing. I forgot to initialize the caller_id + storage. (closes issue ASTERISK-19311) Reported by: tootai Tested + by: rmudgett ........ Merged revisions 354495 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354496 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-09 02:28 +0000 [r354494] Russell Bryant + + * main/channel.c, /: Remove some unnecessary locking from + ast_hangup(). This patch removes some unnecessary locking of the + channels container in ast_hangup(). The reason this came up is + that this lock can very quickly block the entire system. If any + of the channel cleanup code decides to block, it causes a problem + for the whole system. For example, when audiohooks get destroyed, + if that blocks for a while waiting on the mixmonitor thread to + exit because it's busy blocking on some I/O, it causes a problem + for many other threads in the meantime. Review: + https://reviewboard.asterisk.org/r/1712/ ........ Merged + revisions 354492 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354493 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-08 21:29 +0000 [r354459] Kevin P. Fleming + + * res/res_ais.c (removed), contrib/scripts/install_prereq: Revision + 354046 added res_corosync as a replacement for res_ais, but + didn't actually remove res_ais. This commit removes it. In + addition, the 'install_prereq' script has been updated to no + longer install AIS dependency packages, and instead install + Corosync packages instead. + +2012-02-08 21:28 +0000 [r354458] Terry Wilson + + * channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql, + CHANGES, channels/sip/include/sip.h: Add callbackextension + matching & realtime callbackextensions This patch is based on the + one by David Vossel, developer extrodinaire, at + https://reviewboard.asterisk.org/r/344/. If multiple peers are + defined with the same host/port, but differing + callbackextensions, it chooses the peer with the matching + callbackextension. Since callbackextension creates an outbound + registration with the callbackextension as the Contact address, + matching an incoming request by that (in addition to the + host/port) makes a lot of sense. This patch also adds support for + callbackextension to realtime by querying all peers with + callbackextensions on reload and adding registrations for them. + (closes issue ASTERISK-13456) Review: + https://reviewboard.asterisk.org/r/344/ Review: + https://reviewboard.asterisk.org/r/1717/ + +2012-02-08 21:25 +0000 [r354450] Kevin P. Fleming + + * channels/chan_dahdi.c: Restore some variables removed by the + 'cppcheck' patch that were actually needed. + +2012-02-08 20:49 +0000 [r354429] Walter Doekes + + * apps/app_dial.c, main/udptl.c, main/pbx.c, addons/chan_ooh323.c, + funcs/func_env.c, funcs/func_strings.c, utils/astman.c, + main/acl.c, apps/app_disa.c, apps/app_alarmreceiver.c, + apps/app_queue.c, channels/chan_iax2.c, + addons/ooh323c/src/memheap.c, channels/chan_usbradio.c, + channels/chan_dahdi.c, apps/app_osplookup.c, + channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_odbc.c, + main/ast_expr2f.c, apps/app_minivm.c, formats/format_h263.c, + addons/chan_mobile.c, apps/app_chanspy.c, main/ast_expr2.fl, + apps/app_voicemail.c: Avoid cppcheck warnings; removing unused + vars and a bit of cleanup. Patch by: Clod Patry Review: + https://reviewboard.asterisk.org/r/1651 + +2012-02-08 15:28 +0000 [r354395] Kinsey Moore + + * CHANGES: Add CHANGES documentation for the "pri set debug" + bitmask change (related to ASTERISK-17159) + +2012-02-07 21:33 +0000 [r354360] Terry Wilson + + * /, channels/chan_sip.c, contrib/realtime/postgresql/realtime.sql: + Fix multiple SIP realtime issues 1. Set lastms to 0 when clearing + instead of "" 2. Don't set ipaddr or port to the string "(null)" + when they are empty 3. Add missing required fields, set default + for lastms to 0, and modify the length of the ipaddr field to 45 + in the Postgresql realtime.sql file. (closes issue + ASTERISK-19172) Review: https://reviewboard.asterisk.org/r/1703/ + ........ Merged revisions 354348 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354349 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-07 18:07 +0000 [r354312-354314] Sean Bright + + * contrib/scripts/live_ast: Continuation of last patch - since + LIVE_AST_LD_PATH_EXTRA will now never be empty, don't check for + it, instead of check if LD_LIBRARY_PATH is already set and if so, + append LIVE_AST_LD_PATH_EXTRA properly. + + * contrib/scripts/live_ast: Include live/usr/lib in the shared + library search path to that we pick up libasteriskssl.so at run + time when using live_ast. + + * contrib/scripts/live_ast: Whitespace only (remove trailing + spaces) + +2012-02-07 15:29 +0000 [r354275] Jonathan Rose + + * /, cdr/cdr_pgsql.c: Fix column duplication bug in module reload + for cdr_pgsql. Prior to this patch, attempts to reload + cdr_pgsql.so would cause the column list to keep its current data + and then add a second copy during the reload. This would cause + attempts to log the CDR to the database to fail. This patch also + cleans up some unnecessary null checks for ast_free and deals + with a few potential locking problems. (closes issue + ASTERISK-19216) Reported by: Jacek Konieczny Review: + https://reviewboard.asterisk.org/r/1711/ ........ Merged + revisions 354263 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354270 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-06 23:15 +0000 [r354174-354218] Richard Mudgett + + * /, pbx/pbx_config.c: Improved documentation of CLI "dialplan add + extension" command. * Documented dialplan add extension + ,,)> format. * Allow acceptance + of command without the app-data value. There are many + applications that do no need any parameters so it is silly to + require that field for all commands. * Fixed a couple + ast_malloc/ast_free mismatches with ast_add_extension2() calls. + (closes issue ASTERISK-19222) Reported by: Andrey Solovyev Tested + by: rmudgett ........ Merged revisions 354216 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354217 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/sig_pri.h: Restore alternate SIG_PRI_DEBUG_DEFAULT + meaning. + +2012-02-06 20:18 +0000 [r354165] Kinsey Moore + + * channels/sig_pri.h, channels/chan_dahdi.c: Allow more control + over the output of pri debug This changes the debuglevel of 'pri + set debug' to a bit mask allowing the user to independently + select bits of output: 1 libpri internals including state machine + 2 Decoded Q.931 messages 4 Decoded Q.921 headers 8 raw hex dump + of the full frames Additionally, this ensures that the meaning of + "on" does not change and intrudces intense and hex to simplify + usage. (closes issue ASTERISK-17159) Original-patch-by: wimpy + +2012-02-06 17:33 +0000 [r354120] Richard Mudgett + + * /, main/features.c: Add missing headers to AMI UnParkedCall event + to uniquely identify the call. The AMI UnParkedCall event was + missing the Parkinglot and Uniqueid headers that the AMI + ParkedCall event contains. (closes issue ASTERISK-19240) Reported + by: Michael Yara ........ Merged revisions 354116 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354119 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-06 16:38 +0000 [r354084] Joshua Colp + + * apps/app_meetme.c, UPGRADE.txt: Make the 'c' option to MeetMe + work even if the 'q' option is used. (closes issue + ASTERISK-17053) Reported by: justdave + +2012-02-05 10:58 +0000 [r354046] Russell Bryant + + * build_tools/menuselect-deps.in, configure, + include/asterisk/autoconfig.h.in, res/res_corosync.c (added), + configure.ac, configs/res_corosync.conf.sample (added), res/ais + (removed), UPGRADE.txt, configs/ais.conf.sample (removed), + CHANGES, makeopts.in: Replace res_ais with a new module, + res_corosync. This patch removes res_ais and introduces a new + module, res_corosync. The OpenAIS project is deprecated and is + now just a wrapper around Corosync. This module provides the same + functionality using the same core infrastructure, but without the + use of the deprecated components. Technically res_ais could have + been used with an AIS implementation other than OpenAIS, but that + is the only one I know of that was ever used. Review: + https://reviewboard.asterisk.org/r/1700/ + +2012-02-03 21:33 +0000 [r354001] Jonathan Rose + + * /, channels/chan_agent.c: Fixes deadlocks occuring in chan_agent + due to r335976 Bad locking order was added to chan_agent to + prevent segfaults from having no locking in a patch by irroot. + This patch addresses the bad locking order by releasing locks + before getting the right locking order to stop deadlocks from + occuring when doing multiple interactions with agents. (closes + issue ASTERISK-19285) Reported by: Alex Villacis Lasso Review: + https://reviewboard.asterisk.org/r/1708/ ........ Merged + revisions 353999 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 354000 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-03 16:50 +0000 [r353964] Kinsey Moore + + * UPGRADE.txt, cdr/cdr_adaptive_odbc.c, + configs/cdr_adaptive_odbc.conf.sample: Support schema selection + in cdr_adaptive_odbc Asterisk now supports using ODBC with + databases where a single schema must be selected. Previously, + INSERTs would fail because they did not take into account extra + fields cause by having multiple schemas. This also corrects some + SQL resource leaks. (closes issue ASTERISK-17106) Patch-by: + Alexander Frolkin Patch-by: Tilgnman Lesher + +2012-02-03 16:23 +0000 [r353963] Jonathan Rose + + * /, res/res_fax.c: Fixes a segfault occuring when performing + attended transfer with FAXOPT(gateway)=yes (closes issue + ASTERISK-19184) Reported by: Alexandr ........ Merged revisions + 353962 from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-02 22:28 +0000 [r353917] Kinsey Moore + + * /, channels/chan_sip.c: Ensure entering T.38 passthrough does not + cause an infinite loop After R340970 Asterisk was still polling + the RTCP file descriptor after RTCP is shut down and removed. If + the descriptor happened to have data ready when the removal + occured then Asterisk would go into an infinite loop trying to + read data that it can never actually access. This change disables + the audio RTCP file descriptor for the duration of the T.38 + transaction. (closes issue ASTERISK-18951) Reported-by: Kristijan + Vrban ........ Merged revisions 353915 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 353916 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-02 20:18 +0000 [r353872] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: + Restore the 'w' modifier support for ISDN spans. + Dial(DAHDI/g0/1234w888) This feature also causes the sending + complete ie to be sent for switch types that do not automatically + send the ie. (EuroISDN/ETSI) The main difference between dialing + Dial(DAHDI/g0/1234w888) and Dial(DAHDI/g0/1234,,D(888)) is the + sending of the sending complete ie. (closes issue ASTERISK-19176) + Reported by: rmudgett Tested by: rmudgett ........ Merged + revisions 353867 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 353868 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-02 18:55 +0000 [r353821] Mark Michelson + + * main/manager.c, /, main/http.c, configs/manager.conf.sample, + include/asterisk/manager.h, configs/http.conf.sample: Fix TLS + port binding behavior as well as reload behavior: * Removes + references to tlsbindport from http.conf.sample and + manager.conf.sample * Properly bind to port specified in + tlsbindaddr, using the default port if specified. * On a reload, + properly close socket if the service has been disabled. A note + has been added to UPGRADE.txt to indicate how ports must be set + for TLS. (closes issue ASTERISK-16959) reported by Olaf + Holthausen (closes issue ASTERISK-19201) reported by Chris + Mylonas (closes issue ASTERISK-19204) reported by Chris Mylonas + Review: https://reviewboard.asterisk.org/r/1709 ........ Merged + revisions 353770 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 353820 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-02 17:07 +0000 [r353725-353772] Jonathan Rose + + * /, channels/chan_sip.c: Fix sip show peers port output, align + columns, and fix ami port output. A previous patch I committed + from ASTERISK-16930 unexpectedly changed some output for the AMI + action "sippeers" which this patch changes back. Also, this + aligns the output for the cli command "sip show peers" and fixes + another issue that patch introduced by using + ast_sockaddr_stringify calls multiple times without immediately + using the pointer. I also went ahead and did a little janitorial + work to clean up whitespace in _sip_show_peers. (issue + ASTERISK-16930) (closes issue ASTERISK-19281) Reported by: + Patrick El Youssef Patches: ASTERISK-19281.diff uploaded by + Walter Doekes (license 5674) ........ Merged revisions 353769 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 353771 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Use ast_sockaddr_stringify_fmt wrappers + for various functions in chan_sip There are a number of cleaner + looking wrappers for ast_sockaddr_stringify_fmt available which + are slightly more readable than using a direct call to + ast_sockaddr_stringify_fmt. This patch switches a number of those + calls in chan_sip to use those wrappers and is generally + harmless. (Closes issue ASTERISK-16930) Reported by: Michael L. + Young Patches: chan_sip-broken-registration-1.8.diff uploaded by + Michael L. Young (license 5026) ........ Merged revisions 353720 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 353721 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-01 19:53 +0000 [r353647-353685] Richard Mudgett + + * channels/chan_unistim.c, channels/chan_multicast_rtp.c, + channels/chan_local.c, addons/chan_ooh323.c, + channels/chan_vpb.cc, channels/chan_gtalk.c, + channels/chan_iax2.c, main/channel.c, channels/chan_phone.c, + channels/chan_dahdi.c, channels/sig_analog.c, main/manager.c, + pbx/pbx_spool.c, channels/chan_skinny.c, main/features.c, + channels/sig_analog.h, channels/chan_alsa.c, + apps/app_confbridge.c, addons/chan_mobile.c, channels/sig_ss7.c, + channels/chan_mgcp.c, main/pbx.c, channels/sig_ss7.h, + channels/chan_sip.c, channels/chan_bridge.c, + channels/chan_agent.c, include/asterisk/channel.h, + channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c, + channels/chan_usbradio.c, channels/chan_jingle.c, + channels/sig_pri.h, channels/chan_misdn.c, channels/chan_h323.c, + channels/chan_nbs.c, include/asterisk/pbx.h: Constify some more + channel driver technology callback parameters. Review: + https://reviewboard.asterisk.org/r/1707/ + + * cel/cel_pgsql.c, configs/cel_sqlite3_custom.conf.sample, + cel/cel_odbc.c, configs/cel.conf.sample, cel/cel_manager.c, + cel/cel_tds.c, configs/cel_pgsql.conf.sample, + configs/cel_odbc.conf.sample, main/cel.c, + configs/cel_custom.conf.sample: Remove inconsistency in CEL + eventtype for user defined events. The CEL eventtype field for + ODBC and PGSQL backends should be USER_DEFINED instead of the + user defined event name supplied by the CELGenUserEvent + application. If the field is output as a number, the user defined + name does not have a value and is always output as 21 for + USER_DEFINED and the userdeftype field would be required to + supply the user defined name. The following CEL backends + (cel_odbc, cel_pgsql, cel_custom, cel_manager, and + cel_sqlite3_custom) can be independently configured to remove + this inconsistency. * Allows cel_manager, cel_custom, and + cel_sqlite3_custom to behave the same way. (closes issue + ASTERISK-17189) Reported by: Bryant Zimmerman Review: + https://reviewboard.asterisk.org/r/1669/ + + * main/channel.c, include/asterisk/channel.h: Fix ExtenSpy and + simplify the channel search functions. When ast_channel name was + opaquified, the channel search functions did not get converted + correctly. As a result ExtenSpy which uses a channel iterator + search by exten@context could never find anything. * Updated the + doxygen documentation for the search functions in channel.h. + Review: https://reviewboard.asterisk.org/r/1702/ + +2012-02-01 15:59 +0000 [r353600] Sean Bright + + * /, include/asterisk/audiohook.h: Resolve an overlap in the + ast_audiohook_flags values. AST_AUDIOHOOK_TRIGGER_WRITE and + AST_AUDIOHOOK_WANTS_DTMF were overlapping which may have caused + unintended side effects. This patch moves + AST_AUDIOHOOK_TRIGGER_WRITE, and updates + AST_AUDIOHOOK_TRIGGER_MODE to reflect the original intention. + This will affect existing modules that use these flags, so be + sure to recompile as necessary. (closes issue ASTERISK-19246) + Reported by: feyfre ........ Merged revisions 353598 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 353599 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-01 15:07 +0000 [r353552] Matthew Jordan + + * /, contrib/init.d/etc_default_asterisk: Added clarification for + the VERBOSITY setting to etc_default_asterisk Clarified that + using the VERBOSITY setting in etc_default_asterisk is the same + as using the -v command line switch, which causes Asterisk to + launch in console mode. (closes issue ASTERISK-17030) Reported + by: Jonas ........ Merged revisions 353550 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 353551 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-02-01 00:08 +0000 [r353504] Terry Wilson + + * /, res/res_calendar.c: Allow res_calendar to be unloaded The + calendaring tech modules depend on res_calendar and initially + res_calendar just bumped the use count so that it couldn't be + unloaded. res_calendar can potentially create many threads and + I've seen issues where the Asterisk shutdown has failed where it + looked like these threads could be the culprit. This patch adds + unload support for res_calendar. Unloading res_calendar will also + unload the dependant tech modules as well. (closes issue + ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/ + ........ Merged revisions 353502 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 353503 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-31 17:26 +0000 [r353466] Richard Mudgett + + * main/manager.c, /, include/asterisk/channel.h: Fix memory leak in + error paths for action_originate(). * Fix memory leak of vars in + error paths for action_originate(). * Moved struct + fast_originate_helper tech and data members to stringfields. * + Simplified ActionID header handling for fast_originate(). * Added + doxygen note to ast_request() and ast_call() and the associated + channel callbacks that the data/addr parameters should be treated + as const char *. Review: https://reviewboard.asterisk.org/r/1690/ + ........ Merged revisions 353454 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 353463 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-30 23:58 +0000 [r353418] Terry Wilson + + * main/dnsmgr.c, /, channels/chan_sip.c, include/asterisk/dnsmgr.h: + Re-link peers by IP when dnsmgr changes the IP Asterisk's dnsmgr + currently takes a pointer to an ast_sockaddr and updates it + anytime an address resolves to something different. There are a + couple of issues with this. First, the ast_sockaddr is usually + the address of an ast_sockaddr inside a refcounted struct and we + never bump the refcount of those structs when using dnsmgr. This + makes it possible that a refresh could happen after the + destructor for that object is called (despite ast_dnsmgr_release + being called in that destructor). Second, the module using dnsmgr + cannot be aware of an address changing without polling for it in + the code. If an action needs to be taken on address update (like + re-linking a SIP peer in the peers_by_ip table), then polling for + this change negates many of the benefits of having dnsmgr in the + first place. This patch adds a function to the dnsmgr API that + calls an update callback instead of blindly updating the address + itself. It also moves calls to ast_dnsmgr_release outside of the + destructor functions and into cleanup functions that are called + when we no longer need the objects and increments the refcount of + the objects using dnsmgr since those objects are stored on the + ast_dnsmgr_entry struct. A helper function for returning the + proper default SIP port (non-tls vs tls) is also added and used. + This patch also incorporates changes from a patch posted by Timo + Teräs to ASTERISK-19106 for related dnsmgr issues. (closes issue + ASTERISK-19106) Review: https://reviewboard.asterisk.org/r/1691/ + ........ Merged revisions 353371 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 353397 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-30 22:44 +0000 [r353347-353370] Alec L Davis + + * /, channels/chan_sip.c: Merged revisions 353369 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r353369 | alecdavis | 2012-01-31 11:42:28 +1300 + (Tue, 31 Jan 2012) | 9 lines Merged revisions 353368 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r353368 | alecdavis | 2012-01-31 11:40:40 +1300 (Tue, 31 + Jan 2012) | 2 lines prevent debug messsges displaying -ve Cseq + numbers. Missed in R353320 ........ ................ + + * channels/sip/include/dialog.h, /, channels/chan_sip.c, + channels/sip/include/sip.h: Merged revisions 353321 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r353321 | alecdavis | 2012-01-31 11:16:22 +1300 + (Tue, 31 Jan 2012) | 25 lines Merged revisions 353320 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r353320 | alecdavis | 2012-01-31 10:57:49 +1300 (Tue, 31 Jan + 2012) | 18 lines RFC3261 Section 8.1.1.5. The sequence number + value MUST be expressible as a 32-bit unsigned integer * fix: use + %u instead of %d when dealing with CSeq numbers - to remove + possibility of -ve numbers. * fix: change all uses of seqno and + friends (ocseq icseq) from 'int' or 'unsigned int' to uint32_t. + Summary of CSeq numbers. An initial CSeq number must be less than + 2^31 A CSeq number can increase in value up to 2^32-1 An + incrementing CSeq number must not wrap around to 0. Tested with + Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) + Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/1699/ ........ + ................ + +2012-01-30 21:34 +0000 [r353262-353319] Kevin P. Fleming + + * Makefile: Correct serious flaw in the top-level Makefile. + + * include/asterisk.h, /, main/Makefile, main/libasteriskssl.c + (added), configure.ac, Makefile.moddir_rules, main/ssl.c + (removed), addons, CHANGES, include/asterisk/optional_api.h, + Makefile, build_tools/mkpkgconfig, configure, main, makeopts.in, + build_tools/make_defaults_h, main/libasteriskssl.exports.in + (added): Address OpenSSL initialization issues when using + third-party libraries. When Asterisk is used with various + third-party libraries (CURL, PostgresSQL, many others) that have + the ability themselves to use OpenSSL, it is possible for + conflicts to arise in how the OpenSSL libraries are initialized + and shutdown. This patch addresses these conflicts by 'wrapping' + the important functions from the OpenSSL libraries in a new + shared library that is part of Asterisk itself, and is loaded in + such a way as to ensure that *all* calls to these functions will + be dispatched through the Asterisk wrapper functions, not the + native functions. This new library is optional, but enabled by + default. See the CHANGES file for documentation on how to disable + it. Along the way, this patch also makes a few other minor + changes: * Changes MODULES_DIR to ASTMODDIR throughout the build + system, in order to more closely match what is used during + run-time configuration. * Corrects some errors in the configure + script where AC_CHECK_TOOLS was used instead of AC_PATH_PROG. * + Adds a new variable for linker flags in the build system + (DYLINK), used for producing true shared libraries (as opposed to + the dynamically loadable modules that the build system produces + for 'regular' Asterisk modules). * Moves the Makefile bits that + handle installation and uninstallation of the main Asterisk + binary into main/Makefile from the top-level Makefile. * Moves a + couple of useful preprocessor macros from optional_api.h to + asterisk.h. Review: https://reviewboard.asterisk.org/r/1006/ + + * /, channels/chan_sip.c: Clarify log WARNING message when + port-zero SDP 'm' lines received. Previously, if an m-line in an + SDP offer or answer had a port number of zero, that line was + skipped, and resulted in an 'Unsupported SDP media type...' + warning message. This was misleading, as the media type was not + unsupported, but was ignored because the m-line indicated that + the media stream had been rejected (in an answer) or was not + going to be used (in an offer). ........ Merged revisions 353260 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 353261 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-29 22:33 +0000 [r353224] Damien Wedhorn + + * channels/chan_skinny.c: Allow softkey reject while device onhook. + Fixes up softkey endcall. Previous code was a copy of onhook, now + allows for endcall softkey to be used while device is still + onhook. + +2012-01-29 02:45 +0000 [r353177] Russell Bryant + + * /, main/netsock.c: Find even more network interfaces. The + previous change made the code look for emN and pciN in addition + to what it did originally, which was search for ethN. However, it + needed to be looking for pciN#N, so that's what it does now. This + also moves the memset() to be before every ioctl(). ........ + Merged revisions 353175 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 353176 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-28 14:52 +0000 [r353128] Kevin P. Fleming + + * main/rtp_engine.c, /: Add 'L16-256' MIME subtype alias for + slin16. Asterisk has supported the 'L16' MIME subtype for 16kHz + signed linear (PCM) audio for quite some time, but some endpoints + refer to it as 'L16-256'. This commit adds this as an alias for + the existing format. ........ Merged revisions 353126 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 353127 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-28 04:31 +0000 [r353079] Russell Bryant + + * /, main/netsock.c: Update ast_set_default_eid() to find more + network interfaces. As of Fedora 15, ethN is not the name of + ethernet interfaces. The names are emN or pciN. Update some code + that searched for interfaces named ethN to look for the new + names, as well. For more information about why this change was + made, see this page: http://domsch.com/blog/?p=455 ........ + Merged revisions 353077 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 353078 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-27 21:38 +0000 [r352996-353040] Richard Mudgett + + * /, apps/app_queue.c: Audit of ao2_iterator_init() usage for v10. + Missed one. ........ Merged revisions 353039 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, tests/test_format_api.c: Audit of ao2_iterator_init() usage + for v10. Fix double format_cap iterator cleanup. ........ Merged + revisions 352992 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-27 19:26 +0000 [r352981] Jonathan Rose + + * /, res/res_monitor.c: Make failed PauseMonitor and UnpauseMonitor + with no valid channel not close AMI session. I also went ahead + and took a little time to make sure that the manager value + AMI_SUCCESS was used instead of just return 0 being thrown around + everywhere since that's how we handle this stuff these days. + (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches: + res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey + (license 5766) ........ Merged revisions 352959 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352965 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-27 18:47 +0000 [r352957] Richard Mudgett + + * main/pbx.c, /, channels/chan_sip.c, + include/asterisk/indications.h, res/snmp/agent.c, + main/taskprocessor.c, apps/app_queue.c, channels/chan_iax2.c, + apps/app_chanspy.c, main/indications.c, res/res_odbc.c, + res/res_srtp.c: Audit of ao2_iterator_init() usage for v1.8. + Fixes numerous reference leaks and missing ao2_iterator_destroy() + calls as a result. Review: + https://reviewboard.asterisk.org/r/1697/ ........ Merged + revisions 352955 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352956 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-27 15:57 +0000 [r352916] Terry Wilson + + * res/res_calendar_exchange.c, res/res_calendar_caldav.c, + res/res_calendar.c: Add aresult variable for CALENDAR_WRITE This + patch adds a CALENDAR_SUCCESS=1/0 variable that is set to show + whether or not CALENDAR_WRITE has passed. This patch also adds + some debugging for caldav PUT responses and no longer treats + responses with no body as an error (as a PUT gets a 201 Created + with no body). (closes issue ASTERISK-16903) Reported by: Clod + Patry Tested by: Terry Wilson Patches: calendarstatus.diff + uploaded by Clod Patry (License #5138), slightly modified by + Terry Wilson Review: https://reviewboard.asterisk.org/r/1692/ - + This line, and those below, will be ignored-- M + res/res_calendar.c M res/res_calendar_exchange.c M + res/res_calendar_caldav.c + +2012-01-27 00:11 +0000 [r352864] Alec L Davis + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged + revisions 352863 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r352863 | alecdavis | 2012-01-27 13:08:03 +1300 + (Fri, 27 Jan 2012) | 19 lines Merged revisions 352862 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r352862 | alecdavis | 2012-01-27 13:05:30 +1300 (Fri, 27 Jan + 2012) | 12 lines rfc4235 - Section 4.1: Versions MUST be + representable using a non-negative 32 bit integer. If a BLF + subscription exists for long enough, using %d may print negative + version numbers. Unlikely, as 2^32 at 1 update per second is ~137 + years, or half that before the versions number started going + negative. Tested with Asterisk 1.8.8.2 with Grandstream phones. + alecdavis (license 585) Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/1694/ ........ + ................ + +2012-01-26 20:44 +0000 [r352821] Alexandr Anikin + + * addons/chan_ooh323.c, /: Fix outbound DTMF for inband mode (tell + asterisk core to generate DTMF sounds). (Closes issue + ASTERISK-19233) Reported by: Matt Behrens Patches: + chan_ooh323.c.patch uploaded by Matt Behrens (License #6346) + ........ Merged revisions 352807 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352817 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-26 19:09 +0000 [r352757] Jonathan Rose + + * /, channels/chan_sip.c: Copy amaflags to sip_pvt from peer during + create_addr_from_peer For whatever reason, we don't have a single + function for copying data like this from SIP peers to the SIP + pvt. This patch adds the copying of amaflags to the sip_pvt, but + it would probably be worth discussing this function along with + the others that essentially just copy some amount of data from a + peer to a private. (Closes issue ASTERISK-19029) Reported by: + Matt Lehner ........ Merged revisions 352755 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352756 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-26 06:36 +0000 [r352706] Alec L Davis + + * /, channels/chan_sip.c: Merged revisions 352705 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r352705 | alecdavis | 2012-01-26 19:33:11 +1300 + (Thu, 26 Jan 2012) | 27 lines Merged revisions 352704 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r352704 | alecdavis | 2012-01-26 19:27:07 +1300 (Thu, 26 Jan + 2012) | 20 lines Cleanup dialog-info+xml Notify dialog Make + similar to other Notify messages. sample output: + terminated Tested with + Asterisk 1.8.8.2 with Grandstream phones. alecdavis (license 585) + Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/1693/ ........ + ................ + +2012-01-25 22:25 +0000 [r352659] Paul Belanger + + * /, apps/app_voicemail.c: Fix -Werror=unused-but-set-variable + compiler error (gcc 4.6.2) ........ Merged revisions 352643 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352651 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-25 21:31 +0000 [r352626] Kevin P. Fleming + + * Makefile, include/asterisk/version.h (added), main/test.c, + build_tools/make_version_h (removed), include/asterisk: Remove + "asterisk/version.h" in favor of "asterisk/ast_version.h". A long + time ago, in a land far far away, we added + "asterisk/ast_version.h", which provides the ast_get_version() + and ast_get_version_num() functions. These were added so that + modules that needed the version information for the Asterisk + instance they were loaded in could actually get it (as opposed + the version that they were compiled against). We changed + everything in the tree to use the new mechanism (although later + main/test.c was added using the old method). However, the old + mechanism was never removed, and as a result, new code is still + trying to use it. This commit removes asterisk/version.h and + replaces it with a header that will generate a compile-time error + if you try to use it (the error message tells you which header + you should use instead). It also removes the Makefile and + build_tools bits that generated the file, and it updates + main/test.c to use the 'proper' method of getting the Asterisk + version information. This is an API change and thus is being + committed for trunk only, but it's a fairly minor one and + definitely improves the situation for out-of-tree modules. + +2012-01-25 17:33 +0000 [r352565] Terry Wilson + + * /, channels/chan_sip.c: Remove some extraneous debugging from + registry memleak fix ........ Merged revisions 352551 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352556 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-25 17:23 +0000 [r352538] Richard Mudgett + + * /, channels/chan_sip.c, CHANGES, main/message.c, + channels/sip/include/sip.h: Fixes for sending SIP MESSAGE outside + of calls. * Fix authenticate MESSAGE losing custom headers added + by the MESSAGE_DATA function in the authorization attempt. * Pass + up better From header contents for SIP to use. Now is in the + "display-name" format expected by MessageSend. (Note that + this is a behavior change that could concievably affect some + people.) * Block user from adding standard headers that are added + automatically. (To, From,...) * Allow the user to override the + Content-Type header contents sent by MessageSend. * Decrement + Max-Forwards header if the user transferred it from an incoming + message. * Expand SIP short header names so the dialplan and + other code only has to deal with the full names. * Documents what + SIP expects in the MessageSend(from) parameter. (closes issue + ASTERISK-18992) Reported by: Yuri (closes issue ASTERISK-18917) + Reported by: Shaun Clark Review: + https://reviewboard.asterisk.org/r/1683/ ........ Merged + revisions 352520 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-25 17:02 +0000 [r352519] Terry Wilson + + * /, channels/chan_sip.c: Clean up some SIP registry-related memory + leaks 1) Be sure and free at unload the epa_backend we allocate + at startup 2) Do the same sip_registry cleanup at unload we do at + reload Review: https://reviewboard.asterisk.org/r/1689/ ........ + Merged revisions 352514 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352515 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-25 16:54 +0000 [r352517] Kevin P. Fleming + + * main/format.c, /, main/format_cap.c, main/format_pref.c: + Eliminate unnecessary rebuilds of main/format*.c. These files + have no need to include "asterisk/version.h", and doing so forces + them to be rebuilt each time a Subversion checkout moves between + 'modified' and 'unmodified' states. ........ Merged revisions + 352516 from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-25 16:42 +0000 [r352513] Jonathan Rose + + * /, configs/sip.conf.sample: Redocuments sip types peer, user, + friend in sip.conf.sample There was faulty information in the + sample config describing user as a synonym for friend so it has + been changed to better elaborate on the differences between the + three entity types. (closes issue ASTERISK-15537) Reported by: + yarique ........ Merged revisions 352511 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352512 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-25 01:21 +0000 [r352475] Terry Wilson + + * channels/chan_vpb.cc: Fix channel opaquification of stringfields + for chan_vpb + +2012-01-24 22:28 +0000 [r352431] Mark Michelson + + * /, channels/chan_sip.c: Don't do a DNS lookup on an outbound + REGISTER host if there is an outbound proxy configured. (closes + issue ASTERISK-16550) reported by: Olle Johansson ........ Merged + revisions 352424 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352430 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-24 20:37 +0000 [r352377] Jonathan Rose + + * /, sounds/Makefile: Set core sounds version to 1.4.22. Now that + we have the right license for the Russian 1.4.22 sounds as well + as the sounds for the Australian English 1.4.22 sounds, we can + finally set the sounds to use 1.4.22! (closes issue + ASTERISK-18978) Reported by: Cameron Twomey Patches: + confbridge.tar.001 uploaded by Cameron Twomey confbridge.tar.002 + uploaded by Cameron Twomey ........ Merged revisions 352367 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352373 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-24 20:12 +0000 [r352348] Terry Wilson + + * channels/chan_local.c, addons/chan_ooh323.c, main/say.c, + apps/app_record.c, apps/app_sayunixtime.c, channels/chan_iax2.c, + main/cli.c, channels/chan_dahdi.c, channels/sig_analog.c, + channels/chan_skinny.c, main/features.c, apps/app_dumpchan.c, + channels/chan_alsa.c, pbx/pbx_realtime.c, apps/app_externalivr.c, + apps/app_dial.c, main/pbx.c, apps/app_page.c, + channels/chan_bridge.c, apps/app_privacy.c, + channels/chan_agent.c, apps/app_disa.c, + include/asterisk/channel.h, main/aoc.c, apps/app_talkdetect.c, + main/cel.c, res/res_monitor.c, apps/app_playback.c, + apps/app_speech_utils.c, channels/chan_misdn.c, + funcs/func_channel.c, apps/app_chanspy.c, apps/app_voicemail.c, + channels/chan_unistim.c, channels/chan_multicast_rtp.c, + apps/app_meetme.c, apps/app_dictate.c, apps/app_authenticate.c, + apps/app_readexten.c, apps/app_userevent.c, + res/res_musiconhold.c, channels/chan_gtalk.c, + apps/app_followme.c, main/cdr.c, main/channel.c, + channels/chan_phone.c, main/dial.c, main/manager.c, + apps/app_minivm.c, res/res_agi.c, main/app.c, + apps/app_confbridge.c, main/image.c, apps/app_directory.c, + addons/chan_mobile.c, apps/app_rpt.c, channels/chan_mgcp.c, + apps/app_parkandannounce.c, channels/chan_sip.c, res/res_fax.c, + main/channel_internal_api.c, channels/chan_console.c, + channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c, + funcs/func_global.c, channels/chan_jingle.c, + channels/chan_usbradio.c, channels/chan_h323.c, main/file.c, + res/snmp/agent.c, channels/chan_nbs.c, apps/app_stack.c, + addons/app_saycountpl.c: Opaquify channel stringfields Continue + channel opaque-ification by wrapping all of the stringfields. + Eventually, we will restrict what can actually set these + variables, but the purpose for now is to hide the implementation + and keep people from adding code that directly accesses the + channel structure. Semantic changes will follow afterward. + Review: https://reviewboard.asterisk.org/r/1661/ + +2012-01-24 17:04 +0000 [r352293] Richard Mudgett + + * /, funcs/func_odbc.c: Fix locking issues with channel datastores + in func_odbc.c. * Fixed a potential memory leak when an existing + datastore is manually destroyed by inline code instead of calling + ast_datastore_free(). (closes issue ASTERISK-17948) Reported by: + Archie Cobbs Review: https://reviewboard.asterisk.org/r/1687/ + ........ Merged revisions 352291 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352292 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-23 20:31 +0000 [r352229-352232] Mark Michelson + + * /, main/features.c: Fix grammar of comment. ........ Merged + revisions 352230 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352231 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/features.c: Fix blind transfers from failing if an 'h' + extension is present. This prevents the 'h' extension from being + run on the transferee channel when it is transferred via a native + transfer mechanism such as SIP REFER. (closes ASTERISK-19173) + Reported by: Ross Beer Tested by: Kristjan Vrban Patches: + ASTERISK-19173 by Mark Michelson (license 5049) Review: + https://reviewboard.asterisk.org/r/1685 ........ Merged revisions + 352199 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 352228 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-23 19:22 +0000 [r352166] Matthew Jordan + + * /, res/res_fax_spandsp.c: Correctly apply FAXOPT settings (V17, + V27, V29) before starting spandsp layer While the FAXOPT function + could be used to set the modem capabilities, the input to that + function was not being applied correctly to the spandsp layer. + This patch applies the current model capabilities before starting + the spandsp layer. (closes issue: ASTERISK-16409) Reported by: + Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson + Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license + 5081) spandsp-modems-10.diff uploaded by mnicholson (license + 5081) ........ Merged revisions 352144 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352149 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-23 18:34 +0000 [r352093-352134] Jonathan Rose + + * configs/musiconhold.conf.sample, res/res_musiconhold.c, CHANGES: + Add an announcement option to music-on-hold - plays sound when + put on hold/between songs This is a feature patch which allows an + 'announcement' option to be specified in musiconhold.conf which + should be set to the name of a sound. If a valid sound is + specified for this option, then it will be played on that music + on hold class whenever a channel bound to that class is put on + hold as well as when Asterisk is able to detect that a song has + ended before starting the next song (excludes external players). + (closes ASTERISK-18977) Reported by: Timo Teräs Patches: + asterisk-moh-announcement.diff uploaded by Timo Teräs (license + 5409) + + * CHANGES, apps/app_mixmonitor.c: Adds the ability to stop specific + mixmonitors by using unique IDs set at monitor launch. MixMonitor + receives a new option i(channel_variable) which stores the unique + id at said variable. StopMixMonitor now accepts ID as an optional + argument, which if included will make StopMixMonitor specifically + target the mixmonitor on that particular channel. CLI commands + and AMI actions have been ammended to work with the IDs as well. + In addition, monitors across a channel can now be listed be + listed via CLI command "mixmonitor list " which will + display all of the mixmonitors active on that channel along with + the files they each have open. Created by Sergio González Martín. + (closes issue ASTERISK-19096) Reported by: Sergio González Martín + Review: https://reviewboard.asterisk.org/r/1643/ Review: + https://reviewboard.asterisk.org/r/1682/ + +2012-01-23 17:36 +0000 [r352092] Richard Mudgett + + * /, channels/chan_sip.c: Fix sip_cfg.notifycid to be set with the + defined enum values. The invalid value used when notifycid was + enabled was benign. As far as the code was concerned -1 and 1 are + equivalent. (closes issue ASTERISK-19232) Reported by: Eike + Kuiper ........ Merged revisions 352090 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352091 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-21 00:23 +0000 [r352041] Richard Mudgett + + * /, funcs/func_timeout.c, main/app.c: Fix ast_app_dtget() time + unit inconsistency. Note: Noone calls ast_app_dtget() with the + timeout parameter of zero so the bad code normally will never get + executed. * Fix unnecessary floating point division in + func_timeout.c timeout_write() when all other values are + integers. (closes issue ASTERISK-16817) Reported by: Dmitry + Andrianov ........ Merged revisions 352029 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352035 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-21 00:11 +0000 [r352018-352019] Mark Michelson + + * /, channels/chan_sip.c: Remove XXX comment that is not necessary. + ........ Merged revisions 352016 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352017 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Fix RTP reference leak. If a blind + transfer were initiated using a REFER without a prior reINVITE to + place the call on hold, AND if Asterisk were sending RTCP + reports, then there was a reference for the RTP instance of the + transferer. This fixes the issue by merging two similar but + slightly conflicting sections of code into a single area. It also + adds a stop_media_flows() call in the case that the transferer's + UA never sends a BYE to us like it is supposed to. (issue + ASTERISK-19192) Review: https://reviewboard.asterisk.org/r/1681/ + ........ Merged revisions 352014 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 352015 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-20 23:05 +0000 [r351977] Richard Mudgett + + * channels/chan_sip.c: Make CLI sip show channel list the complete + route set. (closes issue ASTERISK-16877) Reported by: klaus3000 + Patches: show-complete-routeset-patch.txt (license #5054) patch + uploaded by klaus3000 (modified) + +2012-01-20 21:26 +0000 [r351939] Kinsey Moore + + * channels/chan_sip.c, UPGRADE.txt: SIP session timeout AMI event + Add an AMI event in the Call category that is issued when a call + is terminated due to either RTP stream inactivity or SIP session + timer expiration. Event description: Event: SessionTimeout + Source: source Channel: channel-name Uniqueid: channel-unique-id + `source` can be either RTPTimeout or SIPSessionTimer (closes + issue ASTERISK-16467) Patch-by: Kirill Katsnelson + +2012-01-20 20:47 +0000 [r351900-351913] Mark Michelson + + * main/features.c, UPGRADE.txt, CHANGES, + configs/features.conf.sample: Various parking improvements. * + Adds per-parking lot options comebackcontext and comebackdialtime + * Makes comebacktoorigin settable per parking lot * Sets a PARKER + channel variable when comebacktoorigin is disabled (closes issue + ASTERISK-16643) Reported by: Mitch Sharp (bluecrow76) Patches: + asterisk-1.6.2.17.2-park-features-comebackcontext-consolidated-v3.diff + by Mitch Sharp (bluecrow76) license 5231 with updates by me. + Review: https://reviewboard.asterisk.org/r/1674 Review: + https://reviewboard.asterisk.org/r/963 Reviewed by Richard + Mudgett + + * apps/app_mixmonitor.c: Prevent potential buffer overflow on AMI + MixMonitor command. Don't be alarmed. This only affected trunk, + and it would have required manager access to your system. + +2012-01-20 19:36 +0000 [r351817-351862] Kinsey Moore + + * /, codecs/ilbc/iLBC_test.c: More corrections for the ilbc code + These changes are in a file that is not compiled by default, and + so were missed on earlier checks. ........ Merged revisions + 351860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 351861 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Restore + LSF_check function calls from set/unused variable removal These + functions are not noops and modify the array that is passed in. + Thanks for the catch Richard. ........ Merged revisions 351818 + from http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, codecs/ilbc/LPCencode.c, codecs/ilbc/iLBC_decode.c: Remove + more set, but unused variables in the ilbc codec GCC 4.6.3 caught + these in dev mode as well. ........ Merged revisions 351816 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-20 16:00 +0000 [r351764] Jonathan Rose + + * /, channels/chan_sip.c: Adds setting of mwi_from field to + check_auth_result check_peer_ok (closes ASTERISK-19057) Reported + By: Yuri Patches: 348360chan_sip.diff uploaded by Yuri (license + 5242) ........ Merged revisions 351759 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351762 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-20 16:00 +0000 [r351763] Matthew Jordan + + * /, codecs/ilbc/helpfun.c: Remove unused variable 'tmp' from + helpfun in ilbc codec gcc version 4.6.2 caught an unused variable + in the ilbc codec library. This would prevent compilation with + --enable-dev-mode; variable removed. ........ Merged revisions + 351760 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 351761 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-20 13:12 +0000 [r351709] Stefan Schmidt + + * /, contrib/asterisk-ng-doxygen: enable doxygen build for files in + the channels/sip folder like reqresp_parser.c ........ Merged + revisions 351707 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351708 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-19 23:31 +0000 [r351667] Richard Mudgett + + * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Misc minor + fixes in reqresp_parser.c and chan_sip.c. * Fix corner cases in + get_calleridname() parsing and ensure that the output buffer is + nul terminated. * Make get_calleridname() truncate the name it + parses if the given buffer is too small rather than abandoning + the parse and not returning anything for the name. Adjusted + get_calleridname_test() unit test to handle the truncation + change. * Fix get_in_brackets_test() unit test to check the + results of get_in_brackets() correctly. * Fix + parse_name_andor_addr() to not return the address of a local + buffer. This function is currently not used. * Fix potential NULL + pointer dereference in sip_sendtext(). * No need to + memset(calleridname) in check_user_full() or tmp_name in + get_name_and_number() because get_calleridname() ensures that it + is nul terminated. * Reply with an accurate response if + get_msg_text() fails in receive_message(). This is academic in + v1.8 because get_msg_text() can never fail. ........ Merged + revisions 351618 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351646 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-19 22:44 +0000 [r351613] Kinsey Moore + + * res/res_rtp_asterisk.c, /: Correct output of RTCP jitter + statistics in SR and RR reports Change the RTCP RR and SR + generation code to convert Asterisk's internal jitter statistics + to be represented in RTP timestamp units based on the rate of the + codec in use instead of in seconds. (closes issue ASTERISK-14530) + ........ Merged revisions 351611 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351612 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-19 21:55 +0000 [r351561] Jonathan Rose + + * /, channels/chan_sip.c, include/asterisk/netsock2.h: Eliminates + doubling the :port part of SIP Notify Message-Account headers. + This patch prevents the domain string from getting mangled during + the initreqprep step by moving the initialization to before its + immediate use. It also documents this pitfall for the + ast_sockaddr_stringify functions. (issue ASTERISK-19057) Reported + by: Yuri Review: https://reviewboard.asterisk.org/r/1678/ + ........ Merged revisions 351559 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351560 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-19 21:13 +0000 [r351506] Joshua Colp + + * /, channels/chan_sip.c: Prevent crash when an SDP offer is + received with an encrypted video stream when support for video is + disabled and res_srtp is loaded. (closes issue ASTERISK-19202) + Reported by: Catalin Sanda ........ Merged revisions 351504 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351505 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-18 21:06 +0000 [r351452] Matthew Jordan + + * codecs/ilbc/syntFilter.c (added), /, codecs/ilbc/iCBConstruct.h + (added), codecs/ilbc/iLBC_test.c (added), + codecs/ilbc/syntFilter.h (added), codecs/ilbc/StateConstructW.c + (added), codecs/ilbc/packing.c (added), + codecs/ilbc/StateConstructW.h (added), codecs/ilbc/packing.h + (added), codecs/ilbc/getCBvec.c (added), codecs/ilbc/LPCdecode.c + (added), codecs/ilbc/enhancer.c (added), codecs/ilbc/lsf.c + (added), codecs/ilbc/iLBC_encode.c (added), + codecs/ilbc/getCBvec.h (added), codecs/ilbc/LPCdecode.h (added), + codecs/ilbc/iLBC_define.h (added), codecs/ilbc/FrameClassify.c + (added), codecs/ilbc/enhancer.h (added), codecs/ilbc/lsf.h + (added), codecs/ilbc/extract-cfile.awk (added), + codecs/ilbc/iLBC_encode.h (added), codecs/ilbc/Makefile, + codecs/ilbc/FrameClassify.h (added), codecs/ilbc/helpfun.c + (added), codecs/ilbc/LICENSE_ADDENDUM (added), + codecs/ilbc/doCPLC.c (added), codecs/ilbc/anaFilter.c (added), + codecs/ilbc/helpfun.h (added), codecs/ilbc/createCB.c (added), + codecs/ilbc/doCPLC.h (added), codecs/ilbc/anaFilter.h (added), + codecs/ilbc/constants.c (added), codecs/ilbc/iLBC_decode.c + (added), codecs/ilbc/createCB.h (added), codecs/ilbc/constants.h + (added), codecs/ilbc/iLBC_decode.h (added), + codecs/ilbc/iCBSearch.c (added), codecs/ilbc/filter.c (added), + codecs/ilbc/gainquant.c (added), codecs/ilbc/hpInput.c (added), + codecs/ilbc/hpOutput.c (added), codecs/ilbc/iCBSearch.h (added), + codecs/ilbc/rfc3951.txt (added), codecs/ilbc/filter.h (added), + codecs/ilbc/gainquant.h (added), codecs/ilbc/LPCencode.c (added), + codecs/ilbc/hpInput.h (added), codecs/ilbc/PATENTS (added), + codecs/ilbc/StateSearchW.c (added), codecs/ilbc/hpOutput.h + (added), codecs/codec_ilbc.c, contrib/scripts/get_ilbc_source.sh, + codecs/ilbc/LICENSE (added), codecs/ilbc/LPCencode.h (added), + codecs/ilbc/StateSearchW.h (added), codecs/ilbc/iCBConstruct.c + (added): Include iLBC source code for distribution with Asterisk + This patch includes the iLBC source code for distribution with + Asterisk. Clarification regarding the iLBC source code was + provided by Google, and the appropriate licenses have been + included in the codecs/ilbc folder. Review: + https://reviewboard.asterisk.org/r/1675 Review: + https://reviewboard.asterisk.org/r/1649 (closes issue: + ASTERISK-18943) Reporter: Leif Madsen Tested by: Matt Jordan + ........ Merged revisions 351450 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351451 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-18 16:02 +0000 [r351409] Stefan Schmidt + + * /, channels/chan_sip.c: The get_pai function in chan_sip.c didn't + recognized a proper callerid name and number from a + P-Asserted-Identity cause the header parsing logic was wrong. + Changing the parsing functions to the sip header parsing APIs in + reqresp_parser.h solves this problem. Review: + https://reviewboard.asterisk.org/r/1673 Reviewed by: wdoekes2 and + Mark Michelson ........ Merged revisions 351396 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351408 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-17 19:45 +0000 [r351360] Walter Doekes + + * Makefile: Fix support for parallel building with make (-j). + Previously make -j would cause a race between doing cleanup + of certain files (defaults.h, menuselect, ...) and creating them + anew. Add a new target that depends on cleanup only and has a + submake doing the rest as command string. This way the cleanup + goes first. (closes issue ASTERISK-18751) Tested by: Jeremy + Kister Reviewed by: Paul Belanger Review: + https://reviewboard.asterisk.org/r/1660 + +2012-01-17 17:23 +0000 [r351311] Mark Michelson + + * res/res_rtp_asterisk.c, /: Eliminate odd initialization of + probation variable. ........ Merged revisions 351306 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351308 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-17 17:15 +0000 [r351290] Jonathan Rose + + * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES: Adds + pjmedia probation concepts to res_rtp_asterisk's learning mode. + In order to better handle RTP sources with strictrtp enabled + (which is now default in 10) using the learning mode to figure + out new sources when they change is handled by checking for a + number of consecutive (by sequence number) packets received to an + rtp struct based on a new configurable value called 'probation'. + Also, during learning mode instead of liberally accepting all + packets received, we now reject packets until a clear source has + been determined. Review: https://reviewboard.asterisk.org/r/1663/ + ........ Merged revisions 351287 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351289 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-17 16:56 +0000 [r351288] Mark Michelson + + * /, channels/chan_sip.c: Use built-in parsing functions for + Contact and Record-Route headers. If a Contact or a Record-Route + header had a quoted string with an item in angle brackets, then + we would mis-parse it. For instance, "Bob <1234>" + <1234@example.org> would be misparsed as having the URI "1234" + The fix for this is to use parsing functions from + reqresp_parser.h since they are heavily tested and are awesome. + (issue ASTERISK-18990) ........ Merged revisions 351284 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351286 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-17 16:08 +0000 [r351235] Matthew Jordan + + * /, channels/chan_sip.c: Fix udptl issue with initial INVITE + introduced by r351027 When an inital INVITE occurs that contains + image media, a channel is not yet associated with the SIP dialog. + The file descriptor associated with the udptl session needs to be + set in initialize_udptl or in sip_new to account for this + scenario. ........ Merged revisions 351233 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351234 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-17 01:48 +0000 [r351184] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 351183 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r351183 | russell | 2012-01-16 20:43:19 -0500 + (Mon, 16 Jan 2012) | 29 lines Merged revisions 351182 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r351182 | russell | 2012-01-16 20:37:03 -0500 (Mon, 16 Jan 2012) + | 22 lines Add some missing locking in chan_sip. This patch adds + some missing locking to the function + send_provisional_keepalive_full(). This function is called from + the scheduler, which is processed in the SIP monitor thread. The + associated channel (or pbx) thread will also be using the same + sip_pvt and ast_channel so locking must be used. The + sip_pvt_lock_full() function is used to ensure proper locking + order in a safe manner. In passing, document a suspected + reference counting error in this function. The "fix" is left + commented out because when the "fix" is present, crashes occur. + My theory is that fixing it is exposing a reference counting + error elsewhere, but I don't know where. (Or my analysis of this + being a problem could have been completely wrong in the first + place). Leave the comment in the code for so that someone may + investigate it again in the future. Also add a bit of doxygen to + transmit_provisional_response(). (closes issue ASTERISK-18979) + Review: https://reviewboard.asterisk.org/r/1648 ........ + ................ + +2012-01-16 21:50 +0000 [r351082-351143] Terry Wilson + + * /, channels/chan_sip.c: Ensure ACK retransmit & hangup on non-200 + response to INVITE When handling a non-2xx final response on an + INVITE transaction, we have to keep the transaction around after + we send an ACK in case we receive a retransmission of the + response so we can re-transmit the ACK, but also tear down the + ast_channel as soon as we transmit the ACK. Before this patch, we + could fail at both of these things. Calling + sip_alreadygone/needdestroy prevented us from keeping the + transaction up and retransmitting the ACK, and queueing + CONGESTION was not sufficient to cause the channel to be torn + down when originating calls via the CLI, for example. This patch + queues a hangup with CONGESTION instead of just queueing + CONGESTION for these responses and removes the sip_alreadygone + and sip_needdestroy calls from handle_response_invite on non-2xx + responses. It relies on the hangup calling sip_scheddestroy. For + more information, see section 17.1.1.1 of RFC 3261. (closes issue + ASTERISK-17717) Review: https://reviewboard.asterisk.org/r/1672/ + ........ Merged revisions 351130 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351131 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c: Don't prematurely stop SIP session timer + When Asterisk is the UAS (incoming call, endpoint is re-inviting) + the SIP session timer expires after half the time the sip + endpoint indicates in the Session-expires header in + proc_session_timer(). The session timer was being stopped totally + and being handled as an error case instead of running again until + the second expiry. This patch treats the half-time expiry as a + non-error case and continues the timer until the true expiry. + (closes issue ASTERISK-18996) Reported by: Thomas Arimont Tested + by: Thomas Arimont Patches: session_timer_fix.diff by Terry + Wilson (License #5357) based on session_timer.patch by Thomas + Arimont (License #5525) ........ Merged revisions 351080 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351081 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-16 19:49 +0000 [r351079] Tilghman Lesher + + * main/ast_expr2.y, CHANGES, main/ast_expr2.c: Add ABS() absolute + value function to the expression parser. + +2012-01-16 19:13 +0000 [r351029] Matthew Jordan + + * /, channels/chan_sip.c: Create and initialize udptl only when + dialog negotiates for image media Prior to this patch, the udptl + struct was allocated and initialized when a dialog was associated + with a peer that supported T.38, when a new SIP channel was + allocated, or what an INVITE request was received. This resulted + in any dialog associated with a peer that supported T.38 having + udptl support assigned to it, including the UDP ports needed for + communication. This occurred even in non-INVITE dialogs that + would never send image media. This patch creates and initializes + the udptl structure only when the SDP for a dialog specifies that + image media is supported, or when Asterisk indicates through the + appropriate control frame that a dialog is to support T.38. + (closes issue ASTERISK-16698) Reported by: under Tested by: + Stefan Schmidt Patches: udptl_20120113.diff uploaded by mjordan + (License #6283) (closes issue ASTERISK-16794) Reported by: Elazar + Broad Tested by: Stefan Schmidt review: + https://reviewboard.asterisk.org/r/1668/ ........ Merged + revisions 351027 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 351028 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-16 17:12 +0000 [r350979] Sean Bright + + * /, main/db.c: Sort the output of 'database showkey' as well. You + can pass wildcards (%) to the database CLI commands, so this will + sort the returned list of matches. ........ Merged revisions + 350978 from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-16 17:07 +0000 [r350977] Joshua Colp + + * main/rtp_engine.c, /: Add missing code to set direct RTP setup + information during dialing. ........ Merged revisions 350975 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350976 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-16 14:31 +0000 [r350939] Sean Bright + + * /, main/db.c: Sort the output of 'database show' by key. This + more closely mimics the behavior of 'database show' before the + conversion to sqlite3. ........ Merged revisions 350938 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-15 20:16 +0000 [r350887-350890] Walter Doekes + + * /, main/asterisk.c: Allow only one thread at a time to do + asterisk cleanup/shutdown. Add locking around the + really-really-quit part of the core stop/restart part. Previously + more than one thread could be called to do cleanup, causing + atexit handlers to be run multiple times, in turn causing + segfaults. (issue ASTERISK-18883) Reviewed by: Terry Wilson + Review: https://reviewboard.asterisk.org/r/1662/ Review: + https://reviewboard.asterisk.org/r/1658/ ........ Merged + revisions 350888 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350889 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, utils/extconf.c: Fix -Werror=unused-but-set-variable compile + error in utils/extconf.c. Note that I'm not confirming legitimacy + of having that file in tree at all. Is anyone using + aelparse/conf2ael? (issue ASTERISK-15350) ........ Merged + revisions 350885 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350886 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-14 16:43 +0000 [r350791-350839] Kevin P. Fleming + + * /, configure, autoconf/ast_gcc_attribute.m4, configure.ac, + autoconf/libcurl.m4: Ensure that all AC_LANG_PROGRAM calls in the + configure script are properly quoted. Recent versions of autoconf + (2.68 on my system) won't properly process the configure script + unless every call to AC_LANG_PROGRAM is m4-quoted. Many calls in + the script were, but many were not. This patch corrects the + unquoted calls. ........ Merged revisions 350837 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350838 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_h323.c, addons/chan_mobile.c, + res/res_pktccops.c, contrib/scripts/install_prereq: Multiple + revisions 350788-350789 ........ r350788 | kpfleming | 2012-01-14 + 09:22:33 -0600 (Sat, 14 Jan 2012) | 8 lines Ensure that two + prerequisites are properly installed on Debian-style + distributions. * Don't specify a specific version of libgmime; + newer versions are available now and acceptable. * Install + libsrtp so that res_srtp can be built. ........ r350789 | + kpfleming | 2012-01-14 09:23:32 -0600 (Sat, 14 Jan 2012) | 3 + lines Correct some 'set-but-not-used' variable warnings. ........ + Merged revisions 350788-350789 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350790 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-13 22:17 +0000 [r350738] Kinsey Moore + + * /, include/asterisk/autoconfig.h.in: Run bootstrap.sh for the for + the ASTERISK-18929 fix configure and autoconfig.h.in were not + regenerated when the fix was committed. ........ Merged revisions + 350736 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 350737 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-13 21:52 +0000 [r350735] Richard Mudgett + + * /, configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample: + Correct eventtype names in cel_odbc and cel_pgsql sample files + ........ Merged revisions 350733 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350734 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-13 21:42 +0000 [r350732] Kinsey Moore + + * /, configure.ac, bootstrap.sh, main/asterisk.c: Make sure + asterisk builds on OpenBSD OpenBSD defines SO_PEERCRED, but it + returns a 'struct sockpeercred', not 'struct ucred', which causes + compilation of main/asterisk.c to fail in read_credentials(). + This allows configure to check for sockpeercred and asterisk to + deal with it properly. (closes issue ASTERISK-18929) Reported-by: + Barry Miller Patch-by: Barry Miller ........ Merged revisions + 350730 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 350731 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-13 20:32 +0000 [r350681] Mark Michelson + + * /, channels/sip/config_parser.c: Set port to a default sane value + if a bogus one is provided when parsing hostnames. ........ + Merged revisions 350679 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350680 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-13 18:52 +0000 [r350605-350644] Richard Mudgett + + * main/features.c: Remove some dead code in ast_bridge_call(). None + of the parameters to ast_bridge_call() can be NULL for the bridge + to work so no need to check for it. + + * configs/cel_sqlite3_custom.conf.sample, cel/cel_odbc.c, + configs/cel.conf.sample, /, cel/cel_manager.c, + configs/cel_pgsql.conf.sample, configs/cel_odbc.conf.sample, + main/cel.c, configs/cel_custom.conf.sample: Add missing CEL + logging fields to various CEL backends. Multiple revisions + 350555,350571 ........ r350555 | rmudgett | 2012-01-13 11:12:51 + -0600 (Fri, 13 Jan 2012) | 12 lines Add missing CEL logging + fields to various CEL backends. * Add missing eventextra to + cel_psql.c and cel_odbc.c. * Add missing PeerAccount and + EventExtra to cel_manager.c. * Add missing userdeftype support + for cel_custom.conf.sample and cel_sqlite3_custom.conf.sample. + (closes issue ASTERISK-17190) Reported by: Bryant Zimmerman + ........ r350571 | rmudgett | 2012-01-13 11:23:57 -0600 (Fri, 13 + Jan 2012) | 8 lines Use compatible names for event extra data for + various CEL backends. * Change eventextra to extra in cel_psql.c + and cel_odbc.c. * Change EventExtra to Extra in cel_manager.c. + (issue ASTERISK-17190) ........ Merged revisions 350555,350571 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 350585 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-13 17:00 +0000 [r350551-350554] Matthew Jordan + + * /, apps/app_queue.c: Realtime queues failed to load queue + information without queue member table Previously, realtime + queues could be loaded without defining the queue member table. + This allowed for queue members to be dynamic, while the realtime + queue definitions could exist in some backing storage. Revision + 342223 broke this when it changed the return value for + realtime_multientry to return NULL when no results are returned. + Previously, an empty ast_config object was expected. (closes + issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene + Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt + Jordan (license 6283) ........ Merged revisions 350552 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350553 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, bridges/bridge_builtin_features.c, channels/chan_bridge.c, + include/asterisk/bridging.h, apps/app_confbridge.c, + main/bridging.c: Fix crash from bridge channel hangup race + condition in ConfBridge This patch addresses two issues in + ConfBridge and the channel bridge layer: 1. It fixes a race + condition wherein the bridge channel could be hung up 2. It + removes the deadlock avoidance from the bridging layer and makes + the bridge_pvt an ao2 ref counted object Patch by David Vossel + (mjordan was merely the commit monkey) (issue ASTERISK-18988) + (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested + by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by + David Vossel (license 5628) (closes issue ASTERISK-19100) + Reported by: Matt Jordan Tested by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/1654/ ........ Merged + revisions 350550 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-12 16:10 +0000 [r350503] Jonathan Rose + + * /, main/features.c: Adds peer to CEL report on CEL_BRIDGE_START + and CEL_BRIDGE_END (closes issue ASTERISK-17940) Reporter: Nic + Colledge Patches: features_18.patch uploaded by Nic Colledge + (license 6245) ........ Merged revisions 350501 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350502 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-11 22:53 +0000 [r350416-350454] Richard Mudgett + + * /, main/cel.c: Remove extraneous BRIDGEPEER AMI VarSet event on a + CEL dummy channel. (closes issue ASTERISK-19180) Reported by: + Corey Farrell Patches: asterisk_cel_noevent_varset.diff (license + #5909) patch uploaded by Corey Farrell ........ Merged revisions + 350452 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 350453 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_dial.c, /, CHANGES, apps/app_followme.c: Make FollowMe + optionally update connected line information when the accepting + endpoint is bridged. Like Dial and Queue, FollowMe needs to deal + with AST_CONTROL_CONNECTED_LINE information so when the parties + are initially bridged, the connected line information will be + correct. * Added the 'I' option just like the app_dial and + app_queue 'I' option. * Made 'N' option ignored if the call is + already answered. (closes issue ASTERISK-18969) Reported by: + rmudgett Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/1656/ ........ Merged + revisions 350364 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350415 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-11 19:19 +0000 [r350365] Terry Wilson + + * main/channel.c: Always treat arguments to get_by_name_cb as + strings Initially, support was left in for the old style of + searching, even though it wasn't actually used. In the case of + name_len != 0, the OBJ_KEY flag isn't passed because we aren't + matching on a full key and therefor can't use the hash function + to optimize. The code left in to support the old way of searching + unfortunately treated a prefix search like this as though an + ast_channel struct was passed as an arg and caused a crash. This + patch also adds needed parentheses around some matching + conditions. (closes issue ASTERISK-19182) + +2012-01-10 22:10 +0000 [r350273-350313] Richard Mudgett + + * /, funcs/func_lock.c: Fix absolute/relative time mismatch in LOCK + function. The time passed by the LOCK function to an internal + function was relative time when the function expected absolute + time. * Don't use C++ keywords in get_lock(). (closes issue + ASTERISK-16868) Reported by: Andrey Solovyev Patches: + 20101102__issue18207.diff.txt (license #5003) patch uploaded by + Andrey Solovyev (modified) ........ Merged revisions 350311 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350312 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/channel.c: Fix compiler warnings reported by gcc v4.2.4. + +2012-01-09 22:15 +0000 [r350223] Terry Wilson + + * main/udptl.c, apps/app_dahdibarge.c, addons/chan_ooh323.c, + channels/chan_local.c, main/rtp_engine.c, main/say.c, + apps/app_record.c, apps/app_test.c, channels/console_video.c, + apps/app_alarmreceiver.c, apps/app_chanisavail.c, + bridges/bridge_multiplexed.c, channels/chan_iax2.c, + main/indications.c, main/cli.c, channels/chan_dahdi.c, + channels/sig_analog.c, channels/chan_skinny.c, main/features.c, + apps/app_dumpchan.c, pbx/pbx_realtime.c, apps/app_amd.c, + channels/chan_alsa.c, apps/app_externalivr.c, main/bridging.c, + apps/app_milliwatt.c, channels/sig_ss7.c, apps/app_dial.c, + main/pbx.c, apps/app_page.c, apps/app_softhangup.c, + apps/app_fax.c, apps/app_dahdiras.c, channels/chan_agent.c, + apps/app_disa.c, include/asterisk/channel.h, main/aoc.c, + apps/app_talkdetect.c, main/cel.c, res/res_mutestream.c, + res/res_monitor.c, apps/app_playback.c, channels/chan_misdn.c, + funcs/func_channel.c, apps/app_macro.c, apps/app_mixmonitor.c, + apps/app_chanspy.c, apps/app_voicemail.c, res/res_calendar.c, + channels/chan_unistim.c, channels/chan_vpb.cc, main/ccss.c, + apps/app_meetme.c, apps/app_readexten.c, res/res_musiconhold.c, + main/autochan.c, channels/chan_gtalk.c, apps/app_followme.c, + res/res_jabber.c, main/cdr.c, main/channel.c, main/dial.c, + channels/chan_phone.c, main/manager.c, funcs/func_groupcount.c, + funcs/func_audiohookinherit.c, funcs/func_frame_trace.c, + res/res_agi.c, apps/app_minivm.c, main/app.c, + apps/app_confbridge.c, apps/app_rpt.c, addons/chan_mobile.c, + apps/app_parkandannounce.c, channels/chan_mgcp.c, + apps/app_jack.c, apps/app_adsiprog.c, channels/chan_sip.c, + res/res_fax.c, apps/app_waitforsilence.c, funcs/func_lock.c, + main/channel_internal_api.c (added), res/res_adsi.c, + pbx/pbx_lua.c, channels/chan_console.c, apps/app_getcpeid.c, + channels/sig_pri.c, apps/app_queue.c, channels/chan_oss.c, + funcs/func_global.c, channels/chan_usbradio.c, + channels/chan_jingle.c, apps/app_flash.c, + apps/app_directed_pickup.c, main/abstract_jb.c, main/file.c, + channels/chan_h323.c, res/snmp/agent.c, pbx/pbx_dundi.c, + apps/app_sms.c, channels/chan_nbs.c, apps/app_stack.c, + main/dsp.c: Replace direct access to channel name with accessor + functions There are many benefits to making the ast_channel an + opaque handle, from increasing maintainability to presenting ways + to kill masquerades. This patch kicks things off by taking things + a field at a time, renaming the field to + '__do_not_use_${fieldname}' and then writing setters/getters and + converting the existing code to using them. When all fields are + done, we can move ast_channel to a C file from channel.h and lop + off the '__do_not_use_'. This patch sets up + main/channel_interal_api.c to be the only file that actually + accesses the ast_channel's fields directly. The intent would be + for any API functions in channel.c to use the accessor functions. + No more monkeying around with channel internals. We should use + our own APIs. The interesting changes in this patch are the + addition of channel_internal_api.c, the moving of the AST_DATA + stuff from channel.c to channel_internal_api.c (note: the + AST_DATA stuff will have to be reworked to use accessor functions + when ast_channel is really opaque), and some re-working of the + way channel iterators/callbacks are handled so as to avoid + creating fake ast_channels on the stack to pass in matching data + by directly accessing fields (since "name" is a stringfield and + the fake channel doesn't init the stringfields, you can't use the + ast_channel_name_set() function). I went with + ast_channel_name(chan) for a getter, and + ast_channel_name_set(chan, name) for a setter. The majority of + the grunt-work for this change was done by writing a semantic + patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: + https://reviewboard.asterisk.org/r/1655/ + +2012-01-09 21:56 +0000 [r350222] Richard Mudgett + + * /, channels/chan_iax2.c: Fix joinable thread terminating without + joiner memory leak in chan_iax.c. The iax2_process_thread() can + exit without anyone waiting to join the thread. If noone is + waiting to join the thread then a large memory leak occurs. * + Made iax2_process_thread() deatach itself if nobody is waiting to + join the thread. (closes issue ASTERISK-17339) Reported by: + Tzafrir Cohen Patches: + asterisk-1.8.4.4-chan_iax2-detach-thread-on-non-stop-exit.patch + (license #5617) patch uploaded by Alex Villacis Lasso (modified) + (closes issue ASTERISK-17825) Reported by: wangjin ........ + Merged revisions 350220 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350221 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-09 19:37 +0000 [r350181] Walter Doekes + + * /, main/db.c: Fix shutdown handling of sqlite3 astdb. If a + db_sync was scheduled just before shutdown, the atexit code + calling db_sync would have no effect, causing the astdb commit + thread to stay alive. This caused the SIP/realtime_sipregs test + to fail. (The fallback kill would run the atexit code again and + that would wreak havoc.) This fixes that the atexit kill + condition is picked up properly. (closes issue ASTERISK-18883) + Reviewed by: Terry Wilson Review: + https://reviewboard.asterisk.org/r/1659 ........ Merged revisions + 350180 from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-09 18:58 +0000 [r350077-350130] Richard Mudgett + + * /, contrib/scripts/valgrind_compare (added): Multiple revisions + 350127-350128 ........ r350127 | rmudgett | 2012-01-09 12:40:33 + -0600 (Mon, 09 Jan 2012) | 12 lines Update contrib script + live_ast to invoke Asterisk with valgrind and suppression file. * + Added valgrind_compare script to compare two valgrind log files + for differences. (issue ASTERISK-17339) Reported by: Tzafrir + Cohen Patches: valgrind_compare (license #5035) script uploaded + by Tzafrir Cohen live_ast_valgrind.diff (license #5035) patch + uploaded by Tzafrir Cohen live_ast_valgrind_v2.diff (license + #5185) patch uploaded by Paul Belanger ........ r350128 | + rmudgett | 2012-01-09 12:54:56 -0600 (Mon, 09 Jan 2012) | 11 + lines live_ast: valgrind: run asterisk under valgrind Adds a new + sub-command, "valgrind" to live_ast. It runs asterisk under + valgrind. The extra command-line parameters are passed to + Asterisk as usual, and parameters to valgrind are passed through + LIVE_AST_VALGRIND_ARGS in live.conf . Review: + https://reviewboard.asterisk.org/r/1109/ Merged revisions 326636 + from http://svn.asterisk.org/svn/asterisk/branches/10 ........ + Merged revisions 350127-350128 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350129 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/asterisk.c: Make Asterisk -x command line parameter imply + -r parameter presence. The Asterisk -x command line parameter is + documented inconsistently. * Made the -x documentation and + behavior consistent. * Since this is also a new year, updated the + copyright notices while here. (closes issue ASTERISK-19094) + Reported by: Eugene Patches: + issueA19094_correct_asterisk_option_x.patch (license #5674) patch + uploaded by Walter Doekes (modified) Tested by: Eugene ........ + Merged revisions 350075 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 350076 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-09 15:40 +0000 [r350025] Kinsey Moore + + * /, apps/app_meetme.c: Prevent SLA settings from getting wiped out + on reload If SLA was reloaded without the config file being + changed, current settings got wiped out before the SLA reload + code decided it wasn't going to reload the file since nothing was + changed. Moving the settings reset later in the reload process + fixes this. (closes issue AST-744) ........ Merged revisions + 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 350024 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-06 23:31 +0000 [r349978] Terry Wilson + + * /, channels/chan_sip.c: Don't leak CID in From header when + presentation=unavailable When someone does + Set(CALLERPRES()=unavailable) (or + Set(CALLERID(pres)=unavailable)) when sendrpid=no, the From + header shows "Anonymous" . When + sendrpid=yes/pai, the From header will still display the callerid + info, even though we supply an rpid header with the anonymous + info. It seems like we shouldn't leak that info in any case. + Skimming http://tools.ietf.org/html/draft-ietf-sip-privacy-04 + seems to indicate that one shouldn't send identifying info in the + From in this case. This patch anonymizes the From header as well + even when sendrpid=yes/pai. (closes issue ASTERISK-16538) Review: + https://reviewboard.asterisk.org/r/1649/ ........ Merged + revisions 349968 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349977 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-06 21:26 +0000 [r349929] Kinsey Moore + + * /, pbx/pbx_lua.c: Fix lua goto detection to prevent unexpected + behavior with confbridge A bug in the pbx_lua goto detection was + causing the dialplan to hangup unexpectedly after confbridge + exited if it had called lua dialplan code during execution. + Patch-by: Timo Teras Acked-by: Matt Nicholson (closes issue + ASTERISK-18976) ........ Merged revisions 349928 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-06 16:50 +0000 [r349874] Richard Mudgett + + * /, apps/app_followme.c: Fix memory leaks in app_followme + find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt + Jordan ........ Merged revisions 349872 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349873 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-05 23:58 +0000 [r349823] Matthew Jordan + + * /, res/res_fax.c: Fix premature free'ing of the frame committed + in r349608 Even though we set the frame to the ast_null_frame and + return that, the caller of the frame hook may still need the + frame. This now is a bit more careful about when it frees the + frame, i.e., only under the same conditions that applied when we + duplicated it in the first place. ........ Merged revisions + 349822 from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-05 23:47 +0000 [r349782-349821] Richard Mudgett + + * /, cel/cel_sqlite3_custom.c: Make not assume that the + cel_sqlite3_custom SQL table primary key is AcctId. If a table is + created by some other application and the primary key is not + named "AcctId", cel/cel_sqlite3_custom.c will always try to + create the table and fail because it already exists. * Change the + SQL table query to not require AcctId as the primary key. (closes + issue ASTERISK-18963) Reported by: socketpair Patches: fix.patch + (license #6337) patch uploaded by socketpair ........ Merged + revisions 349819 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349820 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * UPGRADE.txt, pbx/pbx_config.c: Make pbx_config.c use Gosub + instead of Macro call for stdexten. Users created by users.conf + with hasvoicemail=yes have been documented as using a Gosub to + stdexten since v1.6.0. However, the code still generates dialplan + to access stdexten as a Macro as documented in v1.4; which does + not work with the newer extensions.conf.sample file. * Make + generated dialplan access the stdexten dialplan with the + documented Gosub instead of the older Macro style. (closes issue + ASTERISK-18809) Reported by: Jay Allen Patches: + gosub_patch-pbx_config.patch (license #6323) patch uploaded by + Jay Allen (modified) Tested by: rmudgett + +2012-01-05 22:11 +0000 [r349733] Kinsey Moore + + * /, main/file.c: Allow playback of formats that don't support + seeking ast_streamfile previously did unconditional seeking on + files that broke playback of formats that don't support that + functionality. This patch avoids the seek that was causing the + problem. This regression was introduced in r158062. (closes issue + ASTERISK-18994) Patch-by: Timo Teras ........ Merged revisions + 349731 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 349732 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-05 22:02 +0000 [r349674-349730] Jonathan Rose + + * /, main/dsp.c: Fix an issue where dsp.c would interpret multiple + dtmf events from a single key press. When receiving calls from a + mobile phone into a DISA system on a connection with significant + interference, the reporter's Asterisk system would interpret DTMF + incorrectly and replicate digits received. This patch resolves + that by increasing the number of frames a mismatch has to be + detected before assuming the DTMF is over by 1 frame and adjusts + dtmf_detect function to reset hits and misses only when an edge + is detected. (closes issue ASTERISK-17493) Reported by: Alec + Davis Patches: bug18904-refactor.diff.txt uploaded by Alec Davis + (license 5546) Review: https://reviewboard.asterisk.org/r/1130/ + ........ Merged revisions 349728 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349729 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/asterisk.c: Ensures Asterisk closes when receiving + terminal signals in 'no fork' mode. When catching a signal, in no + fork mode the console thread is identical to the thread + responsible for catching the signal and closing Asterisk, which + requires it to first dispense with the console thread. Prior to + this patch, if these threads were identical, upon receiving a + killing signal, the thread will send an URG signal to itself, + which we also catch and then promptly do nothing with. Obviously + this isn't useful behavior. (closes issue ASTERISK-19127) + Reported By: Bryon Clark Patches: quit_on_signals.patch uploaded + by Bryon Clark (license 6157) ........ Merged revisions 349672 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 349673 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-04 22:23 +0000 [r349609-349634] Matthew Jordan + + * /, apps/confbridge/conf_config_parser.c: Fix for ConfBridge + config parser unlocking channel mutex too many times When looking + up a ConfBridge profile, the config parser would, if it found a + channel datastore on the channel requesting the bridge profile, + unlock the channel mutex twice. Since that's a little aggressive, + it now only unlocks it once. (closes issue ASTERISK-19042) + Reported by: Matt Jordan Tested by: Matt Jordan Patches: 19042 + uploaded by David Vossel (license 5628) ........ Merged revisions + 349619 from http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, res/res_fax.c: Free successfully translated frame in + fax_gateway_framehook A frame that is translated via + ast_translate is also duplicated via ast_frdup. This will + allocate a new frame on the heap, which needs to be free'd at the + appropriate time. This issue reporter used valgrind to find that + this occurred in res_fax's fax_gateway_framehook; a quick search + through the code showed that only place this was currently not + handling the translatted frame properly. (closes issue + ASTERISK-19133) Reported by: Sylvain Rochet ........ Merged + revisions 349608 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-04 20:55 +0000 [r349560] Richard Mudgett + + * channels/chan_dahdi.c, /: Fix segfault in chan_dahdi for + CHANNEL(dahdi_span) evaluation on hangup. * Added NULL private + pointer checks in the following chan_dahdi channel callbacks: + dahdi_func_read(), dahdi_func_write(), dahdi_setoption(), and + dahdi_queryoption(). (closes issue ASTERISK-19142) Reported by: + Diego Aguirre Tested by: rmudgett ........ Merged revisions + 349558 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 349559 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-04 20:24 +0000 [r349506-349535] Kinsey Moore + + * contrib/init.d/rc.debian.asterisk, /: Make debian init script + conform to the LSB standard Previously, this init script would + return 1 if Asterisk was already running. This is incorrect + behavior according to the LSB standard and has been fixed by + returning 0 instead. (closes issue ASTERISK-17958) Reported-by: + johnc ........ Merged revisions 349529 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349532 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, contrib/scripts/autosupport.8, contrib/scripts/autosupport: + Update autosupport script and man page Added information + collection from the output of the utilities: top, free, uptime, + ifconfig Added information collection from the output of the + Asterisk command 'dahdi show status' Added option / flag '-n, + --non-interactive' Updated man page to reflect new option / flag + '-n, --non-interactive' Patch-by: John Bigelow (itzanger) (closes + issue AST-749) ........ Merged revisions 349504 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349505 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-01-04 19:53 +0000 [r349452-349503] Jonathan Rose + + * /, channels/chan_sip.c: Adds Subscription-State header to notify + with call completion. per RFC3265 (Closes issue ASTERISK-17953) + Reported by: George Konopacki Patches: 19400.patch uploaded by + mmichelson (license 5049) ........ Merged revisions 349482 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349502 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/pbx.c, /: Fix documentation for SayNumber to reflect the + fact that language is changed in CHANNEL() (closes issue + ASTERISK-18962) reported by: Nir Simionovich ........ Merged + revisions 349450 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349451 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-31 15:48 +0000 [r349409-349410] Russell Bryant + + * channels/chan_sip.c: Fix some minor formatting issues based on + coding guidelines. + + * channels/sip/include/dialog.h, channels/chan_sip.c, + include/asterisk/astobj2.h, main/astobj2.c: Constify tag argument + in REF_DEBUG related code. + +2011-12-29 15:16 +0000 [r349341] Matthew Jordan + + * main/rtp_engine.c, /: Handle AST_CONTROL_UPDATE_RTP_PEER frames + in local bridge loop Failing to handle + AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop + causes the loop to exit prematurely. This causes a variety of + negative side effects, depending on when the loop exits. This + patch handles the frame by essentially swallowing the frame in + the local loop, as the current channel drivers expect the RTP + bridge to handle the frame, and, in the case of the local bridge + loop, no additional action is necessary. (issue ASTERISK-19040) + (issue ASTERISK-19128) (issue ASTERISK-17725) (issue + ASTERISK-18340) (closes issue ASTERISK-19095) Reported by: Stefan + Schmidt Tested by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/1640/ ........ Merged + revisions 349339 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349340 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-28 21:39 +0000 [r349291] Sean Bright + + * /, main/audiohook.c: Use ast_audiohook_write_list_empty to + determine if our lists are empty instead of duplicating that + logic. ........ Merged revisions 349289 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349290 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-28 19:00 +0000 [r349249-349251] Kevin P. Fleming + + * utils, /: Tell Subversion to gnore the 'astdb2bdb' binary file if + it exists. ........ Merged revisions 349250 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, res/res_fax.c, include/asterisk/dsp.h, + include/asterisk/res_fax.h, res/res_fax_spandsp.c, main/dsp.c: + Improve T.38 gateway V.21 preamble detection. This commit removes + the V.21 preamble detection code previously added to the generic + DSP implementation in Asterisk, and instead enhances the res_fax + module to be able to utilize V.21 preamble detection + functionality made available by FAX technology modules. This + commit also adds such support to res_fax_spandsp, which uses the + Spandsp modem tone detection code to do the V.21 preamble + detection. There should be no functional change here, other than + much more reliable V.21 preamble detection (and thus T.38 gateway + initiation). ........ Merged revisions 349248 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-27 20:55 +0000 [r349196] Matthew Jordan + + * /, res/res_timing_pthread.c, include/asterisk/module.h, + res/res_timing_dahdi.c, res/res_timing_timerfd.c, + res/res_musiconhold.c: Fix timing source dependency issues with + MOH Prior to this patch, res_musiconhold existed at the same + module priority level as the timing sources that it depends on. + This would cause a problem when music on hold was reloaded, as + the timing source could be changed after res_musiconhold was + processed. This patch adds a new module priority level, + AST_MODPRI_TIMING, that the various timing modules are now loaded + at. This now occurs before loading other resource modules, such + that the timing source is guaranteed to be set prior to resolving + the timing source dependencies. (closes issue ASTERISK-17474) + Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, + Wes Van Tlghem, elguero, Thomas Arimont Patches: + asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff + uploaded by elguero (License #5026) + asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff + uploaded by elguero (License #5026) + asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by + elguero (License #5026) Review: + https://reviewboard.asterisk.org/r/1578/ ........ Merged + revisions 349194 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349195 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-27 17:17 +0000 [r349146] Sean Bright + + * /, main/audiohook.c: Once an audiohook is attached to a channel, + we continue to transcode all of the frames, even after all of the + hooks are detached. This patch short-cicuits us out before we + transcode unnecessarily. ........ Merged revisions 349144 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 349145 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-23 21:19 +0000 [r349106] Matthew Jordan + + * contrib/realtime/mysql/voicemail.sql, + configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: + Allow overriding of IMAP server settings on a user by user basis + This patch allows the imapserver, imapport, and imapflags + settings to be overridden for any voicemail user. It also + documents the settings in the sample voicemail.conf file, and + updates the voicemail schema to allow storage of those columns. + (closes issue ASTERISK-16489) Reporter: Hubert Mickael Tested by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/1614/ + +2011-12-23 20:42 +0000 [r349097-349098] Jonathan Rose + + * channels/chan_sip.c, main/features.c, configs/sip.conf.sample, + channels/sip/include/sip.h: INFO/Record request configurable to + use dynamic features Adds two new options to SIP peers allowing + them to specify features (dynamic or builtin) to use when sending + INFO/record requests. Recordonfeature activates whatever feature + is specified when recieving a record: on request while + recordofffeature activates whatever feature is specified when + receiving a record: off request. Both of these features can be + disabled by setting the feature to an empty string. (closes issue + ASTERISK-16507) Reported by: Jon Bright Review: + https://reviewboard.asterisk.org/r/1634/ + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/sip/include/sip.h: chan_sip autocreatepeer=persist + option for auto-created peers to survive reload This patch moves + destruction of sip peers to immediately after the general section + of sip.conf is read so that autocreatepeer setting can be read + before deletion of peers. If autocreatepeer=persist at reload, + then peers created by the autocreatepeer setting will be skipped + when purging the current SIP peer list. (closes ASTERISK-16508) + Reported by: Kirill Katsnelson Patches: + 017797-kkm-persist-autopeers-1.8.patch uploaded by Kirill + Katsnelson (license 5845) + +2011-12-23 17:36 +0000 [r349046] Sean Bright + + * /, apps/app_chanspy.c: Merged revisions 349045 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r349045 | seanbright | 2011-12-23 12:32:33 -0500 + (Fri, 23 Dec 2011) | 25 lines Merged revisions 349044 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r349044 | seanbright | 2011-12-23 12:25:01 -0500 (Fri, 23 Dec + 2011) | 18 lines In ChanSpy, don't create audiohooks that will + never be used. When ChanSpy is initialized it creates and + attaches 3 audiohooks: 1) Read audio off of the channel that we + are spying on 2) Write audio to the channel that we are spying on + 3) Write audio to the channel that is bridged to the channel that + we are spying on. The first is always necessary, but the others + are used only when specific options are passed to the ChanSpy + application (B, d, w, and W to be specific). When those flags are + not passed, neither of those audiohooks are ever sent frames, but + we still try to process the hooks for each voice frame that we + recieve on the channel. So in short - only create and attach + audiohooks that we actually need. ........ ................ + +2011-12-23 15:26 +0000 [r348994] Kinsey Moore + + * apps/app_dial.c, /: Fix missing doc tags found while fixing + ASTERISK-18689 Add missing tags in app_dial + documentation. ........ Merged revisions 348992 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348993 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-23 02:35 +0000 [r348953] Richard Mudgett + + * main/pbx.c, /, channels/chan_sip.c, include/asterisk/pbx.h: Fix + extension state callback references in chan_sip. Chan_sip gives a + dialog reference to the extension state callback and assumes that + when ast_extension_state_del() returns, the callback cannot + happen anymore. Chan_sip then reduces the dialog reference count + associated with the callback. Recent changes (ASTERISK-17760) + have resulted in the potential for the callback to happen after + ast_extension_state_del() has returned. For chan_sip, this could + be very bad because the dialog pointer could have already been + destroyed. * Added ast_extension_state_add_destroy() so chan_sip + can account for the sip_pvt reference given to the extension + state callback when the extension state callback is deleted. * + Fix pbx.c awkward statecbs handling in + ast_extension_state_add_destroy() and handle_statechange() now + that the struct ast_state_cb has a destructor to call. * Ensure + that ast_extension_state_add_destroy() will never return -1 or 0 + for a successful registration. * Fixed pbx.c statecbs_cmp() to + compare the correct information. The passed in value to compare + is a change_cb function pointer not an object pointer. * Make + pbx.c ast_merge_contexts_and_delete() not perform callbacks with + AST_EXTENSION_REMOVED with locks held. Chan_sip is notorious for + deadlocking when those locks are held during the callback. * + Removed unused lock declaration for the pbx.c store_hints list. + (closes issue ASTERISK-18844) Reported by: rmudgett Review: + https://reviewboard.asterisk.org/r/1635/ ........ Merged + revisions 348940 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348952 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-22 22:39 +0000 [r348890] Matthew Jordan + + * cel/cel_pgsql.c, /: Fix for memory leaks / cleanup in cel_pgsql + There were a number of issues in cel_pgsql's pgsql_log method: * + If either sql or sql2 could not be allocated, the method would + return while the pgsql_lock was still locked * If the execution + of the log statement succeeded, the sql and sql2 structs were + never free'd * Reconnection successes were logged as ERRORs. In + general, the severity of several logging statements was reduced + (closes issue ASTERISK-18879) Reported by: Niolas Bouliane Tested + by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1624/ + ........ Merged revisions 348888 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348889 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-22 21:12 +0000 [r348849] Damien Wedhorn + + * channels/chan_skinny.c: Fix segfault on answer. Only + update/change RTP source if RTP has already been started and + connected to the subchannel. + +2011-12-22 20:44 +0000 [r348848] Matthew Jordan + + * /, main/say.c, main/file.c, main/app.c, apps/app_confbridge.c, + main/bridging.c: Add Asterisk TestSuite event hooks to support + ConfBridge testing This patch adds initial testsuite event hooks + so that ConfBridge tests can be executed in the Asterisk + TestSuite. (issue ASTERISK-19059) ........ Merged revisions + 348846 from http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-22 20:39 +0000 [r348847] Terry Wilson + + * /, include/asterisk/format_pref.h: Allow packetization vaules > + 127 According to the RTP packetization documentation, and the + maximum values listed in AST_FORMAT_LIST, we should support + values > that the signed char array that ast_codec_pref makes + available to store the value. All places in the code treat the + framing field as though it were an int array instaead of a char + array anyway, so this just fixes the type of the array. (closes + issue ASTERISK-18876) Review: + https://reviewboard.asterisk.org/r/1639/ ........ Merged + revisions 348833 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348845 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-21 20:13 +0000 [r348737-348794] Richard Mudgett + + * /, codecs/speex: Make codecs/speex ignore *.i files also. + ........ Merged revisions 348793 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/confbridge: Make apps/confbridge ignore *.i files also. + ........ Merged revisions 348790 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_iax2.c: Fix chan_iax2 to not report an RDNIS + number if it is blank. Some ISDN switches complain or block the + call if the RDNIS number is empty. * Made chan_iax2 not save a + RDNIS number into the ast_channel if the string is blank. This is + what other channel drivers do. (closes issue ASTERISK-17152) + Reported by: rmudgett ........ Merged revisions 348735 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348736 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-20 20:06 +0000 [r348698] Matthew Nicholson + + * contrib/scripts/safe_asterisk: This adds support for setting + several safe_asterisk parameters using environment variables and + also enables a custom run directory for asterisk (instead of + defaulting to /tmp). Patch by: Byron Clark (byronclark) (closes + ASTERISK-17810) + +2011-12-19 21:43 +0000 [r348649] Richard Mudgett + + * /, configure, configure.ac: Fix crashes on other platforms caused + by interference from Darwin weak symbol support. Support weak + symbols on a platform specific basis. The Mac OS X (Darwin) + support must be isolated from the other platforms because it has + caused other platforms to crash. Several other platforms + including Linux have GCC versions that define the weak attribute. + However, this attribute is only setup for use in the code by + Darwin. (closes issue ASTERISK-18728) Reported by: Ben Klang + Review: https://reviewboard.asterisk.org/r/1617/ ........ Merged + revisions 348647 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348648 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-19 19:55 +0000 [r348606] Leif Madsen + + * /, main/message.c: Update documentation for MESSAGE_SEND_STATUS + variable. (Closes issue ASTERISK-19056) Reported by: Yuri + Patches: 348360.diff uploaded by Yuri (license #5242) ........ + Merged revisions 348605 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-19 01:36 +0000 [r348567] Terry Wilson + + * /, res/res_srtp.c: Add a separate buffer for SRTCP packets The + function ast_srtp_protect used a common buffer for both SRTP and + SRTCP packets. Since this function can be called from multiple + threads for the same SRTP session (scheduler for SRTCP and + channel for SRTP) it was possible for the packets to become + corrupted as the buffer was used by both threads simultaneously. + This patch adds a separate buffer for SRTCP packets to avoid the + problem. (closes issue ASTERISK-18889, Reported/patch by Daniel + Collins) ........ Merged revisions 347995 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347996 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-18 18:29 +0000 [r348518] Kevin P. Fleming + + * /, configs/sip.conf.sample: Correct two flaws in sip.conf.sample + related to AST-2011-013. * The sample file listed *two* values + for the 'nat' option as being the default. Only 'force_rport' is + the default. * The warning about having differing 'nat' settings + confusingly referred to both peers and users. ........ Merged + revisions 348515 from + http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ + Merged revisions 348516 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348517 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-16 23:58 +0000 [r348466] Richard Mudgett + + * main/channel.c, /, main/features.c: Clean-up on isle five for + __ast_request_and_dial() and ast_call_forward(). * Add locking + when a channel inherits variables and datastores in + __ast_request_and_dial() and ast_call_forward(). Note: The + involved channels are not active so there was minimal potential + for problems. * Remove calls to ast_set_callerid() in + __ast_request_and_dial() and ast_call_forward() because the set + information is for the wrong direction. * Don't use C++ keywords + for variable names in ast_call_forward(). * Run the redirecting + interception macro if defined when forwarding a call in + ast_call_forward(). Note: Currently will never execute because + the only callers that supply a calling channel supply a hungup or + zombie channel. * Make feature_request_and_dial() put the + transferee into autoservice when it calls ast_call_forward() in + case a redirection interception macro is run. Note: Currently + will never happen because the caller channel (Party B) is always + hungup at this time. * Make feature_request_and_dial() ignore the + AST_CONTROL_PROCEEDING frame to silence a log message. ........ + Merged revisions 348464 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348465 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-16 22:00 +0000 [r348416] Jonathan Rose + + * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: + Voicemail with the saycid option will now play a caller's name + based on cid if available. In order to check the availability of + the caller's name, app_voicemail will check for an audio file in + /recordings/callerids/ This change sets a precedent + for where to put recordings of names. Currently the idea is that + recordings here could also be used for applications like + confbridge and meetme to find recorded names in this folder from + callerid (when another recording isn't available) (closes issue + ASTERISK-18565) Reporter: Russell Brown Patches: r uploaded by + Russel Brown (license 6182) + +2011-12-16 21:30 +0000 [r348312-348408] Richard Mudgett + + * main/channel.c, /: Fix cut and past error in ast_call_forward(). + (issue ASTERISK-18836) ........ Merged revisions 348401 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348405 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/channel.c, main/pbx.c, /, apps/app_authenticate.c, + funcs/func_cdr.c, main/features.c, include/asterisk/cdr.h, + apps/app_followme.c, apps/app_queue.c, res/res_monitor.c: Fix + crash during CDR update. The ast_cdr_setcid() and + ast_cdr_update() were shown in ASTERISK-18836 to be called by + different threads for the same channel. The channel driver thread + and the PBX thread running dialplan. * Add lock protection around + CDR API calls that access an ast_channel pointer. (closes issue + ASTERISK-18836) Reported by: gpluser Review: + https://reviewboard.asterisk.org/r/1628/ ........ Merged + revisions 348362 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348363 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_parkandannounce.c: Fix ParkAndAnnounce to pass the + CallerID to the announcing channel. ParkAndAnnounce tried to pass + the CallerID to the announcing channel but the ID was wiped out + by the channel masquerade done when parking the call. * Save the + CallerID before parking the channel to pass it to the announcing + channel. * Fixed a minor memory leak in ParkAndAnnounce. * + Updated some ParkAndAnnounce log messages. ........ Merged + revisions 348310 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348311 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-14 22:36 +0000 [r348215-348266] Matthew Jordan + + * /, apps/app_originate.c: Added support for all slin formats to + app_originate Previously, app_originate could not originate a + call into a non-8kHz conference bridge as the formats for + non-8kHz slin codecs were not applied to the created channel. + This patch adds all of the formats by default, such that if a + created channel has a codec that supports a higher sampling rate, + a translation path can be built between it and other channels. + ........ Merged revisions 348265 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_queue.c: Fixed Asterisk crash when function + QUEUE_MEMBER receives invalid input The function QUEUE_MEMBER has + two required parameters (queuename, option). It was only checking + for the presence of queuename. The patch checks for the existence + of the option parameter and provides better error logging when + invalid values are provided for the option parameter as well. + ........ Merged revisions 348211 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-14 22:05 +0000 [r348214] Matthew Nicholson + + * /, res/res_fax.c: Don't clear LOCALSTATIONID before sending or + receiving. The user may set that variable. ASTERISK-18921 + ........ Merged revisions 348212 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348213 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-14 21:08 +0000 [r348161] Jonathan Rose + + * main/features.c, configs/features.conf.sample: Add and document + PARKEDCALL variable set during timeout PARKEDCALL variable tracks + which parking lot the call was last parked in. This can be used + afterwards for flow control when returntoorigin is set to off. I + went ahead and documented both this and the existing variable set + during timeout (PARKINGSLOT) in the sample features.conf since + there was no prior mention of variables being set during timeout. + (closes issue ASTERISK-16239) Reported By: Clod Patry Patches: + M17503.diff uploaded by Clod Patry (license 5138) + +2011-12-14 20:51 +0000 [r348160] Matthew Jordan + + * apps/app_confbridge.c: Improve error message in CONFBRIDGE_INFO + Provided a more descriptive error message when a value supplied + for the parameter type is not one of the acceptable values. + (closes issue ASTERISK-18717) Reported by: Paul Belanger Patches: + __20111103-better-confbridge_info-error-msg.txt (License #4999) + +2011-12-14 20:37 +0000 [r348156-348159] Jonathan Rose + + * /, configs/features.conf.sample: Fix accidental use of tabs + instead of spaces from previous features.conf.sample change + ........ Merged revisions 348157 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348158 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, configs/features.conf.sample: Document PARKINGSLOT variable in + features.conf.sample (issue ASTERISK-16239) ........ Merged + revisions 348154 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348155 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-13 23:10 +0000 [r348103] Richard Mudgett + + * /, bridges/bridge_builtin_features.c, apps/app_followme.c: Fix + FollowMe CallerID on outgoing calls. The addition of the + Connected Line support changed how CallerID is passed to outgoing + calls. The FollowMe application was not updated to pass CallerID + to the outgoing calls. * Fix FollowMe CallerID on outgoing calls. + * Restructured findmeexec() to fix several memory leaks and + eliminate some duplicated code. * Made check the return value of + create_followme_number(). Putting a NULL into the numbers list is + bad if create_followme_number() fails. * Fixed a couple uses of + ast_strdupa() inside loops. * The changes to + bridge_builtin_features.c fix a similar CallerID issue with the + bridging API attended and blind transfers. (Not used at this + time.) (closes issue ASTERISK-17557) Reported by: hamlet505a + Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/1612/ ........ Merged + revisions 348101 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348102 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-13 15:22 +0000 [r348061] Stefan Schmidt + + * channels/chan_sip.c: Fix possible misshandling of an incoming SIP + response as a peer poke response. Also make sure peer has even + qualify enabled when handle a peer poke response. (closes issue + ASTERISK-18940) Reported by: Vitaliy Tested by: Vitaliy and + UnixDev Review: https://reviewboard.asterisk.org/r/1620 Reviewed + by: David Vossel ........ Merged revisions 348048 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 348056 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-12 19:35 +0000 [r347997] Matthew Jordan + + * include/asterisk/logger.h, utils/refcounter.c, main/logger.c, + utils/hashtest.c, UPGRADE.txt, utils/ael_main.c, + utils/hashtest2.c, CHANGES, main/asterisk.c, main/config.c, + configs/logger.conf.sample, main/loader.c, main/cli.c: Backed out + core changes from r346391 During testing, it was discovered that + there were a number of side effects introduced by r346391 and + subsequent check-ins related to it (r346429, r346617, and + r346655). This included the /main/stdtime/ test 'hanging', as + well as the remote console option failing to receive the + appropriate output after a period of time. I only backed out the + changes to main/ and utils/, as this was adequate to reverse the + behavior experienced. (issue ASTERISK-18974) + +2011-12-12 17:34 +0000 [r347954] Richard Mudgett + + * configs/iax.conf.sample, configs/chan_dahdi.conf.sample, /, + configs/chan_ooh323.conf.sample, configs/vpb.conf.sample, + configs/extensions.lua.sample, configs/sip.conf.sample, + configs/extensions.conf.sample: Update sample configs to put + incoming calls into context public. * Add warning about the SIP + allowguest option in context public. (closes issue + ASTERISK-14122) Reported by: Alec Davis Review: + https://reviewboard.asterisk.org/r/719/ ........ Merged revisions + 347953 from http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-09 21:47 +0000 [r347866-347903] Jonathan Rose + + * apps/app_mixmonitor.c: Adds MixMonitor and StopMixMonitor AMI + commands to the manager These commands work much like the + dialplan applications that would otherwise invoke them. A nice + benefit of these is that they can be invoked on a call remotely + and at any time during a call. They work much like the Monitor + and StopMonitor ami commands. (closes issue ASTERISK-17726) + Reported by: Sergio González Martín Patches: + mixmonitor_actions.diff uploaded by Sergio González Martín + (license 5644) Review: https://reviewboard.asterisk.org/r/1193/ + + * include/asterisk/file.h, apps/app_sayunixtime.c, CHANGES: Remove + autojump extensions from SayUnixTime, make an option to perform + automatic jumps. When a caller sends DTMF while the SayUnixTime + application is saying the time, The call would jump to the next + extension much like it does during Background(). This patch adds + option 'j' to SayUnixTime which when used employs the old + behavior. Also, this patch allows arguments to sayunixtime to not + be used as empty strings in the case of something like + 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern). (closes issue + ASTERISK-16675) Reported by: jlpedrosa Patches: + patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license + 5959) Review: https://reviewboard.asterisk.org/r/956/ + +2011-12-09 01:33 +0000 [r347813] Richard Mudgett + + * main/pbx.c, /: Fix some parsing issues in + add_exten_to_pattern_tree(). * Simplify compare_char() and avoid + potential sign extension issue. * Fix infinite loop in + add_exten_to_pattern_tree() handling of character set escape + handling. * Added buffer overflow checks in + add_exten_to_pattern_tree() character set collection. * Made + ignore empty character sets. * Added escape character handling to + end-of-range character in character sets. This has a slight + change in behavior if the end-of-range character is an escape + character. You must now escape it. * Fix potential sign extension + issue when expanding character set ranges. * Made remove + duplicated characters from character sets. The duplicate + characters lower extension matching priority and prevent + duplicate extension detection. * Fix escape character handling + when the escape character is trying to escape the end-of-string. + We could have continued processing characters after the end of + the exten string. We could have added the previous character to + the pattern matching tree incorrectly. (closes issue + ASTERISK-18909) Reported by: Luke-Jr ........ Merged revisions + 347811 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 347812 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-08 21:32 +0000 [r347735] Walter Doekes + + * /, channels/chan_sip.c: Fix regression when using tcpenable=no + and tlsenable=yes. The tlsenable settings are tucked away in + main/tcptls.c, so I missed them when resolving ASTERISK-18837. + This should resolve the test suite breakage of the sip tls tests. + Review: https://reviewboard.asterisk.org/r/1615 Reviewed by: Matt + Jordan ........ Merged revisions 347718 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347727 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-08 20:55 +0000 [r347658] Jonathan Rose + + * /, apps/app_queue.c: Fix regressed behavior of queue set penalty + to work without specifying 'in ' r325483 caused a + regression in Asterisk 10+ that would make Asterisk segfault when + attempting to set penalty on an interface without specifying a + queue in the queue set penalty CLI command. In addition, no + attempt would be made whatsoever to perform the penalty setting + on all the queues in the core list with either the cli command or + the non-segfaulting ami equivalent. This patch fixes that and + also makes an attempt to document and rename some functions + required by this command to better represent what they actually + do. Oh yeah, and the use of this command without specifying a + specific queue actually works now. Review: + https://reviewboard.asterisk.org/r/1609/ ........ Merged + revisions 347656 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-08 17:55 +0000 [r347601] Richard Mudgett + + * /, main/features.c: Mark channel running the h exten with the + soft-hangup flag. When a bridge is broken, ast_bridge_call() + might execute the h exten on the calling channel. However, that + channel may not have been the channel that broke the bridge by + hanging up. The channel executing the h exten must be in a hung + up state so things like AGI run in the correct mode. * Make sure + ast_bridge_call() marks the channel it is executing the h exten + on as hung up. (The AST_SOFTHANGUP_APPUNLOAD flag is used so as + to match the pbx.c main dialplan execution loop when it executes + the h exten.) (closes issue ASTERISK-18811) Reported by: David + Hajek Patches: jira_asterisk_18811_v1.8.patch (license #5621) + patch uploaded by rmudgett Tested by: David Hajek, rmudgett + ........ Merged revisions 347595 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347600 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-08 16:24 +0000 [r347533] Terry Wilson + + * /, channels/chan_sip.c: Don't crash on INFO automon request with + no channel AST-2011-014. When automon was enabled in + features.conf, it was possible to crash Asterisk by sending an + INFO request if no channel had been created yet. (closes issue + ASTERISK-18805) ........ Merged revisions 347530 from + http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ + Merged revisions 347531 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347532 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-08 06:59 +0000 [r347490] Damien Wedhorn + + * channels/chan_skinny.c: Fix segfault on answer. Fix a segfault if + an attempt to answer a call is made between when the inbound call + gives up (and the channel is removed) and when the device is + notified and removes the call from the device. + +2011-12-07 21:42 +0000 [r347440] Richard Mudgett + + * main/manager.c, /: Update AMI Getvar and Setvar documentation + about supplying a channel name. (closes issue ASTERISK-18958) + Reported by: Red Patches: jira_asterisk_18958_v1.8.patch (license + #5621) patch uploaded by rmudgett ........ Merged revisions + 347438 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 347439 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-07 20:34 +0000 [r347395] Jonathan Rose + + * /, apps/app_meetme.c: Fix: Meetme recording variables from + realtime DB use null entries over channel variables Meetme would + attempt to substitute the realtime values of RECORDING_FILE and + RECORDING_FORMAT from the meetme db entry instead of using the + channel variable set for those variables in spite of those + database entries being NULL or even lacking a column to represent + them. (closes issue ASTERISK-18873) Reported by: Byron Clark + Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license + 6157) ........ Merged revisions 347369 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347383 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-07 20:15 +0000 [r347345] Terry Wilson + + * Makefile, include/asterisk/paths.h, /, + configs/asterisk.conf.sample, build_tools/make_defaults_h, + main/asterisk.c, main/db.c: Add ASTSBINDIR to the list of + configurable paths This patch also makes astdb2sqlite3 and + astcanary use the configured directory instead of relying on + $PATH. (closes issue ASTERISK-18959) Review: + https://reviewboard.asterisk.org/r/1613/ ........ Merged + revisions 347344 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-06 23:58 +0000 [r347294] Richard Mudgett + + * /, channels/chan_sip.c: Make SIP INFO messages for dtmf-relay + signals case insensitive. (closes issue ASTERISK-18924) Reported + by: Kevin Taylor ........ Merged revisions 347292 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347293 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-06 22:01 +0000 [r347241] Jonathan Rose + + * main/pbx.c, /: Documents CHANNEL(musicclass) taking priority over + m([x]) in waitExten If waitExten specifies a music class to use + with its music on hold option, it will use CHANNEL(musicclass) + instead if that channel variable has been set on the initiating + channel. This documents that behavior in the waitExten app so + that this can be known without checking the documentation of the + code in function local_ast_moh_start. (closes issue + ASTERISK-18804) ........ Merged revisions 347239 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347240 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-06 20:23 +0000 [r347157-347192] Walter Doekes + + * UPGRADE.txt, CHANGES, apps/app_voicemail.c: Add VM_INFO() + dialplan function to gather information about a mailbox. + Deprecates MAILBOX_EXISTS. Provides count, email, exists, + fullname, language, locale, pager, password, tz. (closes issue + ASTERISK-18634) Patch by: Kris Shaw Review: + https://reviewboard.asterisk.org/r/1568 Reviewed by: Walter + Doekes + + * /, channels/chan_sip.c: Don't allow transport=tcp when + tcpenable=no. When tcpenable=no, sending to transport=tcp hosts + was still allowed. Resolving the source address wasn't possible + and yielded the string "(null)" in SIP messages. Fixed that and a + couple of not-so-correct log messages. (closes issue + ASTERISK-18837) Reported by: Andreas Topp Review: + https://reviewboard.asterisk.org/r/1585 Reviewed by: Matt Jordan + ........ Merged revisions 347166 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347167 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_voicemail.c: Add regression tests for issue + ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572 + Reviewed by: Matt Jordan ........ Merged revisions 347131 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347146 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_voicemail.c: The voicemail [general] zonetag and + locale variables weren't loaded until after the mailboxes were + initialized. This caused the settings to be unset for those + mailboxes until a reload was performed. (closes issue + ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570 + Reviewed by: Matt Jordan ........ Merged revisions 347111 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347124 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-06 19:09 +0000 [r347110] Richard Mudgett + + * include/asterisk/dlinkedlists.h, tests/test_linkedlists.c: Doubly + linked lists unit test and update to implementation. Update the + doubly linked list implementation. Now safe traversing can insert + before and after the current node when traversing in either + direction. Updated the linked lists unit test test_linkedlist to + also test doubly linked lists. The old test_dlinkedlist requires + a manual check of results and probably should be removed. Review: + https://reviewboard.asterisk.org/r/1569/ + +2011-12-06 17:34 +0000 [r347069] Matthew Jordan + + * /, channels/chan_sip.c: Fixed crash from orphaned MWI + subscriptions in chan_sip This patch resolves the issue where MWI + subscriptions are orphaned by subsequent SIP SUBSCRIBE messages. + When a peer is removed, either by pruning realtime SIP peers or + by unloading / loading chan_sip, the MWI subscriptions that were + orphaned would still be on the event engine list of valid + subscriptions but have a pointer to a peer that no longer was + valid. When an MWI event would occur, this would cause a seg + fault. (closes issue ASTERISK-18663) Reported by: Ross Beer + Tested by: Ross Beer, Matt Jordan Patches: + blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283) + Review: https://reviewboard.asterisk.org/r/1610/ ........ Merged + revisions 347058 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347068 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-05 17:44 +0000 [r347008] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Restore call progress code for analog + ports. Extracting sig_analog from chan_dahdi lost call progress + detection functionality. * Fix analog ports from considering a + call answered immediately after dialing has completed if the + callprogress option is enabled. (closes issue ASTERISK-18841) + Reported by: Richard Miller Patches: chan_dahdi.diff (license + #5685) patch uploaded by Richard Miller (Modified by me) + sig_analog.c.diff (license #5685) patch uploaded by Richard + Miller (Modified by me) sig_analog.h.diff (license #5685) patch + uploaded by Richard Miller ........ Merged revisions 347006 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 347007 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-05 15:04 +0000 [r346956] Jonathan Rose + + * main/pbx.c, /: Resolve duplicate label used in multiple + priorities for the same extension. Prior to this patch, if labels + with the same name were used for different priorities in the same + extension, the new label would be accepted, but it would be + unusable since attempts to reach that label would just go to the + first one. Now pbx.c detects this, generates a warning in logs, + and culls the label before adding it to the dialplan. (closes + issue ASTERISK-18807) Reported by: Kenneth Shumard Patches: + pbx.c.patch uploaded by Kenneth Shumard (License 5077) ........ + Merged revisions 346954 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 346955 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-05 14:47 +0000 [r346953] Kinsey Moore + + * res/res_jabber.exports.in, /: Fix chan_jingle/gtalk load + regression introduced in r346087 Add missing symbol exports for + ast_aji_client_destroy and ast_aji_buddy_destroy for usage + outside res_jabber. Testing of these changes focused on + res_jabber itself, so this problem was missed. Reported-by: + Michael Spiceland ........ Merged revisions 346951 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 346952 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-04 10:08 +0000 [r346901] Walter Doekes + + * /, channels/chan_sip.c: For SIP REGISTER fix domain-only URIs and + domain ACL bypass. The code that allowed admins to create users + with domain-only uri's had stopped to work in 1.8 because of the + reqresp parser rewrites. This is fixed now: if you have a + [mydomain.com] sip user, you can register with useraddr + sip:mydomain.com. Note that in that case -- if you're using + domain ACLs (a configured domain list) -- mydomain.com must be in + the allow list as well. Reviewboard r1606 shows a list of + registration combinations and which SIP response codes are + returned. Review: https://reviewboard.asterisk.org/r/1533/ + Reviewed by: Terry Wilson (closes issue ASTERISK-18389) (closes + issue ASTERISK-18741) ........ Merged revisions 346899 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 346900 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-02 23:30 +0000 [r346857] Matthew Jordan + + * /, channels/chan_sip.c: Update SIP MESSAGE To parsing to + correctly handle URI The previous patch (r346040) incorrectly + parsed the URI in the presence of a port, e.g., + user@hostname:port would fail as the port would be double + appended to the SIP message. This patch uses the parse_uri + function to correctly parse the URI into its username and + hostname parts, and places them in the correct fields in the + sip_pvt structure. (issue ASTERISK-18903) Review: + https://reviewboard.asterisk.org/r/1597/ ........ Merged + revisions 346856 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-02 19:40 +0000 [r346777-346816] Alexandr Anikin + + * addons/chan_ooh323.c: implement nat option for rtp channels with + ooh323 + + * addons/chan_ooh323.c, /, channels/chan_h323.c: Merged revisions + 346763 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r346763 | may | 2011-12-02 20:42:32 +0400 (Fri, + 02 Dec 2011) | 14 lines Merged revisions 346762 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7 + lines process null frame pointer returned by + ast_rtp_instance_read correctly (closes issue ASTERISK-16697) + Reported by: under Patches: segfault.diff (License #5871) patch + uploaded by under ........ ................ + +2011-12-01 21:19 +0000 [r346709] Richard Mudgett + + * main/stun.c, /, res/res_stun_monitor.c, + configs/res_stun_monitor.conf.sample, include/asterisk/stun.h: + Re-resolve the STUN address if a STUN poll fails for + res_stun_monitor. The STUN socket must remain open between polls + or the external address seen by the STUN server is likely to + change. However, if the STUN request poll fails then the STUN + server address needs to be re-resolved and the STUN socket needs + to be closed and reopened. * Re-resolve the STUN server address + and create a new socket if the STUN request poll fails. * Fix + ast_stun_request() return value consistency. * Fix + ast_stun_request() to check the received packet for expected + message type and transaction ID. * Fix ast_stun_request() to read + packets until timeout or an associated response packet is found. + The stun_purge_socket() hack is no longer required. * Reduce + ast_stun_request() error messages to debug output. * No longer + pass in the destination address to ast_stun_request() if the + socket is already bound or connected to the destination. (closes + issue ASTERISK-18327) Reported by: Wolfram Joost Tested by: + rmudgett Review: https://reviewboard.asterisk.org/r/1595/ + ........ Merged revisions 346700 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 346701 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-12-01 20:46 +0000 [r346699] Jonathan Rose + + * /, channels/chan_sip.c: Change 183 Ringing in sipfrag body to 180 + ringing. 183 Ringing isn't even a thing. 183 is actually a + session progress message. (closes issue ASTERISK-18925) Reported + by: Sebastian Denz Tested by: jrose Patches: + asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian + Denz (License #6139) ........ Merged revisions 346697 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 346698 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-30 23:38 +0000 [r346617-346655] Tilghman Lesher + + * channels/chan_unistim.c, main/tcptls.c, channels/chan_sip.c, + main/config.c, main/loader.c: Remove the few places where we try + to ast_verbose() without a newline. + + * main/asterisk.c: Fix edge case for overflow buffer. + +2011-11-30 22:03 +0000 [r346525-346566] Jonathan Rose + + * main/tcptls.c, /, channels/chan_sip.c, include/asterisk/tcptls.h: + r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | + 18 lines Cleaning up chan_sip/tcptls file descriptor closing. + This patch attempts to eliminate various possible instances of + undefined behavior caused by invoking close/fclose in situations + where fclose may have already been issued on a + tcptls_session_instance and/or closing file descriptors that + don't have a valid index for fd (-1). Thanks for more than a + little help from wdoekes. (closes issue ASTERISK-18700) Reported + by: Erik Wallin (issue ASTERISK-18345) Reported by: Stephane + Cazelas (issue ASTERISK-18342) Reported by: Stephane Chazelas + Review: https://reviewboard.asterisk.org/r/1576/ ........ Merged + revisions 346564 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 346565 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h: + Reverting 346525 due to accidental patch against trunk instead of + 1.8 + + * main/tcptls.c, channels/chan_sip.c, include/asterisk/tcptls.h: + Cleaning up chan_sip/tcptls file descriptor closing. This patch + attempts to eliminate various possible instances of undefined + behavior caused by invoking close/fclose in situations where + fclose may have already been issued on a tcptls_session_instance + and/or closing file descriptors that don't have a valid index for + fd (-1). Thanks for more than a little help from wdoekes. (closes + issue ASTERISK-18700) Reported by: Erik Wallin (issue + ASTERISK-18345) Reported by: Stephane Cazelas (issue + ASTERISK-18342) Reported by: Stephane Chazelas Review: + https://reviewboard.asterisk.org/r/1576/ + +2011-11-30 19:37 +0000 [r346474] Leif Madsen + + * configs/queues.conf.sample: Update queues.conf.sample + documentation. Update the documentation surrounding the use of + MONITOR_EXEC to make it more clear that it can be used for both + Monitor() and MixMonitor() usage. (closes issue ASTERISK-17413) + Reported by: David Woolley Patches: + issue18817_mixmonitor_queues_doc.diff by Michael L. Young + (License #5026) ........ Merged revisions 346472 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 346473 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-29 20:32 +0000 [r346391-346429] Tilghman Lesher + + * utils/refcounter.c, utils/hashtest.c, utils/ael_main.c, + utils/hashtest2.c: Fix compilation of utilities (caught by + Bamboo). + + * addons/chan_ooh323.c, channels/chan_sip.c, main/say.c, + res/res_fax.c, UPGRADE.txt, res/res_musiconhold.c, + res/res_jabber.c, CHANGES, configs/logger.conf.sample, + main/cli.c, channels/chan_usbradio.c, include/asterisk/logger.h, + main/dial.c, channels/chan_skinny.c, main/logger.c, + codecs/codec_dahdi.c, apps/app_rpt.c, apps/app_verbose.c, + main/asterisk.c, main/bridging.c, res/res_clialiases.c, + addons/res_config_mysql.c, apps/app_voicemail.c: Allow each + logging destination and console to have its own notion of the + verbosity level. Review: https://reviewboard.asterisk.org/r/1599 + +2011-11-29 00:03 +0000 [r346350] David Vossel + + * /, include/asterisk/message.h, main/message.c: Merged revisions + 346349 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011) + | 10 lines Fixes memory leak in message API. The ast_msg_get_var + function did not properly decrement the ref count of the var it + retrieves. The way this is implemented is a bit tricky, as we + must decrement the var and then return the var's value. As long + as the documentation for the function is followed, this will not + result in a dangling pointer as the ast_msg structure owns its + own reference to the var while it exists in the var container. + ........ + +2011-11-28 14:34 +0000 [r346294] Stefan Schmidt + + * res/res_rtp_asterisk.c, /: Fix regression that 'rtp/rtcp set + debup ip' only works when also a port was specified. (closes + issue ASTERISK-18693) Reported by: Davide Dal Fra Review: + https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter + Doekes ........ Merged revisions 346292 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 346293 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-23 23:03 +0000 [r346241] Richard Mudgett + + * include/asterisk/acl.h, /, channels/chan_skinny.c, + channels/chan_h323.c, main/acl.c, channels/chan_iax2.c: Fix calls + to ast_get_ip() not initializing the address family. ........ + Merged revisions 346239 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 346240 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-23 20:48 +0000 [r346146-346199] Walter Doekes + + * /, channels/chan_sip.c: Minor cleanup in chan_sip get_msg_text() + function. In r116240, get_msg_text() got an extra parameter to + fix the unwanted addition of trailing newlines to SIP MESSAGE + bodies. This caused all linefeeds to be trimmed, which isn't + right either. This is a stop-gap; the right fix is to return the + original SIP request body. Review: + https://reviewboard.asterisk.org/r/1586 Reviewed by: Matt Jordan + ........ Merged revisions 346147 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 346198 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, include/asterisk/strings.h: Fix ast_str_truncate signedness + warning and documentation. Review: + https://reviewboard.asterisk.org/r/1594 ........ Merged revisions + 346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 346145 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-23 17:16 +0000 [r346088] Kinsey Moore + + * channels/chan_jingle.c, /, include/asterisk/jabber.h, + channels/chan_gtalk.c, res/res_jabber.c: Fix res_jabber resource + leaks This should fix almost all resource leaks in res_jabber + that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous + situation where ast_aji_get_client would sometimes bump an + object's refcount and sometimes not. Review: + https://reviewboard.asterisk.org/r/1553 ........ Merged revisions + 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 346087 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-23 16:23 +0000 [r346053] Matthew Jordan + + * /, channels/chan_sip.c: Fixed SendMessage stripping extension + from To: header in SIP MESSAGE When using the MessageSend + application to send a SIP MESSAGE to a non-peer, chan_sip + attempted to validate the hostname or IP Address. In the process, + it stripped off the extension and failed to add it back to the + sip_pvt structure before transmitting. This patch adds the full + URI passed in from the message core to the sip_pvt structure. + (closes issue ASTERISK-18903) Reported by: Shaun Clark Tested by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/1597/ + ........ Merged revisions 346040 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-23 16:12 +0000 [r346033] Terry Wilson + + * /, res/res_musiconhold.c: Resume playing existing hold music for + cached realtime MOH As a result of the fix for ASTERISK-18039, + realtime caching MOH no longer properly resumes playing back a + file between different holds in the same call. This is because + scanning for new files causes the existing file array to be + emptied and we were just comparing that the saved pointer to the + filename matched the pointer to the filename in a particular + position in the array. An easy fix is to save the filename + instead of a pointer to it and then do a strcmp instead of + comparing the addresses. (closes issue ASTERISK-18912) Review: + https://reviewboard.asterisk.org/r/1596/ ........ Merged + revisions 346030 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 346031 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-23 16:10 +0000 [r346032] Paul Belanger + + * /, res/res_format_attr_silk.c, res/res_format_attr_celt.c: Added + support level for new modules ........ Merged revisions 346029 + from http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-22 23:06 +0000 [r345978] Richard Mudgett + + * main/dnsmgr.c, /, include/asterisk/dnsmgr.h: Fix dnsmgr entries + to ask for the same address family each time. The dnsmgr refresh + would always get the first address found regardless of the + original address family requested. So if you asked for only IPv4 + addresses originally, you might get an IPv6 address on refresh. * + Saved the original address family requested by + ast_dnsmgr_lookup() to be used when the address is refreshed. + ........ Merged revisions 345976 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345977 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-22 20:32 +0000 [r345925] Walter Doekes + + * include/asterisk/logger.h, /: Clarify why the AST_LOG_* macros + exist next to the LOG_* macros. (issue ASTERISK-17973) ........ + Merged revisions 345923 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345924 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-22 16:41 +0000 [r345883] Paul Belanger + + * /, apps/confbridge/conf_config_parser.c: Add missing + sound_only_one config variable (closes issue ASTERISK-18895) + Reported by: zvision Patches: conf_config_parser.diff (license + #5755) patch uploaded by zvision ........ Merged revisions 345882 + from http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-21 21:09 +0000 [r345831] Terry Wilson + + * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Default + to nat=yes; warn when nat in general and peer differ It is + possible to enumerate SIP usernames when the general and + user/peer nat settings differ in whether to respond to the port a + request is sent from or the port listed for responses in the Via + header. In 1.4 and 1.6.2, this would mean if one setting was + nat=yes or nat=route and the other was either nat=no or + nat=never. In 1.8 and 10, this would mean when one was + nat=force_rport and the other was nat=no. In order to address + this problem, it was decided to switch the default behavior to + nat=yes/force_rport as it is the most commonly used option and to + strongly discourage setting nat per-peer/user when at all + possible. For more discussion of the issue, please see: + http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html + (closes issue ASTERISK-18862) Review: + https://reviewboard.asterisk.org/r/1591/ ........ Merged + revisions 345776 from + http://svn.asterisk.org/svn/asterisk/branches/1.4 ........ Merged + revisions 345800 from + http://svn.asterisk.org/svn/asterisk/branches/1.6.2 ........ + Merged revisions 345828 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345830 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-21 16:40 +0000 [r345735] Paul Belanger + + * CHANGES, main/config.c: Add #tryinclude statement This provides + the same functionality as #include however an asterisk module + will still load if the filename does not exist. Review: + https://reviewboard.asterisk.org/r/1476/ + +2011-11-19 15:11 +0000 [r345643-345684] Tilghman Lesher + + * /, main/db.c: Update the documentation to better clarify how the + existing commands work. Review: + https://reviewboard.asterisk.org/r/1593/ ........ Merged + revisions 345682 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345683 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/db.c: Fix a change in behavior in 'database show' from + 1.8. In 1.8 and previous versions, one could use any fullword + portion of the key name, including the full key, to obtain the + record. Until this patch, this did not work for the full key. + Closes issue ASTERISK-18886 Patch: by tilghman Review: by twilson + (http://pastebin.com/7rtu6bpk) on #asterisk-dev ........ Merged + revisions 345640 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-17 19:47 +0000 [r345560-345601] Matthew Jordan + + * contrib/realtime/mysql/sipfriends.sql (removed): Accidentally + readded sipfriends.sql in r345560. This was removed in r342871 + + * configs/confbridge.conf.sample, + apps/confbridge/include/confbridge.h, apps/app_confbridge.c, + CHANGES, contrib/realtime/mysql/sipfriends.sql (added), + apps/confbridge/conf_config_parser.c: Add admin toggle mute all + and participant count menu options to app_confbridge This patch + adds two new menu features to app_confbridge, admin_toggle_menu_ + participants and participant_count. The admin action will + globally mute / unmute all non-admin participants on a + converence, while the participant count simply exposes the + existing participant count function to the conference bridge + menu. This also adds configuration options to change the sound + played when the conference is globally muted / unmuted, as well + as the necessary config hooks to place these functions in the + DTMF menus. (closes issue ASTERISK-18204) Reported by: Kevin + Reeves Tested by: Matt Jordan Patches: + app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt, + confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281) + Review: https://reviewboard.asterisk.org/r/1518/ + +2011-11-17 17:31 +0000 [r345559] Richard Mudgett + + * /, channels/sig_pri.c: Remove dead code since pri_grab() can + never fail. Dead code makes programmers sick. I am sick of + looking at it. ........ Merged revisions 345546 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345558 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-16 14:56 +0000 [r345489] Jonathan Rose + + * /, apps/app_voicemail.c: Guarantee messages go into the right + folders with multiple recipients Before, using the U flag in + Voicemail with multiple recipients would put urgent messages in + the INBOX folder for all users past the first thanks to a bug + with the message copying function. This would also cause messages + to fail to be sent if the INBOX directory hadn't been created for + that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt + Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/1589/ ........ Merged + revisions 345487 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345488 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-15 20:11 +0000 [r345221-345433] Richard Mudgett + + * /, res/res_agi.c: Make FastAGI HANGUP show up in AGI debug + output. * Change from using send() to ast_agi_send() so the + HANGUP shows up in the AGI debug output. (closes issue + ASTERISK-18723) Reported by: James Van Vleet Patches: + jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by + rmudgett ........ Merged revisions 345431 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345432 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/sig_pri.c: Fix typo in sig_pri using wrong structure + name. It is fortunate that the typo does not alter generated code + since the e->restart.channel and e->ring.channel members are in + the same position. (closes issue ASTERISK-18868) Reported by: + zvision Patches: sig_pri.c.diff (License #5755) patch uploaded by + zvision ........ Merged revisions 345370 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345371 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_queue.c: Make queue log indicate if ADDMEMBER is + paused for AMI and realtime. * Add parameter to queue log + ADDMEMBER to indicate if the member is paused. (closes issue + ASTERISK-18645) Reported by: garlew Patches: paused.diff (License + #5337) patch uploaded by garlew Tested by: rmudgett, garlew + Review: https://reviewboard.asterisk.org/r/1469/ ........ Merged + revisions 345285 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345290 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c, configs/sip.conf.sample, UPGRADE-1.8.txt, + channels/sip/include/sip.h: Restore SIP DTMF overlap dialing + method. The recent fix for ASTERISK-17288 to get RFC3578 SIP + overlap support working correctly removed a long standing ability + to do overlap dialing using DTMF in the early media phase of a + call. See ASTERISK-18702 it has a very good description of the + issue. I started with Pavel Troller's chan_sip.diff patch on + issue ASTERISK-18702. * Added 'dtmf' enum value to sip.conf + allowoverlap config option. The new option value causes the + Incomplte application to not send anything with chan_sip so the + caller can supply more digits via DTMF. * Renames + SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE + since that is what it really means. * Fixed get_destination() + inconsistency with the pickup extension matching. * Fixed + initialization of PAGE3 of global_flags in reload_config(). + (closes issue ASTERISK-18702) Reported by: Pavel Troller Review: + https://reviewboard.asterisk.org/r/1517/ Review: + https://reviewboard.asterisk.org/r/1582/ ........ Merged + revisions 345273 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345275 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/pbx.c, /: Fix Progress spelling error in main/pbx.c. (closes + issue ASTERISK-18857) Reported by: David M Patches: + mainpbx-trivial.patch (License #6326) patch uploaded by David M + ........ Merged revisions 345219 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345220 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-14 19:12 +0000 [r345165] Terry Wilson + + * main/channel.c, /: Don't read past end of input when calling + write() int blah = 1; ... write(chan->alertpipe[1], &blah, + new_frames * sizeof(blah)) != (new_frames * sizeof(blah))) is + only valid when new_frames == 1. Otherwise we start reading into + adjacent variables declared on the stack. The read end discards + what is read, so the values don't matter but it's not a good idea + to read past where we want even though new_frames is almost + always 1 and should never be large. This patch is basically taken + out of kpfleming's eventfd branch, as he mentioned that he + remembered fixing it there when I talked to him about this issue. + Review: https://reviewboard.asterisk.org/r/1583/ ........ Merged + revisions 345163 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345164 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-14 19:03 +0000 [r345162] Walter Doekes + + * /, channels/sip/include/reqresp_parser.h: Update reqresp_parser + parse_uri doxygen comments. The issue mentioned in the bug report + had been fixed recently by twilson. The reporter included this + documentation fix. (closes issue ASTERISK-18572) Reported by: + Richard Miller Patch by: Richard Miller (modified) ........ + Merged revisions 345160 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345161 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-14 16:21 +0000 [r345120] Jonathan Rose + + * /, apps/app_voicemail.c: Moves voicemail setup password entry to + the end of the setup process. This change was made because + forcegreeting and forcename settings in voicemail could be + circumvented by hanging up after entering a password, because the + only way voicemail currently observes whether a mailbox is new or + not is by checking to see if the password is the same as the + mailbox number or not. (closes issue ASTERISK-18282) Reported by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/ + ........ Merged revisions 345062 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345117 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-14 15:11 +0000 [r345065] Kinsey Moore + + * /, channels/chan_sip.c: Ensure that a null vmexten does not cause + a segfault When sip_send_mwi_to_peer was modified recently to + avoid deadlocks, vmexten was not expected to be null. This change + handles that situation to avoid a segfault. ........ Merged + revisions 345063 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 345064 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-14 01:25 +0000 [r345023] TransNexus OSP Development + + * apps/app_osplookup.c: Increased max number of destinations. + +2011-11-12 16:32 +0000 [r344979] Gregory Nietsky + + * channels/chan_misdn.c, /: mISDN Round Robin break when no channel + is available Prevent channels been parsed repetitively. ........ + Merged revisions 344965 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344966 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-12 00:36 +0000 [r344901] Terry Wilson + + * /, res/res_musiconhold.c: Don't forget to rescan MOH files for + cached realtime classes Realtime MOH class caching was + implemented because without it, you would build a completely new + MOH class and would start the music over at the beginning each + time hold was pressed in a conversation. Unfortunately, this + broke re-scanning for file changes for realtime MOH classes. This + patch corrects that issue. (closes issue ASTERISK-18039) Review: + https://reviewboard.asterisk.org/r/1579/ ........ Merged + revisions 344899 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344900 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-11 22:00 +0000 [r344846] Walter Doekes + + * include/asterisk/utils.h, /, main/utils.c, + include/asterisk/stringfields.h: Use __alignof__ instead of + sizeof for stringfield length storage. Kevin P Fleming suggested + that r343157 should use __alignof__ instead of sizeof. For most + systems this won't be an issue, but better fix it now while it's + still fresh. Review: https://reviewboard.asterisk.org/r/1573 + ........ Merged revisions 344843 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344845 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-11 21:57 +0000 [r344844] Matthew Jordan + + * /, main/file.c: Video format was treated as audio when removed + from the file playback scheduler This patch fixes the format type + check in ast_closestream and filestream_destructor. Previously a + comparison operator was used, but since audio formats are no + longer contiguous (and AST_FORMAT_AUDIO_MASK includes formats + that have a value greater than the video formats), a bitwise AND + operation is used instead. Duplicated code was also moved to + filestream_close. (closes issue ASTERISK-18682) Reported by: Aldo + Bedrij Tested by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/1580/ ........ Merged + revisions 344823 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344842 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-11 21:37 +0000 [r344838-344840] Walter Doekes + + * /, channels/sip/reqresp_parser.c: Remove unneeded if(params) + checks in reqresp_parser. Nick Lewis added them in + https://reviewboard.asterisk.org/r/549/diff/1-2/ for no apparent + reason. There is no way that params could become NULL in that + piece of code, so I removed these excess checks again. ........ + Merged revisions 344837 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344839 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * main/manager.c, /: Fix bad quoting of multiline mxml opaque_data + that caused invalid xml. The opaque_data was added and enclosed + in single quotes, assuming it would be only a single line. The + rest of the lines were appended after the closing quote. (closes + issue ASTERISK-18852) Reported by: peep_ on IRC Review: + https://reviewboard.asterisk.org/r/1577 ........ Merged revisions + 344835 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 344836 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-11 20:15 +0000 [r344771] Kinsey Moore + + * /, channels/chan_sip.c: Fix regression introduced by SDP fixups + If capability is adjusted when switching to UDPTL during fax + transmission, fax teardown fails. Make sure capability is only + touched if RTP is active. This regression was introduced in + R344385. ........ Merged revisions 344769 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344770 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-11 18:37 +0000 [r344663-344717] Richard Mudgett + + * /, channels/chan_sip.c: Check sip.conf maxforwards parameter for + range 1 <= x <= 255. JIRA AST-710 ........ Merged revisions + 344715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 344716 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, main/cli.c: Make CLI "core show channel" not hold the channel + lock during console output. Holding the channel lock while the + CLI "core show channel" command is executing can slow down the + system. It could block the system if the console output is halted + or paused. * Made capture the CLI "core show channel" output into + a buffer to be output after the channel is unlocked. * Removed + use of C++ keyword as a variable name. out renamed to obuf. * + Checked allocation of obuf for failure so will not crash. (closes + issue ASTERISK-18571) Reported by: Pavel Troller Tested by: + rmudgett ........ Merged revisions 344661 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344662 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-11 15:47 +0000 [r344610] Jonathan Rose + + * main/pbx.c, /: Fix a segmentation fault when using an extension + with CID matching and no CID. Attempting to call an extension + which used Caller ID matching with a channel that has an empty + caller id string would result in a segmentation fault. (closes + issue ASTERISK-18392 Reported By: Ales Zelenik ........ Merged + revisions 344608 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344609 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-10 23:21 +0000 [r344538-344560] Richard Mudgett + + * /, apps/app_macro.c: Fix app_macro.c MODULEINFO section + termination. (closes issue ASTERISK-18848) Reported by: Tony + Mountifield ........ Merged revisions 344557 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_queue.c: Fix potential deadlock calling ast_call() + with channel locks held. Fixed app_queue.c:ring_entry() calling + ast_call() with the channel locks held. Chan_local attempts to do + deadlock avoidance in its ast_call() callback and could deadlock + if a channel lock is already held. ........ Merged revisions + 344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 344540 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_queue.c: Make AMI event AgentCalled get + CallerID/ConnectedLine info from the incoming channel. It was + strange that the AgentCalled AMI event would get most of its + information from the incoming channel but then get the CallerID + information from the outgoing channel. Before connected line + support was added, this information was always the same at this + point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham + Tested by: rmudgett ........ Merged revisions 344536 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344537 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-10 21:56 +0000 [r344494] David Vossel + + * /, main/bridging.c: Merged revisions 344493 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r344493 | dvossel | 2011-11-10 15:54:42 -0600 (Thu, 10 Nov 2011) + | 12 lines Fixes issue with ConfBridge participants hanging up + during DTMF feature menu usage getting stuck in conference + forever. When a conference user enters the DTMF menu they are + suspended from the bridge while the channel is handed off to the + DTMF feature code. If a user entered this state and hungup, there + existed a race condition where the channel could not exit the + conference because it was waiting on a signal that would never + arrive. This patch fixes that, because it would stupid for me to + talk about the problem and commit a patch for something else. + (closes issue ASTERISK-18829) Reported by: zvision ........ + +2011-11-10 21:15 +0000 [r344387-344441] Kinsey Moore + + * /, apps/app_meetme.c: Fix another incorrect case with meetme's + PIN logic and add documentation This fixes an issue where a user + of a dynamic conference was asked for a PIN twice. This also adds + documentation to assist in future modifications to the piece of + code responsible for PIN checking. (closes issue AST-670) + ........ Merged revisions 344439 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344440 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Fix several + bugs with SDP parsing and well-formedness of responses Fix bug + ASTERISK-16558 which dealt with the order of responses to + incoming streams defined by SDP. Fix unreported bug where + offering multiple same-type streams would cause Asterisk to reply + with an incorrect SDP response missing one or more streams + without a proper declination. Fix bugs related to a single + non-audio stream being offered with responses requesting codecs + that were not offered in the initial invite along with an + additional audio stream that was not in the initial invite. + Review: https://reviewboard.asterisk.org/r/1516/ ........ Merged + revisions 344385 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344386 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-10 16:29 +0000 [r344335] Matthew Nicholson + + * res/res_rtp_asterisk.c, /: only attempt to do stun handling on + ipv4 or ipv4 mapped to ipv6 addresses Patch by: jkonieczny + (modified) ASTERISK-18490 ........ Merged revisions 344330 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344334 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-09 20:55 +0000 [r344272] Richard Mudgett + + * /, channels/chan_sip.c: Fix deadlock during dialplan reload. + Another deadlock between the conlock/hints and channels/channel + locking orders. * Don't hold the channel and private lock in + sip_new() when calling ast_exists_extension(). (closes issue + ASTERISK-18740) Reported by: Byron Clark Patches: + sip_exists_exten_dlock_3.diff (license #5041) patch uploaded by + Gregory Hinton Nietsky ASTERISK-18740.patch (license #6157) patch + uploaded by Byron Clark Tested by: Byron Clark ........ Merged + revisions 344268 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344271 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-09 20:10 +0000 [r344214-344217] Terry Wilson + + * /, channels/chan_sip.c, channels/sip/reqresp_parser.c, + channels/sip/include/sip.h, + channels/sip/include/reqresp_parser.h: Don't treat a host:port + string as a domain The domain matching code prior to 1.8 used to + manually remove the port from the host:port string when + determining if an incoming request matched the list of domains. + When switching to the new parsing functions, the documentation + implied that the "domain" was being returned by these functions, + when instead it was returning the "hostport" as defined by RFC + 3261. This led to confusion and resulted in 1.8+ rejecting an + incoming request from x.x.x.x:xxxxx when domain=x.x.x.x was set + in sip.conf. This patch renames the "domain" variables in the + parsing functions to "hostport" to more accurately describe what + it is that they are returning and also properly truncates the + resulting hostport strings when dealing with domain matching. + Review: https://reviewboard.asterisk.org/r/1574/ ........ Merged + revisions 344215 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344216 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, tests/test_netsock2.c: Add a unit test for + ast_sockaddr_split_hostport Review: + https://reviewboard.asterisk.org/r/1575/ ........ Merged + revisions 344157 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344175 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-09 19:08 +0000 [r344161] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, /, addons/ooh323c/src/ooh245.c, + addons/ooh323c/src/ooq931.h, addons/ooh323c/src/ootypes.h, + addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c: + Generate response to Status Enquiry message with Status q.931 + message. Some PBXes require this for call status checking (closes + issue ASTERISK-18748) Reported by: Fabrizio Lazzaretti Patches: + ASTERISK-18748-5.patch (License #5415) patch uploaded by may213 + Tested by: Fabrizio Lazzaretti ........ Merged revisions 344158 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 344159 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-09 17:15 +0000 [r344104] Kinsey Moore + + * /, apps/app_meetme.c: Fix pin parameter behavior regression in + MeetMe The last time this code was touched (by me), a subtlety + was missed based on the difference between needing to check a + pin's validity and the need to prompt for a pin. (closes issue + ASTERISK-18488) ........ Merged revisions 344102 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344103 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-09 15:28 +0000 [r344050] Matthew Nicholson + + * /, formats/format_wav.c: don't call ltohl() twice on the same + value ASTERISK-18739 Patch by: pawel (modified) ........ Merged + revisions 344048 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 344049 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-08 22:14 +0000 [r344005] Richard Mudgett + + * /, channels/chan_sip.c: Residual changes for Asterisk v10 branch + from ASTERISK-18747. Residual changes for Asterisk v10 branch + from ASTERISK-18747 after + https://reviewboard.asterisk.org/r/1564/ commit and associated + dialogs callid hash key change fix. * Make check_rtp_timeout() + return CMP_MATCH if need to delete dialog from dialogs_rtpcheck. + This is an optimization to avoid an unneeded lock/unlock and + object search when using ao2_unlink. * Prevent crash in + check_rtp_timeout() if dialog->rtp is NULL. Review: + https://reviewboard.asterisk.org/r/1557/ ........ Merged + revisions 344004 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-08 19:29 +0000 [r343951] Walter Doekes + + * /, pbx/pbx_config.c: Fix crash when dialplan remove include is + called with too few arguments. "dialplan remove include x from y" + crashed when the amount of arguments was less than 6. (closes + issue ASTERISK-18762) Reported by: Andrey Solovyev Tested by: + Andrey Solovyev ........ Merged revisions 343936 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 343944 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-08 18:35 +0000 [r343905] David Vossel + + * /, channels/chan_sip.c: Merged revisions 343900 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r343900 | dvossel | 2011-11-08 12:29:33 -0600 (Tue, 08 Nov 2011) + | 11 lines Fixes regression caused by r343635 There was a missing + unlock for a function return that is only present in Asterisk 10 + and Asterisk Trunk. (closes issue ASTERISK-18839) Reported by: + Michael L. Young Patches: + asterisk-18839-missing-lock-trunk-v2.diff (License #5026) patch + uploaded by Michael L. Young ........ + +2011-11-08 18:02 +0000 [r343853] Richard Mudgett + + * /, channels/chan_sip.c, main/acl.c: Fixed reference to incorrect + variable if unknown host configured crash. * Fixed a LOG_ERROR + message referencing the config variable list v that had + previously been processed and became NULL. * Added error return + value set that was missing in an ast_append_ha() error return + path. (closes issue ASTERISK-18743) Reported by: Michele Patches: + issueA18743-fix_dynamic_exclude_static_bad_host_log.patch + (license #5674) patch uploaded by Walter Doekes Tested by: + Michele ........ Merged revisions 343851 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 343852 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-08 13:23 +0000 [r343790] Leif Madsen + + * /, build_tools/prep_tarball: Fix boo-boo in prep_tarball script. + A hardcoded a branch number was in the prep_tarball which could + not work. Changed it to the variable. ........ Merged revisions + 343789 from http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-07 22:37 +0000 [r343744] Kinsey Moore + + * /, channels/chan_sip.c: Make "sip show settings" CLI command get + RPID flags from the right global page The "Trust RPID" and "Send + RPID" entries in the "sip show settings" CLI command pulled the + flags from the incorrect global flags page. These are now read + from sip global flags page 0. (closes issue AST-711) ........ + Merged revisions 343743 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-07 21:58 +0000 [r343693] Leif Madsen + + * configs/dundi.conf.sample, pbx/pbx_dundi.c, CHANGES: Allow built + in variables to be used with dynamic weights. You can now use the + built in variables , , and within a dynamic weight. For example, + this could be useful when you want to pass requested lookup + number to the SHELL() function which could be used to execute a + script to dynamically set the weight of the result. (Closes issue + ASTERISK-13657) Reported by: Joel Vandal Tested by: Leif Madsen, + Russell Bryant Patches: asterisk-1.6-dundi-varhead.patch uploaded + by Joel Vandal (License #5374) + +2011-11-07 21:44 +0000 [r343692] Matthew Nicholson + + * /, channels/chan_sip.c: respect case changes in peer names on sip + reload ASTERISK-18669 ........ Merged revisions 343690 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 343691 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-07 21:29 +0000 [r343684] Richard Mudgett + + * /, channels/chan_sip.c: Fix __sip_subscribe_mwi_do() incorectly + changing dialogs hash key callid. Changing an object value used + as a container key requires removing the object from the + container and reinserting it. * Created change_callid_pvt() to + call instead of build_callid_pvt(). The change_callid_pvt() will + correctly change the dialog callid so the ao2 conainter can + explicitly unlink it. ........ Merged revisions 343637 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 343677 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-07 20:35 +0000 [r343636] Kinsey Moore + + * /, channels/chan_sip.c: Prevent BLF subscriptions from causing + deadlocks Fix a locking inversion in sip_send_mwi_to_peer that + was causing deadlocks. This function now requires that both the + peer and associated pvt be unlocked before it is called for cases + where peer and peer->mwipvt form a circular reference. (closes + issue ASTERISK-18663) Review: + https://reviewboard.asterisk.org/r/1563/ ........ Merged + revisions 343621 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 343635 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-07 19:58 +0000 [r343581] Walter Doekes + + * main/udptl.c, /, UPGRADE.txt: Correct the default udptl port + range. The udptl port range was defined as 4000-4999 in the + udptl.conf.sample, as 4500-4599 if you didn't have a config and + 4500-4999 if your config was broken. Default is now 4000-4999. + (closes issue ASTERISK-16250) Reviewed by: Tilghman Lesher + Review: https://reviewboard.asterisk.org/r/1565 ........ Merged + revisions 343580 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-07 19:54 +0000 [r343579] Richard Mudgett + + * /, channels/chan_sip.c: Fix deadlock if peer is destroyed while + sending MWI notice. A dialog cannot be destroyed by the + ao2_callback dialog_needdestroy because of a deadlock between the + dialogs container lock and the RWLOCK of the events subscription + list. * Create dialogs_to_destroy container to hold dialogs that + will be destroyed. * Ensure that the event subscription callback + will never happen with an invalid peer pointer by making the + event callback removal the first thing in the peer destructor + callback. NOTE: This particular deadlock will not happen with + Asterisk 10, but some of the changes still apply. (closes issue + ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review: + https://reviewboard.asterisk.org/r/1564/ ........ Merged + revisions 343577 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 343578 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-07 18:42 +0000 [r343534] Matthew Nicholson + + * main/format.c, /: list all of the codecs associated with a + particular format id for CLI command "core show codec" AST-699 + ........ Merged revisions 343533 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-06 09:51 +0000 [r343492] Olle Johansson + + * main/tcptls.c, include/asterisk/tcptls.h: Formatting and doxygen + improvements + +2011-11-04 19:50 +0000 [r343448] Alexandr Anikin + + * addons/ooh323c/src/ooGkClient.c, addons/ooh323c/src/ooTimer.c, + addons/ooh323c/src/dlist.c, /, addons/ooh323c/src/dlist.h, + addons/ooh323c/src/printHandler.c, addons/ooh323c/src/ooq931.c: + Final fix memleaks in GkClient codes, same for Timer codes. + (these memleaks stop development of gk codes, now i can continue) + Fix printHandler 'Unbalanced Structure' issues with locking + printHandler data for single thread. ........ Merged revisions + 343281 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 343445 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-03 20:37 +0000 [r343394] Walter Doekes + + * /, res/res_config_sqlite.c: Fix sqlite config driver segfault and + broken queries The sqlite realtime handler assumed you had a + static config configured as well. The realtime multientry handler + assumed that you weren't using dynamic realtime. (closes issue + ASTERISK-18354) (closes issue ASTERISK-18355) Review: + https://reviewboard.asterisk.org/r/1561 ........ Merged revisions + 343375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 343393 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-03 19:57 +0000 [r343338] Richard Mudgett + + * /, funcs/func_dialgroup.c: Remove invalid flag given to iterator + in func_dialgroup.c ........ Merged revisions 343336 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 343337 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-03 15:40 +0000 [r343222-343278] Terry Wilson + + * /, channels/sip/include/sip.h: Make room for the fax detect flags + The original REGISTERTRYING flag, in addition to being impossible + to check, also encroached on the space for the flag above it. + This patch moves the flags that were below REGISTERTRYING back to + where they were as though we had just removed the REGISTERTRYING + option. ........ Merged revisions 343276 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 343277 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * contrib/realtime/mysql/sippeers.sql, /, channels/chan_sip.c, + channels/sip/include/sip.h: Remove registertrying option in + chan_sip This option is not only useless, but has been broken + since inception since the flag was never copied from the peer + where it is set to the pvt where it was checked. RFC 3261 + specificially states that you should not send a provisional + response to a non-INVITE request, and if we did fix the code so + that it worked, it would cause the same kind of user enumeration + vulnerability that we've discussed with the nat= setting. This + patch removes registertrying option and any code that would have + sent a 100 response to a register. Review: + https://reviewboard.asterisk.org/r/1562/ ........ Merged + revisions 343220 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 343221 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-02 22:46 +0000 [r343163-343219] Walter Doekes + + * /, channels/chan_sip.c: Fix improper warning introduced by + r342927 and more tweaks Changeset r342927 introduced a warning + which was only supposed to be emitted when a found realtime peer + had an empty (or no) name. It turned out that there were some + inconsistencies left. Now found peers with an empty name are + explicitly ignored like before r342927 but better. Reviewed by: + Stefan Schmidts, Terry Wilson Review: + https://reviewboard.asterisk.org/r/1560 ........ Merged revisions + 343181 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 343192 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * include/asterisk/utils.h, /, main/utils.c, + include/asterisk/stringfields.h: Ensure that string field lengths + are properly aligned Integers should always be aligned. For some + platforms (ARM, SPARC) this is more important than for others. + This changeset ensures that the string field string lengths are + aligned on *all* platforms, not just on the SPARC for which there + was a workaround. It also fixes that the length integer can be + resized to 32 bits without problems if needed. (closes issue + ASTERISK-17310) Reported by: radael, S Adrian Reviewed by: + Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review: + https://reviewboard.asterisk.org/r/1549 ........ Merged revisions + 343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 343158 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-02 19:33 +0000 [r343049-343104] Leif Madsen + + * apps/app_authenticate.c: Add note about how Authenticate() + application with option 'd' works. (closes issue ASTERISK-17422) + Reported by: Leif Madsen ........ Merged revisions 343102 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 343103 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * configs/queues.conf.sample: Update documentation for leastrecent + strategy. In queues.conf.sample the leastrecent strategy was + incorrectly described. Now updated to reflect how the strategy + actually checks peers. (closes issue ASTERISK-17854) Reported by: + Sebastian Denz Patches: queues.conf-doc_issue.patch (License + #6139) ........ Merged revisions 343047 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 343048 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-02 13:46 +0000 [r342992] Kevin P. Fleming + + * /, apps/app_meetme.c: Modify comments in MeetMe application + documentation about DAHDI. The MeetMe application documentation + has some comments about usage of DAHDI, and they were a bit + outdated relative to modern DAHDI releases. This patch changes + the comment to just tell the user that a functional DAHDI timing + source is required, and no longer mention 'dahdi_dummy', since + that module does not exist in current DAHDI releases. ........ + Merged revisions 342990 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342991 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-11-01 21:02 +0000 [r342871-342930] Walter Doekes + + * /, channels/chan_sip.c, configs/extconfig.conf.sample, + include/asterisk/config.h, main/config.c: Several fixes to the + chan_sip dynamic realtime peer/user lookup There were several + problems with the dynamic realtime peer/user lookup code. The + lookup logic had become rather hard to read due to lots of + incremental changes to the realtime_peer function. And, during + the addition of the sipregs functionality, several possibilities + for memory leaks had been introduced. The insecure=port matching + has always been broken for anyone using the sipregs family. And, + related, the broken implementation forced those using sipregs to + *still* have an ipaddr column on their sippeers table. Thanks + Terry Wilson for comprehensive testing and finding and fixing + unexpected behaviour from the multientry realtime call which + caused the realtime_peer to have a completely unused code path. + This changeset fixes the leaks, the lookup inconsistenties and + that you won't need an ipaddr column on your sippeers table + anymore (when you're using sipregs). Beware that when you're + using sipregs, peers with insecure=port will now start matching! + (closes issue ASTERISK-17792) (closes issue ASTERISK-18356) + Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry + Wilson Review: https://reviewboard.asterisk.org/r/1395 ........ + Merged revisions 342927 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342929 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * contrib/realtime/mysql/sippeers.sql (added), + configs/res_config_mysql.conf.sample, /, + configs/extconfig.conf.sample, configs/res_ldap.conf.sample, + res/res_realtime.c, UPGRADE-1.8.txt, configs/dbsep.conf.sample, + main/config.c, contrib/realtime/mysql/sipfriends.sql (removed): + Cleanup references to sipusers and sipfriends dynamic realtime + families Somewhere between 1.4 and 1.8 the sipusers family has + become completely unused. Before that, the sipfriends family had + been obsoleted in favor of separate sipusers and sippeers + families. Apparently, they have been merged back again into a + single family which is now called "sippeers". Reviewed by: + irroot, oej, pabelanger Review: + https://reviewboard.asterisk.org/r/1523 ........ Merged revisions + 342869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ........ Merged revisions 342870 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-31 17:51 +0000 [r342825] Richard Mudgett + + * main/format.c, /, main/format_cap.c: Misc format capability + fixes. * Fixed typo in format_cap.c:joint_copy_helper() using the + wrong variable. * Fix potential race between checking if an + interface exists and adding it to the container in + format.c:ast_format_attr_reg_interface(). * Fixed double rwlock + destroy in format.c:ast_format_attr_init() error exit path. * + Simplified format.c:find_interface() and + format.c:has_interface(). ........ Merged revisions 342824 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-31 16:10 +0000 [r342771] Matthew Jordan + + * main/pbx.c, /, channels/chan_iax2.c: Fixed invalid memory access + when adding extension to pattern match tree When an extension is + removed from a context, its entry in the pattern match tree is + not deleted. Instead, the extension is marked as deleted. When an + extension is removed and re-added, if that extension is also a + prefix of another extension, several log messages would report an + error and did not check whether or not the extension was deleted + before accessing the memory. Additionally, if the extension was + already in the tree but previously deleted, and the pattern was + at the end of a match, the findonly flag was not honored and the + extension would be erroneously undeleted. Additionaly, it was + discovered that an IAX2 peer could be unregistered via the CLI, + while at the same time it could be scheduled for unregistration + by Asterisk. The unregistration method now checks to see if the + peer was already unregistered before continuing with an + unregistration. (closes issue ASTERISK-18135) Reported by: Jaco + Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan + Review: https://reviewboard.asterisk.org/r/1526 ........ Merged + revisions 342769 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342770 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-30 02:31 +0000 [r342716] Terry Wilson + + * /, res/res_calendar.c: Don't crash on empty notify channel + ........ Merged revisions 342715 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-29 04:41 +0000 [r342663-342664] Richard Mudgett + + * include/asterisk/linkedlists.h: Whitespace and some better macro + variable names. * Renamed AST_LIST_TRAVERSE_SAFE_BEGIN __new_prev + to __list_current. * Renamed AST_LIST_MOVE_CURRENT __list_cur to + __extracted. + + * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c: Fix + AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable. + AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an + iteration or before AST_LIST_REMOVE_CURRENT() without corrupting + the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the + list if AST_LIST_INSERT_BEFORE_CURRENT() or + AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed + cut and paste error using the wrong variable in + AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests + for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and + AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 342662 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-27 20:11 +0000 [r342606] Matthew Nicholson + + * /, main/dsp.c: tweak the v21 detector to detect an additional + pattern of hits and misses ........ Merged revisions 342605 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-27 19:48 +0000 [r342557-342604] Jonathan Rose + + * res/res_rtp_multicast.c, /: Fix sequence number overflow over 16 + bits causing codec change in RTP packets. Sequence number was + handled as an unsigned integer (usually 32 bits I think, more + depending on the architecture) and was put into the rtp packet + which is basically just a bunch of bits using an or operation. + Sequence number only has 16 bits allocated to it in an RTP packet + anyway, so it would add to the next field which just happened to + be the codec. This makes sure the sequence number is set to be a + 16 bit integer regardless of architecture (hopefully) and also + makes it so the incrementing of the sequence number does bitwise + or at the peak of a 16 bit number so that the value will be set + back to 0 when going beyond 65535 anyway. (closes issue + ASTERISK-18291) Reported by: Will Schick Review: + https://reviewboard.asterisk.org/r/1542/ ........ Merged + revisions 342602 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342603 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, res/res_jabber.c: Cleanup reference leaks in res_jabber + res_jabber.c had a number of places where astobjs would be + referenced and have their reference counts bumped without having + a dereference made before the object lost scope. This patch adds + a number of ASTOBJ_UNREFs to resolve that. Review: + https://reviewboard.asterisk.org/r/1478/ ........ Merged + revisions 342545 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342546 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-25 22:06 +0000 [r342486-342489] Richard Mudgett + + * /, main/astobj2.c: Check fopen return value for ao2 reference + debug output. Reported by: wdoekes Patched by: wdoekes Review: + https://reviewboard.asterisk.org/r/1539/ ........ Merged + revisions 342487 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342488 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, channels/sig_pri.c: Change D-channel warning to be less + confusing on non-NFAS setups. The "No D-channels available! Using + Primary channel as D-channel anyway!" WARNING message has been + confusing on non-NFAS setups. The message refers to things that + are NFAS specific. * Changed the warning to several different + warnings to be more accurate for the situation and less confusing + as a result: "No D-channels up! Switching selected D-channel from + X to Y.", "No D-channels up!", and "D-channel is down!". ........ + Merged revisions 342484 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342485 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-25 21:11 +0000 [r342382-342437] Terry Wilson + + * /, apps/app_queue.c: Use int for storing ao2_container_count + instad of size_t AST-676 ........ Merged revisions 342435 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342436 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_queue.c: Simplify queue membercount code Despite an + ominous sounding comment stating that membercount was for "logged + in" members only and thus we couldn't use ao2_container_count(), + I could not find a single place in the code where that seemed to + be accurate. The only time we decremented membercount was when we + were marking something dead or actually removing it. The only + places we incremented it were either after ao2_link(), or trying + to correct for having set it to 0 during a reload. In every case + where we were correcting the value, it seemed that we were trying + to make the count actually match what ao2_container_count() would + return. The only place I could find where we made a determination + about something being "logged in" or not, we didn't trust the + membercount, but instead looked at devicestate, paused, etc. This + patch removes membercount, replaces its use with + ao2_container_count, and manually adds the results of + ao2_container_count to a "membercount" field for ast_data queue + query results. This patch also would fix AST-676, but as it is + slightly riskier than the previously committed fix, the two + commits have been made separately. Reivew: + https://reviewboard.asterisk.org/r/1541/ ........ Merged + revisions 342383 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342384 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_queue.c: Properly update membercount for reloaded + members Since q->membercount is set to 0 before reloading, it is + important to increment it again for reloaded members as well as + added. (closes issue AST-676) Review: + https://reviewboard.asterisk.org/r/1541/ ........ Merged + revisions 342380 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342381 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-25 19:09 +0000 [r342278-342330] Kinsey Moore + + * pbx/pbx_spool.c, /: Fix compilation on Snow Leopard/FreeBSD for + pbx_spool.c One of the changes in the recent spool handling of + hardlinks patch was just outside a HAVE_INOTIFY block and caused + compilation to fail in some build environments. This has been + corrected. ........ Merged revisions 342328 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342329 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * pbx/pbx_spool.c, /: Merged revisions 342277 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r342277 | kmoore | 2011-10-25 11:08:04 -0500 + (Tue, 25 Oct 2011) | 25 lines Merged revisions 342276 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r342276 | kmoore | 2011-10-25 11:06:57 -0500 (Tue, 25 Oct 2011) | + 18 lines Fix spool handling to allow call files to be hardlinked + into place This fixes the inotify code to handle call files being + hardlinked into the spool directory. The smsq utility does this, + instead of rename(), to ensure that it cannot accidentally + overwrite an existing spool file. A rename() might do that, but + link() will definitely not. The inotify code had broken this, + because it would wait for an IN_CLOSE_WRITE event on the file... + which was never forthcoming, since it was never opened. Now we + look for IN_OPEN events following the IN_CREATE event, and only + wait for an IN_CLOSE_WRITE if the file was actually opened. + Patch-by: dwmw2 (closes issue ASTERISK-18331) Review: + https://reviewboard.asterisk.org/r/1391/ ........ + ................ + +2011-10-25 01:29 +0000 [r342225] Terry Wilson + + * /, include/asterisk/config.h, main/config.c: Return NULL when no + results returned for realtime_multientry It was not documented + what the return value should be when no entries were returned + with the multientry realtime callback. This change forces + consistent behavior even if the backends return an empty + ast_config. Review: https://reviewboard.asterisk.org/r/1521/ + ........ Merged revisions 342223 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 342224 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-24 22:37 +0000 [r342184] Richard Mudgett + + * /, include/asterisk/astobj2.h: Fix ao2obj.h comment typos and add + missing link/unlink nolock debug defines. ........ Merged + revisions 342183 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-24 22:09 +0000 [r342148] Jonathan Rose + + * main/features.c: Fixes a segfault caused by referencing null + frames introduced in r338623 + +2011-10-24 21:01 +0000 [r342112] Richard Mudgett + + * apps/app_queue.c: Fix use of OBJ_KEY in Queue application. To use + the new OBJ_KEY flag, the container hash and compare callback + functions must be updated to support OBJ_KEY. Otherwise, bad + things happen. (issue ASTERISK-14769) + +2011-10-24 20:01 +0000 [r342063] Jonathan Rose + + * /, channels/chan_sip.c: Outbound SIP OPTIONS messages will now + include fromuser of related peer. This behavior matches up more + closely with the way invite/register/etc are handled. This patch + also modifies some adjacent code for code style compliance. + Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy + Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded + by Jeremy Kister (license #6232) ........ Merged revisions 342061 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + Merged revisions 342062 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-24 07:40 +0000 [r341923-342018] Gregory Nietsky + + * /, apps/app_queue.c: queues container needs locking when using + the OBJ_NOLOCK flag ........ Merged revisions 342017 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_queue.c: Remove some ref leaks and a return without + unlock. There some resource leaks introduced in asterisk 10 make + sure that locks are not held on return and we release ref's held. + ........ Merged revisions 341972 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_queue.c: Whitespace Fixups / Add Braces This janitorial + patch is related to work on RB1538 + +2011-10-22 12:03 +0000 [r341869] Alexandr Anikin + + * addons/chan_ooh323.c, /: Merged revisions 341313 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r341313 | may | 2011-10-19 03:33:49 +0400 (Wed, + 19 Oct 2011) | 10 lines Merged revisions 341312 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r341312 | may | 2011-10-19 03:20:53 +0400 (Wed, 19 Oct 2011) | 3 + lines fix issue on channel numbering (calls could have same + channel number on heavy loaded system) ........ ................ + +2011-10-21 16:42 +0000 [r341808-341811] Matthew Nicholson + + * /, pbx/pbx_lua.c: only process args that exist ASTERISK-18395 + ........ Merged revisions 341809 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341810 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, pbx/pbx_lua.c: don't limit the length of app and function + arguments ASTERISK-18395 ........ Merged revisions 341806 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341807 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-21 09:16 +0000 [r341769] Gregory Nietsky + + * res/res_fax.c: White space fixes in res_fax + +2011-10-20 22:03 +0000 [r341719] Richard Mudgett + + * /, main/features.c, res/res_agi.c, include/asterisk/features.h: + Fix AGI exec Park to honor the Park application parameters. The + fix for ASTERISK-12715 and ASTERISK-12685 added a check for the + Park application because the channel needed to be masqueraded to + prevent a crash. Since the Park application now always + masquerades the channel into the parking lot, the special check + is no longer needed. The fix also resulted in AGI exec Park + attempting to double park the call and not honor the Park + application parameters. * Removed no longer necessary call to + ast_masq_park_call() by AGI exec for the Park application. + (Reverts -r146923) * Fix Park application to only return 0 or -1. + The AGI exec Park was causing broken pipe error messages because + the Park application returned 1 on successful park. (closes issue + ASTERISK-18737) ........ Merged revisions 341717 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341718 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-20 21:28 +0000 [r341666-341713] Paul Belanger + + * /, funcs/func_callerid.c: Fixed typo from previous commit + ........ Merged revisions 341704 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341707 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, funcs/func_callerid.c: Updated documentation for the optional + CID parameter with CALLERID ........ Merged revisions 341664 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341665 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-20 18:27 +0000 [r341583-341624] Gregory Nietsky + + * /, configs/queues.conf.sample: Merged revisions 341599 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ........ r341599 | irroot | 2011-10-20 20:20:08 +0200 (Thu, 20 + Oct 2011) | 8 lines add documentation for check_state_unknown in + configs/queues.conf.sample app_queue allows calls to members in a + "Unknown" state to be treated as available setting + check_state_unknown = yes will cause app_queue to query the + channel driver to better determine the state this only applies to + queues with ringinuse or ignorebusy set appropriately. ........ + + * /, CHANGES, apps/app_queue.c: Merged revisions 341580 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ........ r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20 + Oct 2011) | 15 lines Add option to check state when state is + unknown r341486 reverts r325483 this is a rework of the patch. + optimize to minimize load. add option check_state_unknown to + control whether a member with unknown device state is checked + there is a small % chance that calls will be sent to the member + when they on a call. app_queue will see a device with unknown + state as available and does not try verify the state without this + option enabled. Review: https://reviewboard.asterisk.org/r/1535/ + ........ + +2011-10-20 15:17 +0000 [r341533] Terry Wilson + + * /, include/asterisk/strings.h: Clean up ast_check_digits The code + was originally copied from the is_int() function in the AEL code. + wdoekes pointed out that the function should take a const char* + and that their was an unneeded variable. This is now fixed. + ........ Merged revisions 341529 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341530 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-19 21:24 +0000 [r341487] Matthew Nicholson + + * /, apps/app_queue.c: Merged revisions 341486 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r341486 | mnicholson | 2011-10-19 16:23:17 -0500 (Wed, 19 Oct + 2011) | 18 lines Fix a performance regression introduced in + r325483. The regression was caused by a call to + ast_parse_device_state() in app_queue's ring_entry() function. + The ast_parse_device_state() function eventually calls + ast_channel_get_full() with a channel name prefix which causes it + to walk the channel list causing massive lock contention and slow + downs. This patch fixes the regression by removing the call to + ast_parase_device_state() which should be unnecessary. Queue + member device state should be maintained by device state events. + Some users have seen instances where busy agents were called when + they shouldn't have, which is the reason the call to + ast_parse_device_state() was added. That change appears to have + resolved that issue but also causes this performance regression. + There may still be issues with queue member status, and if so, + alternative methods should be investigated to resolve them. + AST-695 ........ + +2011-10-19 19:02 +0000 [r341437] Paul Belanger + + * /, channels/chan_gtalk.c: Outgoing calls with Google Voice Google + has recently make some changes (again) to their protocol. Rather + then patching asterisk to flip between the two different methods, + we now allow both. Lets hope this keeps Google Voice happy for a + while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov + Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses + 6311) ........ Merged revisions 341435 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341436 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-19 07:45 +0000 [r341381] Terry Wilson + + * /, channels/chan_sip.c, include/asterisk/strings.h: Don't use + is_int() since it doesn't link well on all platforms Just create + an normal API function in strings.h that does the same thing just + to be safe. ASTERISK-17146 ........ Merged revisions 341379 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341380 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-19 07:27 +0000 [r341378] Stefan Schmidt + + * /, channels/chan_sip.c: Don't sent in-dialog requests like UPDATE + when Asterisk has not yet received a Contact URI from a UAS + ........ Merged revisions 341366 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341377 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-18 23:45 +0000 [r341316] Terry Wilson + + * /, channels/chan_sip.c: Don't resolve numeric hosts or contact + unresolved hosts If a SIP dial string contains a numeric hostname + that is not a peer name, don't try to resolve it as it is + unlikely that someone really means Dial(SIP/0.0.4.26) when + Dial(SIP/1050) is called. Also, make sure that create_addr + returns -1 if an address isn't resolved so that we don't attempt + to send SIP requests to an address that doesn't resolve. (closes + issue ASTERISK-17146, ASTERISK-17716) Review: + https://reviewboard.asterisk.org/r/1532/ ........ Merged + revisions 341314 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341315 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-18 21:15 +0000 [r341256] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_sip.c, main/features.c, channels/chan_iax2.c, + channels/sip/include/sip.h, channels/chan_mgcp.c, + include/asterisk/features.h: More parking issues. * Fix potential + deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, + IAX, DAHDI analog, and MGCP channel drivers to respect the + parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) + parameter). Created ast_park_call_exten() and + ast_masq_park_call_exten() to maintian API compatibility. * Made + masq_park_call() handle a failed ast_channel_masquerade() setup. + * Reduced excessive struct parkeduser.peername[] size. ........ + Merged revisions 341254 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341255 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-17 17:58 +0000 [r341198] Tzafrir Cohen + + * /, pbx/pbx_realtime.c: Remove an unused include of md5.h Unused + include of asterisk/md5.h in pbx_realtime.c . A commit needed to + test the commit message. Merged-From: + http://svn.asterisk.org/svn/asterisk/branches/1.8@341074 + Merged-From: + http://svn.asterisk.org/svn/asterisk/branches/10@341148 + +2011-10-17 17:38 +0000 [r341191] Terry Wilson + + * /, channels/chan_sip.c: Initialize variables before calling + parse_uri If parse_uri was called with an empty URI, some + pointers would be modified and an invalid read could result. This + patch avoids calling parse_uri with an empty contact uri when + parsing REGISTER requests. AST-2011-012 (closes issue + ASTERISK-18668) ........ Merged revisions 341189 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341190 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-17 16:39 +0000 [r341126-341147] Paul Belanger + + * /, tests/test_format_api.c: Set 'core' support level for + test_format_api.c ........ Merged revisions 341146 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_voicemail.c: Multiple revisions 341108,341112 + ........ r341108 | pabelanger | 2011-10-17 12:22:19 -0400 (Mon, + 17 Oct 2011) | 2 lines Voicemail compiler flags are 'core' + support ........ r341112 | pabelanger | 2011-10-17 12:23:33 -0400 + (Mon, 17 Oct 2011) | 2 lines Fix previous commit ........ Merged + revisions 341108,341112 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341122 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-17 16:18 +0000 [r341096] Jason Parker + + * /, CHANGES: Add information about limitations of new codec + support in channel drivers. (issue ASTERISK-18680) ........ + Merged revisions 341094 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-17 15:45 +0000 [r341090] Terry Wilson + + * /, channels/chan_sip.c: Don't try to remove peers without IPs + from peers_by_ip (closes issue ASTERISK-18696) ........ Merged + revisions 341088 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341089 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-14 21:37 +0000 [r341024] Kevin P. Fleming + + * /, build_tools/embed_modules.xml, Makefile.moddir_rules: Change + the internal name of the menuselect options that are used to + control whether modules are embedded or not; using just the bare + category name led to accidentally enabling these options when + users used the wrong "--enable" operation on the menuselect + command line. Now the internal option names are prefixed with + "EMBED_", so they won't be the same as the name of the category + containing the modules they control the embedding of. ........ + Merged revisions 341022 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 341023 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-14 21:15 +0000 [r340973] Damien Wedhorn + + * channels/chan_skinny.c: Fix simple switch to not progress a call + when call already progressed. If a simple switch was started on a + device and then a specific call made (such as redial or speed + dial), on timeout of the simple switch the call would be + attempted again. This patch only allows the simple switch to make + a call if the substate is still in the collecting digits mode. + Also added small debug message to dialAndAactivate sub. Tested by + snuff and myself. + +2011-10-14 20:51 +0000 [r340972] Kinsey Moore + + * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions + 340971 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r340971 | kmoore | 2011-10-14 15:50:37 -0500 + (Fri, 14 Oct 2011) | 15 lines Merged revisions 340970 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r340970 | kmoore | 2011-10-14 15:49:39 -0500 (Fri, 14 Oct 2011) | + 8 lines Quiet RTCP Receiver Reports during fax transmission RTCP + is now disabled for "inactive" RTP audio streams during SIP T.38 + sessions. The ability to disable RTCP streams in res_rtp_asterisk + was missing, so this code was added to support the bug fix. + (closes issue ASTERISK-18400) ........ ................ + +2011-10-14 18:38 +0000 [r340932] Jonathan Rose + + * utils/utils.xml, /, funcs/func_jitterbuffer.c: Some additional + module documentation changes for 10 for the menuselect change. + (issue ASTERISK-18268) ........ Merged revisions 340931 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-14 16:45 +0000 [r340880] Terry Wilson + + * main/channel.c, /: Avoid unnecessary WARNING message Add + AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid + displaying a WARNING message. (closes issue ASTERISK-18610) Patch + by: Kristijan_Vrban ........ Merged revisions 340878 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 340879 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-13 23:08 +0000 [r340811-340813] Richard Mudgett + + * /, main/features.c: Fix DTMF blind transfer continuing to execute + dialplan after transfer. Party A calls Party B. Party A DTMF + blind transfers Party B to Party C. Party A channel continues to + execute dialplan. * Fixed the return value of + builtin_blindtransfer() to return the correct value after a + transfer so the dialplan will not keep executing. * Removed + unnecessary connected line update that did not really do + anything. * Made access to GOTO_ON_BLINDXFR thread safe in + check_goto_on_transfer(). * Fixed leak of xferchan for failure + cases in check_goto_on_transfer(). * Updated debug messages in + builtin_blindtransfer() and check_goto_on_transfer(). (closes + issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett + ........ Merged revisions 340809 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 340810 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /: Update 10 merged property. + + * /: Restore branch 10 merge properties. + +2011-10-13 08:53 +0000 [r340771] Gregory Nietsky + + * /: Merged revisions 339463 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) | + 9 lines Only change the capabilities on the gateway when the + session is been destroyed there is still a race condition that + ends in a segfault. if the caps are changed the logic in + res_fax_spandsp will run T30 code not gateway code to end the + session. this has been experienced on a "slower" under spec + system. ........ + +2011-10-13 07:05 +0000 [r340720] Stefan Schmidt + + * channels/chan_sip.c: Merged revisions 340718 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r340718 | schmidts | 2011-10-13 06:59:50 +0000 + (Thu, 13 Oct 2011) | 9 lines Merged revisions 340717 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r340717 | schmidts | 2011-10-13 06:58:00 +0000 (Thu, 13 + Oct 2011) | 3 lines storing the route-set also on a 181 response + not only on 180,182 or 183. ........ ................ + +2011-10-13 07:02 +0000 [r340665-340719] Terry Wilson + + * /, channels/chan_sip.c: Initialize ast_sockaddr before calling + ast_sockaddr_resolve Avoid possible jump based on unitialized + value ........ Merged revisions 340715 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 340716 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, res/res_config_sqlite.c: Don't skip the query field on a + realtime multi query There is no documented reason to not add the + query field to the varlist returned by a realtime multi query, + despite the config category being set to its value. Of course, + there is no documentation that the category should be set to the + value either. There is lots of no documentation when it comes to + realtime. But, other engines do not skip this field so I am + forcing this backend to follow the convention, because not doing + so is very silly. ........ Merged revisions 340662 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 340663 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-12 21:28 +0000 [r340626] Stefan Schmidt + + * channels/chan_sip.c: Merged revisions 340577 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r340577 | schmidts | 2011-10-12 20:33:37 +0000 + (Mit, 12 Okt 2011) | 9 lines Merged revisions 340576 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r340576 | schmidts | 2011-10-12 20:30:37 +0000 (Mit, 12 + Okt 2011) | 3 lines Store route-set from provisional SIP + responses so early-dialog requests can be routed properly + ........ ................ + +2011-10-12 21:02 +0000 [r340579] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 340578 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r340578 | twilson | 2011-10-12 13:57:19 -0700 + (Wed, 12 Oct 2011) | 16 lines Merged revisions 340534 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r340534 | twilson | 2011-10-12 13:19:36 -0700 (Wed, 12 Oct 2011) + | 9 lines Update SIP realtime fullcontact regardless of caching + We should update the fullcontact field in the realtime table + whether or not rtcachefriends is set. There is no reason to treat + a non-cached realtime entity differently than a cached in this + regard. (closes issue ASTERISK-18446) Reported by: wdoekes + ........ ................ + +2011-10-12 20:09 +0000 [r340472-340524] Richard Mudgett + + * channels/chan_dahdi.c, /: Initialize the PRI channel alarms + properly on startup. The PRI channel alarms were initialized with + an inverted sense. (closes issue ASTERISK-18710) Reported by: + Tzafrir Cohen ........ Merged revisions 340522 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 340523 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_meetme.c: Update MeetMe p and X option documentation + when interacting with the s option. ASTERISK-12175 changed the p + and X options to not interfere with the s option when they are + used together. It makes more sense for the s option to have + priority for the DTMF '*' key since it cannot change its + activation code. Otherwise, you could not use option s with the p + or X options. JIRA AST-671 ........ Merged revisions 340470 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 340471 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-12 16:29 +0000 [r340420] Paul Belanger + + * /, channels/chan_sip.c: Fix verbose messages when IPv6 logic was + added (closes issue ASTERISK-18612) Reported by: Tim Osman + ........ Merged revisions 340418 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 340419 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-11 21:06 +0000 [r340318-340367] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_ss7.h, /, channels/sig_ss7.c: + Add protection for SS7 channel allocation and better glare + handling. * Added a CLI "ss7 show channels" command that might + prove useful for future debugging. * Made the incoming SS7 + channel event check and gripe message uniform. * Made sure that + the DNID string for an incoming call is always initialized. + (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven + Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621) + patch uploaded by rmudgett ........ Merged revisions 340365 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 340366 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * channels/sip/include/dialog.h, /, channels/chan_sip.c: Fix some + potential deadlocks pointed out by helgrind. * Fixed deadlock + potential calling dialog_unlink_all() in __sip_autodestruct(). + Found by helgrind. * Fixed deadlock potential in + handle_request_invite() after calling sip_new(). Found by + helgrind. * The sip_new() function now returns with the created + channel already locked. * Removed the dead code that starts a PBX + in in sip_new(). No sip_new() callers caused that code to be + executed and it was a bad thing to do anyway. * Removed unused + parameters and return value from dialog_unlink_all(). * Made + dialog_unlink_all() and __sip_autodestruct() safely obtain the + owner and private channel locks without a deadlock avoidance + loop. ........ Merged revisions 340284 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 340310 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-11 19:06 +0000 [r340283] Tzafrir Cohen + + * main/channel.c, /, main/sha1.c, include/asterisk/sha1.h: Update + SHA1 code to RFC 6234 RFC 6234 is an update to RFC 3174 from + which the code was originally taken. It has a slightly better + code, and a better phrased license (simple 3-clause BSD). * + main/sha1.c is sha1.c from RFC 6234 with formatting changes only. + * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC + 6234. * Removed unused include of asterisk/sha1.h from + main/channels.c Review: https://reviewboard.asterisk.org/r/1503/ + Merge-From: + http://svn.asterisk.org/svn/asterisk/branches/1.8@340263 + Merge-From: + http://svn.asterisk.org/svn/asterisk/branches/10@340280 + +2011-10-11 18:57 +0000 [r340282] Richard Mudgett + + * main/manager.c, /, include/asterisk/manager.h: Convert registered + AMI actions to ao2 objects. * Fixed race between calling an AMI + action callback and unregistering that action. Refixes + ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential + memory leak if an AMI action failed to get registered because is + already was registered. Part of the ao2 conversion. * Fixed AMI + ListCommands action not walking the actions list with a lock + held. * Fix usage of ast_strdupa() and alloca() in loops. Excess + stack usage. * Fix AMI Originate action Variable header requiring + a space after the header colon. Reported by Yaroslav Panych on + the asterisk-dev list. * Increased the number of listed variables + allowed per AMI Originate action Variable header to 64. * Fixed + AMI GetConfigJSON action output format. * Fixed usage of res + contents outside of scope in append_channel_vars(). * Fixed + inconsistency of config file channelvars option. The values no + longer accumulate with every channelvars option in the config + file. Only the last value is kept to be consistent with the CLI + "manager show settings" command. (closes issue ASTERISK-18479) + Reported by: Jaco Kroon ........ Merged revisions 340279 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 340281 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-10-10 23:10 +0000 [r340221-340224] Terry Wilson + + * UPGRADE.txt, main/db.c: Return error when no rows are deleted for + AMI DBDelTree (closes issue AST-654) + + * /, main/db.c: Merged revisions 340222 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r340222 | twilson | 2011-10-10 15:55:39 -0700 (Mon, 10 Oct 2011) + | 8 lines On astdb conversion, also warn about permissions + requirements The user running Asterisk must have permission to + the directory the Asterisk database resides in since SQLite 3 + needs to be able to create a journal file. (closes issue + ASTERISK-18174) ........ + + * utils/Makefile, utils/utils.xml, /, UPGRADE.txt, + utils/astdb2bdb.c (added): Merged revisions 340219-340220 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ........ r340219 | twilson | 2011-10-10 15:38:06 -0700 (Mon, 10 + Oct 2011) | 8 lines Add astdb conversion utility for Berkeley to + SQLite 3 If someone wants to backtrack from Asterisk 1.8 to 10 + they can use the astdb2bdb utility to convert the database back + to the Berkeley format that Asterisk 1.8 uses. Review: + https://reviewboard.asterisk.org/r/1502/ ........ r340220 | + twilson | 2011-10-10 15:39:41 -0700 (Mon, 10 Oct 2011) | 2 lines + Add a missing file for the astdb2bdb conversion utility ........ + +2011-10-10 20:39 +0000 [r340166] Matthew Jordan + + * /, channels/chan_sip.c: Merged revisions 340165 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r340165 | mjordan | 2011-10-10 15:30:18 -0500 + (Mon, 10 Oct 2011) | 20 lines Merged revisions 340164 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r340164 | mjordan | 2011-10-10 15:23:48 -0500 (Mon, 10 Oct 2011) + | 13 lines Updated chan_sip to place calls on hold if SDP address + in INVITE is ANY This patch fixes the case where an INVITE is + received with c=0.0.0.0 or ::. In this case, the call should be + placed on hold. Previously, we checked for the address being + null; this patch keeps that behavior but also checks for the ANY + IP addresses. Review: https://reviewboard.asterisk.org/r/1504/ + (closes issue ASTERISK-18086) Reported by: James Bottomley Tested + by: Matt Jordan ........ ................ + +2011-10-10 14:16 +0000 [r340110] Matthew Nicholson + + * main/pbx.c, main/manager.c, /, res/res_fax.c, apps/app_fax.c, + include/asterisk/module.h, res/res_agi.c, + include/asterisk/xmldoc.h, doc/appdocsxml.dtd, main/loader.c, + main/xmldoc.c: Merged revisions 340109 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r340109 | mnicholson | 2011-10-10 09:15:41 -0500 + (Mon, 10 Oct 2011) | 18 lines Merged revisions 340108 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r340108 | mnicholson | 2011-10-10 09:14:48 -0500 (Mon, 10 Oct + 2011) | 11 lines Load the proper XML documentation when multiple + modules document the same application. This patch adds an + optional "module" attribute to the XML documentation spec that + allows the documentation processor to match apps with identical + names from different modules to their documentation. This patch + also fixes a number of bugs with the documentation processor and + should make it a little more efficient. Support for multiple + languages has also been properly implemented. ASTERISK-18130 + Review: https://reviewboard.asterisk.org/r/1485/ ........ + ................ + +2011-10-10 00:57 +0000 [r339993-340071] Damien Wedhorn + + * channels/chan_skinny.c: Add skinny version 17 protocol support. + Added some data to skinny packet structures to make compatible + with v17. Added protocolversion to device, set on registration + based on the version provided by device. v17 includes some + increased ip space for ip6. This patch increases ip space in the + packets but still only uses ip4. Some packet structures + duplicated (ip4 and ip6 types). ip4 type used unless version is + greater or equal to 17. Tested by snuff and myself on 7961 with + recent 8.5 firmware. Also tested compatible with old 7960 and + older 30VIPs. + + * channels/chan_skinny.c: Increase SKINNY_MAX_PACKET and add some + logging. Increase SKINNY_MAX_PACKET to 2000 bytes to handle some + messages in v17 that are greater than the old 1000 bytes. Also + add some useful logging regarding packet and session handling. A + device (with protocol v17) was sending a packet with length + greater than 1000 which resulted in the TCP session being + destroyed and registration being retryed. + + * /, channels/chan_skinny.c: Merged revisions 340031 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r340031 | wedhorn | 2011-10-10 09:18:27 +1100 (Mon, 10 Oct 2011) + | 8 lines Return -1 to skinny_session if register rejected. If + device registration is rejected, return -1 so that the session is + destroyed immediately. Previously, a segfault would occur on a + graceful shutdown if a register is rejected and the + skinny_session has not yet timed out. ........ + + * /, channels/chan_skinny.c: Merged revisions 339992 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r339992 | wedhorn | 2011-10-10 08:09:12 +1100 (Mon, 10 Oct 2011) + | 9 lines Remove log message on traverse session list. On + destroying a session, a list of sessions is traversed to find the + matching session. For each session not matching, skinny + erroneously logged that the session was not matched. While + technically correct the message was misleading, and tended to + indicate errors that were not there. ........ + +2011-10-09 01:19 +0000 [r339832-339947] Igor Goncharovskiy + + * channels/chan_unistim.c, /: Merged revisions 339942 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339942 | igorg | 2011-10-09 08:18:02 +0700 + (Вск, 09 Окт 2011) | 12 lines Merged revisions 339938 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) | + 6 lines Fix compilation issue, caused by missed session structure + (closes issue ASTERISK-18694) Reported by: alex70 ........ + ................ + + * channels/chan_unistim.c, /: Merged revisions 339885 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339885 | igorg | 2011-10-08 22:46:27 +0700 + (Сбт, 08 Окт 2011) | 13 lines Merged revisions 339884 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) | + 7 lines Fix segfault in Unistim channel (closes issue + ASTERISK-18638) Reported by: jonnt ........ ................ + + * channels/chan_unistim.c, /: Merged revisions 339831 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339831 | igorg | 2011-10-08 22:01:35 +0700 + (Сбт, 08 Окт 2011) | 14 lines Merged revisions 339830 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) | + 8 lines Fix char array cast as short array in send_client() + function (for ARM platform) (closes issue ASTERISK-17314) + Reported by: jjoshua ........ ................ + +2011-10-07 19:37 +0000 [r339721-339778] Richard Mudgett + + * /, apps/app_url.c: Merged revisions 339777 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339777 | rmudgett | 2011-10-07 14:36:24 -0500 + (Fri, 07 Oct 2011) | 12 lines Merged revisions 339776 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339776 | rmudgett | 2011-10-07 14:34:55 -0500 (Fri, 07 Oct 2011) + | 5 lines Initialize option flags for SendURL application. + (closes issue ASTERISK-18574) Reported by: marcelloceschia + ........ ................ + + * /: Recorded merge of revisions 339681 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r339681 | wedhorn | 2011-10-06 15:47:08 -0500 (Thu, 06 Oct 2011) + | 10 lines Fixed segfault on core stop gracefully. There was an + issue that the cap and confcap pointers for each line and device + were being memcpy'd so they all pointed to the same + ast_format_cap. On destroying, a segfault occured on the second + call to the same struct. skinny reload now works again as well. + Tested by snuff (in trunk) and myself. ........ + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + autoconf/ast_ext_lib.m4: Merged revisions 339720 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339720 | rmudgett | 2011-10-06 17:58:40 -0500 + (Thu, 06 Oct 2011) | 27 lines Merged revisions 339719 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) + | 20 lines Fix regression in configure script for libpri + capability checks. JIRA AST-598 added the PRI_L2_PERSISTENCE + option to fix BRI PTMP TE layer 2 persistence issues with some + telcos. ASTERISK-18535 attempted to fix the unexpected + requirement that libpri *must* have that feature to work with + Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI + optional features required. Unfortunately, I thought + AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri + and deleted those lines for libpri. The result was the + HAVE_PRI_xxx defines that control the ability to use optional + libpri features were also deleted. * Created + AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional + features in a library that the source code could take advantage + of if the code supports the feature. (closes issue + ASTERISK-18687) Reported by: Norbert Tested by: rmudgett ........ + ................ + +2011-10-06 20:18 +0000 [r339680] Damien Wedhorn + + * channels/chan_skinny.c: Fixed segfault on core stop gracefully. + There was an issue that the cap and confcap pointers for each + line and device were being memcpy'd so they all pointed to the + same ast_format_cap. On destroying, a segfault occured on the + second call to the same struct. skinny reload now works again as + well. Tested by snuff and myself. + +2011-10-06 17:54 +0000 [r339627] Richard Mudgett + + * main/udptl.c, /, channels/chan_sip.c: Merged revisions 339626 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339626 | rmudgett | 2011-10-06 12:53:00 -0500 + (Thu, 06 Oct 2011) | 25 lines Merged revisions 339625 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339625 | rmudgett | 2011-10-06 12:49:38 -0500 (Thu, 06 Oct 2011) + | 18 lines Fix debugging messages generated by 'udptl debug'. * + Makes chan_sip set the tag to the channel name. * Fixes received + debug message sequence number. * Removed tx/rx debug message type + since it was hard coded to 0. * Made udptl.c logged message + header consistent if possible: "UDPTL (%s): ". * Removed unused + rx_expected_seq_no from struct ast_udptl. (closes issue + ASTERISK-18401) Reported by: Kevin P. Fleming Patches: + jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by + rmudgett Tested by: Matthew Nicholson ........ ................ + +2011-10-06 13:43 +0000 [r339587] Leif Madsen + + * build_tools/prep_tarball: Merged revisions 339586 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339586 | lmadsen | 2011-10-06 08:43:21 -0500 + (Thu, 06 Oct 2011) | 16 lines Merged revisions 339566 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339566 | lmadsen | 2011-10-05 16:30:11 -0500 (Wed, 05 Oct 2011) + | 8 lines Update prep_tarball script to download pre-exported + documentation. I've updated the prep_tarball script to now + download the pre-exported documentation from the Asterisk wiki. + This will give us more control over what is being included in the + tarball releases, and will make both the PDF and HTML exported + documentation look much better (especially when viewing from a + console). (Closes issue ASTERISK-18677) ........ ................ + +2011-10-05 17:02 +0000 [r339510-339513] Richard Mudgett + + * apps/app_dial.c, /: Merged revisions 339512 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339512 | rmudgett | 2011-10-05 12:01:46 -0500 + (Wed, 05 Oct 2011) | 9 lines Merged revisions 339511 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r339511 | rmudgett | 2011-10-05 12:01:01 -0500 (Wed, 05 + Oct 2011) | 1 line Fix Dial F option notes formatting. ........ + ................ + + * main/manager.c, /: Merged revisions 339508 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339508 | rmudgett | 2011-10-05 11:35:02 -0500 + (Wed, 05 Oct 2011) | 18 lines Merged revisions 339504,339506 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339504 | rmudgett | 2011-10-05 11:26:45 -0500 (Wed, 05 Oct 2011) + | 7 lines Add missing documentation of required AMI action + Challenge AuthType header. (closes issue ASTERISK-18554) Reported + by: Vlad Povorozniuc Patches: + __20110919-manager-challenge-docs.patch.txt (license #4999) patch + uploaded by Leif Madsen ........ r339506 | rmudgett | 2011-10-05 + 11:32:03 -0500 (Wed, 05 Oct 2011) | 1 line Fix XML error in AMI + action Challenge. ........ ................ + +2011-10-05 16:35 +0000 [r339509] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 339507 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339507 | mnicholson | 2011-10-05 11:32:59 -0500 + (Wed, 05 Oct 2011) | 10 lines Merged revisions 339505 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339505 | mnicholson | 2011-10-05 11:31:21 -0500 (Wed, 05 Oct + 2011) | 3 lines The app name in the documentation must match what + we register the application as. ........ ................ + +2011-10-05 06:50 +0000 [r339464-339465] Gregory Nietsky + + * res/res_fax.c, include/asterisk/res_fax.h, CHANGES: Add generic + faxdetect framehook to res_fax Added func + FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no to enable dialplan + faxdetect allowing more flexibility. as soon as a fax tone is + detected the framehook is removed. there is a penalty involved in + running this framehook on non G711 channels as they will be + transcoded. CNG tone is suppresed using the SQUELCH flag to allow + WaitForNoise to be run on the channel to detect Voice. (Closes + issue ASTERISK-18569) Reported by: Myself Reviewed by: Matthew + Nicholson, Kevin Fleming Review: + https://reviewboard.asterisk.org/r/1116/ + + * /, res/res_fax.c: Merged revisions 339463 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r339463 | irroot | 2011-10-05 08:28:46 +0200 (Wed, 05 Oct 2011) | + 9 lines Only change the capabilities on the gateway when the + session is been destroyed there is still a race condition that + ends in a segfault. if the caps are changed the logic in + res_fax_spandsp will run T30 code not gateway code to end the + session. this has been experienced on a "slower" under spec + system. ........ + +2011-10-04 22:59 +0000 [r339408] Richard Mudgett + + * Makefile, /: Merged revisions 339407 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339407 | rmudgett | 2011-10-04 17:56:25 -0500 + (Tue, 04 Oct 2011) | 15 lines Merged revisions 339406 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339406 | rmudgett | 2011-10-04 17:54:15 -0500 (Tue, 04 Oct 2011) + | 8 lines Make always create the MOH directory + (/var/lib/asterisk/moh). (closes issue ASTERISK-18409) Reported + by: abelbeck Patches: asterisk-1.8-makefile-moh.patch (license + #5903) patch uploaded by abelbeck Tested by: abelbeck, Michael + Keuter ........ ................ + +2011-10-04 19:51 +0000 [r339315-339354] Jonathan Rose + + * /, main/say.c: Merged revisions 339353 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339353 | jrose | 2011-10-04 14:44:02 -0500 + (Tue, 04 Oct 2011) | 18 lines Merged revisions 339352 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339352 | jrose | 2011-10-04 14:33:12 -0500 (Tue, 04 Oct 2011) | + 12 lines Removes improper use of sound 'and' in German language + mode from application saynumber Asterisk would say 'Five hundert + und sechs und zwanzig' instead of 'Five hundert sechs und + zwanzig'... which is both weird sounding and wrong. This patch + makes sure Asterisk will only say the 'and' word between the + single digit and double digit places. (closes issue + ASTERISK-18212) Reported By: Lionel Elie Mamane Patches: + upstream_germand_no_and.diff (License #5402) uploaded by Lionel + Elie Mamane ........ ................ + + * /, res/res_jabber.c: Merged revisions 339298 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339298 | jrose | 2011-10-04 09:09:50 -0500 + (Tue, 04 Oct 2011) | 19 lines Merged revisions 339297 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339297 | jrose | 2011-10-04 09:01:05 -0500 (Tue, 04 Oct 2011) | + 13 lines Reverting revision 333265 due to component connection + problems it introduces. I'm going to attempt some generic + res_jabber cleanup and come up with a new fix for this problem, + but first it seems prudent to remove this rather broad attempt to + fix it and instead approach this problem either from the same + angle but looking only at canceling (or possibly rescheduling) + the send when we absolutely know it will cause a segfault or, if + that can't be easily accomplished, strictly from the devstate + side of things. Also, I'm pretty sure a lot of the code in + res_jabber isn't thread safe. (issue ASTERISK-18626) (issue + ASTERISK-18078) ........ ................ + +2011-10-04 12:27 +0000 [r339262] Alexandr Anikin + + * /, addons/ooh323c/src/memheap.c: Merged revisions 339245 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339245 | may | 2011-10-04 15:49:49 +0400 (Tue, + 04 Oct 2011) | 9 lines Merged revisions 339244 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339244 | may | 2011-10-04 15:44:55 +0400 (Tue, 04 Oct 2011) | 2 + lines fix forget declaration in previous change ........ + ................ + +2011-10-04 09:43 +0000 [r339206] Olle Johansson + + * main/manager.c, CHANGES: Generate error message when AMI action + originate extension doesn't exist Review: + https://reviewboard.asterisk.org/r/1445/ Is this a bug or a new + feature? No responses on Asterisk-dev so I'm committing to trunk + only. + +2011-10-03 20:13 +0000 [r339146-339149] Leif Madsen + + * channels/chan_sip.c: Merged revisions 339148 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339148 | lmadsen | 2011-10-03 15:13:16 -0500 + (Mon, 03 Oct 2011) | 14 lines Merged revisions 339147 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339147 | lmadsen | 2011-10-03 15:12:43 -0500 (Mon, 03 Oct 2011) + | 6 lines Remove duplicated Maxforwards line in AMI output. + (Closes issue ASTERISK-18637) Reported by: Jacek Konieczny + Patches: asterisk-sipshowpeer.patch (License #6298) uploaded by + Jacek Konieczny ........ ................ + + * apps/app_dial.c: Merged revisions 339145 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339145 | lmadsen | 2011-10-03 14:55:15 -0500 + (Mon, 03 Oct 2011) | 13 lines Merged revisions 339144 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339144 | lmadsen | 2011-10-03 14:54:52 -0500 (Mon, 03 Oct 2011) + | 6 lines Make documentation for Dial() options 'F' and 'F()' + more clear. (Closes issue ASTERISK-18646) Reported by: Physis + Heckman Tested by: Richard Mudgett ........ ................ + +2011-10-03 19:16 +0000 [r339091] Alexandr Anikin + + * /, addons/ooh323c/src/memheap.c: Merged revisions 339089 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339089 | may | 2011-10-03 22:52:55 +0400 (Mon, + 03 Oct 2011) | 10 lines Merged revisions 339087 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339087 | may | 2011-10-03 22:42:49 +0400 (Mon, 03 Oct 2011) | 4 + lines destroy memheap mutex properly before memheap deleted (fix + memory leak occured after r304950 changes with DEBUG_THREAD + compile option) ........ ................ + +2011-10-03 18:58 +0000 [r339090] Terry Wilson + + * /, channels/chan_sip.c, main/file.c: Merged revisions 339088 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r339088 | twilson | 2011-10-03 11:44:27 -0700 + (Mon, 03 Oct 2011) | 17 lines Merged revisions 339086 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r339086 | twilson | 2011-10-03 11:40:52 -0700 (Mon, 03 Oct 2011) + | 10 lines Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more + places After the change in r336294, the new + AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite + happens. If we receive a re-invite from a device the + waitstream_core was not aware of the new control frame and would + drop the call. (closes issue ASTERISK-18610) Reported by: + Kristijan_Vrban ........ ................ + +2011-10-03 15:55 +0000 [r339021-339046] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 339045 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r339045 | mnicholson | 2011-10-03 10:54:55 -0500 (Mon, 03 Oct + 2011) | 4 lines Ported ast_fax_caps_to_str() to 10, not sure why + it wasn't already here. This function prints a list of caps + instead of a hex bitfield. ........ + + * /, res/res_fax.c: Merged revisions 339043 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r339043 | mnicholson | 2011-10-03 10:41:36 -0500 (Mon, 03 Oct + 2011) | 2 lines Don't clear the AST_FAX_TECH_MULTI_DOC flag right + after we set it. ........ + + * /, res/res_fax.c: Merged revisions 339011 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r339011 | mnicholson | 2011-10-03 10:19:44 -0500 (Mon, 03 Oct + 2011) | 2 lines properly remove the AST_FAX_TECH_GATEWAY flag + (instead of setting all of the other flags) ........ + +2011-10-03 14:40 +0000 [r338905-338998] Gregory Nietsky + + * /, CHANGES: Merged revisions 338997 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r338997 | irroot | 2011-10-03 16:38:25 +0200 (Mon, 03 Oct 2011) | + 1 line Documentation noting the extension of CHANNEL() for + chan_ooh323 ........ + + * addons/chan_ooh323.c, /, funcs/func_channel.c: Merged revisions + 338995 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r338995 | irroot | 2011-10-03 16:21:40 +0200 (Mon, 03 Oct 2011) | + 6 lines Remove the channel function OOH323() and place its + options into CHANNEL() channel drivers should not have there own + dialplan functions. ........ + + * /, res/res_fax.c: Merged revisions 338950 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r338950 | irroot | 2011-10-03 11:37:59 +0200 (Mon, 03 Oct 2011) | + 14 lines Fixup a race condition in res_fax.c where + FAXOPT(gateway)=no will turn off the gateway but the framehook is + not destroyed. this problem happens when a gateway is attempted + in the dialplan and the device is not available i may want to do + fax to mail in the server it will not be allowed. instead of + checking only AST_FAX_TECH_GATEWAY also check gateway_id Reverts + 338904 Fix some white space. ........ + + * /, res/res_fax.c: Merged revisions 338904 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r338904 | irroot | 2011-10-02 16:17:32 +0200 (Sun, 02 Oct 2011) | + 8 lines Remove T38 Gateway capability when detaching framehook. + SET(FAXOPT(gateway)=no) does not remove the capability when + detaching the framehook. small patch to fix this problem. + ........ + +2011-10-01 01:56 +0000 [r338855] TransNexus OSP Development + + * configure: Update "configure" based on r338139. + +2011-09-30 22:08 +0000 [r338802] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 338801 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r338801 | rmudgett | 2011-09-30 17:06:48 -0500 + (Fri, 30 Sep 2011) | 19 lines Merged revisions 338800 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r338800 | rmudgett | 2011-09-30 17:05:10 -0500 (Fri, 30 Sep 2011) + | 12 lines Fix segfault in analog_ss_thread() not checking + ast_read() for NULL. NOTE: The problem was reported against + v1.6.2. It is unlikely to ever happen on v1.8 and above since + chan_dahdi.c:analog_ss_thread() is unlikely to be used. The + version in sig_analog.c has largely replaced it. (closes issue + ASTERISK-18648) Reported by: Stephan Bosch Patches: + jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by + rmudgett Tested by: Stephan Bosch ........ ................ + +2011-09-30 19:25 +0000 [r338755] Olle Johansson + + * channels/chan_sip.c: Formatting changes only --Denna och + nedanstående rader kommer inte med i loggmeddelandet-- M + channels/chan_sip.c + +2011-09-30 18:59 +0000 [r338720] Jonathan Rose + + * /, configs/queues.conf.sample: Merged revisions 338719 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r338719 | jrose | 2011-09-30 13:55:27 -0500 + (Fri, 30 Sep 2011) | 9 lines Merged revisions 338718 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r338718 | jrose | 2011-09-30 13:54:30 -0500 (Fri, 30 Sep + 2011) | 1 line Adds documentation for QueueMemberStatus event + generation ........ ................ + +2011-09-30 16:40 +0000 [r338665] Richard Mudgett + + * /, channels/chan_sip.c: Fix formatting of AMI header for SIP show + peer. ASTERISK-17486 exposed the problem for AMI parsers. (closes + issue ASTERISK-18649) Reported by: Jacek Konieczny Patches: + asterisk-sipshowpeer_response_end.patch (license #6298) patch + uploaded by Jacek Konieczny ........ Merged revisions 338663 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged + revisions 338664 from + http://svn.asterisk.org/svn/asterisk/branches/10 + +2011-09-30 13:21 +0000 [r338623] Olle Johansson + + * main/features.c: Preserve DTMF length in main/features.c Review: + https://reviewboard.asterisk.org/r/1463/ A small part of much + larger work with DTMF duration in Asterisk, funded by IPvision AS + in Denmark. Thanks to irroot for the review! + +2011-09-29 21:16 +0000 [r338557] Paul Belanger + + * tests/test_security_events.c, /, tests/test_locale.c, + tests/test_logger.c, tests/test_dlinklists.c, + tests/test_linkedlists.c, tests/test_amihooks.c: Merged revisions + 338556 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r338556 | pabelanger | 2011-09-29 17:14:34 -0400 + (Thu, 29 Sep 2011) | 9 lines Merged revisions 338555 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r338555 | pabelanger | 2011-09-29 17:12:21 -0400 (Thu, + 29 Sep 2011) | 2 lines Test modules should depend on the + TEST_FRAMEWORK flag ........ ................ + +2011-09-29 20:55 +0000 [r338553] Jason Parker + + * /, tests/test_db.c, tests/test_netsock2.c: Merged revisions + 338552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r338552 | qwell | 2011-09-29 15:54:55 -0500 + (Thu, 29 Sep 2011) | 9 lines Merged revisions 338551 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r338551 | qwell | 2011-09-29 15:54:13 -0500 (Thu, 29 Sep + 2011) | 1 line Test modules have a support level of core. + ........ ................ + +2011-09-29 12:22 +0000 [r338435] Gregory Nietsky + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged + revisions 338417 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r338417 | irroot | 2011-09-29 14:16:42 +0200 + (Thu, 29 Sep 2011) | 19 lines Merged revisions 338416 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r338416 | irroot | 2011-09-29 14:13:05 +0200 (Thu, 29 Sep 2011) | + 12 lines The rtptimeout setting is ignored on a per peer basis. + Not only is the rtptimeout ignored in some cases but rtpkeepalive + and rtpholdtimeout is affected. this commit also removes + rtptimeout/rtpholdtimeout on text rtp. (closes issue + ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 + ........ ................ + +2011-09-29 12:03 +0000 [r338377-338415] Olle Johansson + + * cdr/cdr_pgsql.c, CHANGES: Add CLI command "cdr show pgsql status" + based on "cdr mysql status" Review: + https://reviewboard.asterisk.org/r/923/ Thanks all for the code + reviews and feedback. + + * res/res_agi.c: Just formatting. + +2011-09-28 22:38 +0000 [r338284-338324] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 338323 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r338323 | rmudgett | 2011-09-28 17:36:57 -0500 + (Wed, 28 Sep 2011) | 12 lines Merged revisions 338322 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r338322 | rmudgett | 2011-09-28 17:35:52 -0500 (Wed, 28 Sep 2011) + | 5 lines Make duplicate call ptr warning message more helpful. * + Adds the value of the call ptr to the duplicate call ptr message + to help trace why there is a duplicate call ptr. ........ + ................ + + * include/asterisk/logger.h, /: Merged revisions 338253 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r338253 | rmudgett | 2011-09-28 16:22:05 -0500 + (Wed, 28 Sep 2011) | 14 lines Merged revisions 338235 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r338235 | rmudgett | 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011) + | 7 lines Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE + declaration. (closes issue ASTERISK-17973) Reported by: Luke H + Patches: logger_h.patch (license #6278) patch uploaded by Luke H + ........ ................ + +2011-09-28 20:55 +0000 [r338229] Jason Parker + + * build_tools/cflags.xml, channels/chan_usbradio.c, + build_tools/cflags-devmode.xml, agi/agi.xml, utils/utils.xml, /, + build_tools/embed_modules.xml, tests/test_db.c, + tests/test_netsock2.c: Merged revisions 338228 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r338228 | qwell | 2011-09-28 15:54:35 -0500 + (Wed, 28 Sep 2011) | 9 lines Merged revisions 338227 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r338227 | qwell | 2011-09-28 15:52:47 -0500 (Wed, 28 Sep + 2011) | 1 line Add support levels to non-module sections of + menuselect (cflags, utils, etc). ........ ................ + +2011-09-28 20:28 +0000 [r338226] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 338225 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r338225 | rmudgett | 2011-09-28 15:26:39 -0500 + (Wed, 28 Sep 2011) | 12 lines Merged revisions 338224 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r338224 | rmudgett | 2011-09-28 15:24:41 -0500 (Wed, 28 Sep 2011) + | 5 lines Fix chan_dahd compiling with gcc 4.6 when PRI and SS7 + not present. (closes issue ASTERISK-18357) Reported by: Matthew + Nicholson ........ ................ + +2011-09-28 17:00 +0000 [r338187-338188] Terry Wilson + + * CHANGES: Update CHANGES to reflect autopausebusy not being in + Asterisk 10 + + * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Add + autopausebusy and autopauseunavail queue options Make it possible + to autopause on a busy or unavailable response from a device. + (closes issue ASTERISK-16112) Reported by: jlpedrosa Patches: + autopausebusy.txt by twilson Review: + https://reviewboard.asterisk.org/r/1399/ + +2011-09-28 07:30 +0000 [r338136-338139] TransNexus OSP Development + + * configure.ac: Updated for checking OSP Toolkit version 4.0.0. + + * apps/app_osplookup.c: Updated for OSP Toolkit 4.0.0. + +2011-09-27 20:15 +0000 [r338086] Paul Belanger + + * /, apps/app_macro.c: Merged revisions 338085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r338085 | pabelanger | 2011-09-27 16:13:14 -0400 + (Tue, 27 Sep 2011) | 9 lines Merged revisions 338084 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r338084 | pabelanger | 2011-09-27 16:10:13 -0400 (Tue, + 27 Sep 2011) | 2 lines Upgrade app_macro to core ........ + ................ + +2011-09-27 12:45 +0000 [r338042] Olle Johansson + + * channels/chan_sip.c: Whitespace (red blobs) fixes + +2011-09-26 19:40 +0000 [r337975] Richard Mudgett + + * apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, /, + include/asterisk/cel.h, cdr/cdr_syslog.c, tests/test_gosub.c, + include/asterisk/channel.h, main/cel.c, main/manager.c, + funcs/func_odbc.c, cel/cel_custom.c, apps/app_minivm.c, + main/logger.c, cel/cel_sqlite3_custom.c, cdr/cdr_custom.c, + cdr/cdr_manager.c, apps/app_voicemail.c: Merged revisions 337974 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r337974 | rmudgett | 2011-09-26 14:35:23 -0500 + (Mon, 26 Sep 2011) | 37 lines Merged revisions 337973 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) + | 30 lines Fix deadlock when using dummy channels. Dummy channels + created by ast_dummy_channel_alloc() should be destoyed by + ast_channel_unref(). Using ast_channel_release() needlessly grabs + the channel container lock and can cause a deadlock as a result. + * Analyzed use of ast_dummy_channel_alloc() and made use + ast_channel_unref() when done with the dummy channel. (Primary + reason for the reported deadlock.) * Made + app_dial.c:dial_exec_full() not call ast_call() holding any + channel locks. Chan_local could not perform deadlock avoidance + correctly. (Potential deadlock exposed by this issue. Secondary + reason for the reported deadlock since the held lock was part of + the deadlock chain.) * Fixed some uses of + ast_dummy_channel_alloc() not checking the returned channel + pointer for failure. * Fixed some potential chan=NULL pointer + usage in func_odbc.c. Protected by testing the bogus_chan value. + * Fixed needlessly clearing a 1024 char auto array when setting + the first char to zero is enough in manager.c:action_getvar(). + (closes issue ASTERISK-18613) Reported by: Thomas Arimont + Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch + uploaded by rmudgett Tested by: Thomas Arimont ........ + ................ + +2011-09-23 19:20 +0000 [r337855-337910] Gregory Nietsky + + * /, contrib/init.d/rc.archlinux.asterisk: Merged revisions 337902 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r337902 | irroot | 2011-09-23 21:18:14 +0200 + (Fri, 23 Sep 2011) | 10 lines Merged revisions 337898 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337898 | irroot | 2011-09-23 21:14:30 +0200 (Fri, 23 Sep 2011) | + 4 lines Spelling fix ........ ................ + + * /, apps/app_queue.c: Merged revisions 337840 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r337840 | irroot | 2011-09-23 10:39:22 +0200 + (Fri, 23 Sep 2011) | 17 lines Merged revisions 337839 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337839 | irroot | 2011-09-23 10:34:03 +0200 (Fri, 23 Sep 2011) | + 11 lines Make sure a CDR is on the stack for call in the Queue. + Only let update_cdr act on the last CDR in the stack. In some + circumstances [Attended transfer to queue] a CDR record is not + inserted for this call where it should. (closes issue + ASTERISK-18567) Review: https://reviewboard.asterisk.org/r/1266 + ........ ................ + +2011-09-23 00:47 +0000 [r337776] Russell Bryant + + * /, configs/res_pktccops.conf.sample: Merged revisions 337775 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r337775 | russell | 2011-09-22 19:45:35 -0500 + (Thu, 22 Sep 2011) | 18 lines Merged revisions 337774 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337774 | russell | 2011-09-22 19:44:19 -0500 (Thu, 22 Sep 2011) + | 11 lines Comment out entries in sample res_pktccops.conf. With + these options enabled, they can cause Asterisk to freak out by + SYN flooding a network and eating the CPU. Obviously it would be + good to fix the code so that this can't happen, but we can at + least change the default configuration so it doesn't happen. This + was reported downstream to the Fedora issue tracker: + https://bugzilla.redhat.com/show_bug.cgi?id=658431 ........ + ................ + +2011-09-22 21:42 +0000 [r337722] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 337721 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r337721 | rmudgett | 2011-09-22 16:37:41 -0500 + (Thu, 22 Sep 2011) | 25 lines Merged revisions 337720 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337720 | rmudgett | 2011-09-22 16:29:46 -0500 (Thu, 22 Sep 2011) + | 18 lines Made ISDN not add numbering plan prefix strings to + empty numbers. When the Caller-ID is restricted, the expected + behavior is for the Caller-ID to be blank. In chan_dahdi, the + national prefix is placed onto the Caller-ID number even if it is + restricted (empty) causing the Caller-ID to be the national + prefix rather than blank. This behavior was lost when sig_pri was + extracted from chan_dahdi. * Made not add prefix strings to empty + connected line, calling, and ANI number strings. (closes issue + ASTERISK-18577) Reported by: Kris Shaw Patches: + jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by + rmudgett Tested by: Kris Shaw ........ ................ + +2011-09-22 16:35 +0000 [r337600] Jonathan Rose + + * /, channels/chan_sip.c, include/asterisk/event_defs.h, + main/security_events.c, channels/sip/security_events.c (added), + main/event.c, CHANGES, channels/sip/include/security_events.h + (added), channels/sip/include/sip.h, + include/asterisk/security_events_defs.h, + configs/logger.conf.sample: Merged revisions 337595,337597 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ........ r337595 | jrose | 2011-09-22 10:35:50 -0500 (Thu, 22 Sep + 2011) | 12 lines Generate Security events in chan_sip using new + Security Events Framework Security Events Framework was added in + 1.8 and support was added for AMI to generate events at that + time. This patch adds support for chan_sip to generate security + events. (closes issue ASTERISK-18264) Reported by: Michael L. + Young Patches: security_events_chan_sip_v4.patch (license #5026) + by Michael L. Young Review: + https://reviewboard.asterisk.org/r/1362/ ........ r337597 | jrose + | 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) | 10 lines Forgot + to svn add new files to r337595 Part of Generating security + events for chan_sip (issue ASTERISK-18264) Reported by: Michael + L. Young Patches: security_events_chan_sip_v4.patch (License + #5026) by Michael L. Young Reviewboard: + https://reviewboard.asterisk.org/r/1362/ ........ + +2011-09-22 11:46 +0000 [r337432-337543] Gregory Nietsky + + * /, res/res_srtp.c: Merged revisions 337542 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r337542 | irroot | 2011-09-22 13:44:22 +0200 + (Thu, 22 Sep 2011) | 14 lines Merged revisions 337541 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337541 | irroot | 2011-09-22 13:39:49 +0200 (Thu, 22 Sep 2011) | + 8 lines Add warned to ast_srtp to prevent errors on each frame + from libsrtp The first 9 frames are not reported as some devices + dont use srtp from first frame these are suppresed. the warning + is then output only once every 100 frames. ........ + ................ + + * /, channels/chan_h323.c: Merged revisions 337487 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r337487 | irroot | 2011-09-22 11:26:26 +0200 + (Thu, 22 Sep 2011) | 16 lines Merged revisions 337486 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337486 | irroot | 2011-09-22 11:22:26 +0200 (Thu, 22 Sep 2011) | + 10 lines If IP address is used in chan_h323 host parameter of + peer configuration. module tries to resolve IP address to IP + address and fails. Simple fix to set family of socket this is a + hangover from ipv6 changes. (closes issue ASTERISK-18237) (issue + ASTERISK-17278) (issue ASTERISK-17500) ........ ................ + + * main/channel.c, /: Merged revisions 337431 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r337431 | irroot | 2011-09-22 08:29:09 +0200 + (Thu, 22 Sep 2011) | 25 lines Merged revisions 337430 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337430 | irroot | 2011-09-22 08:18:33 +0200 (Thu, 22 Sep 2011) | + 19 lines Its possible to loose audio on ast_write when the + channel is not transcoded correctly. in the case of DAHDI the + channel is hungup. This patch tries to "fix" the problem and make + the channel compatiable and warn the user of this problem. Please + note there is a underlying problem with codec negotion this does + not fix the problem it does try to rectify it and prevent loss of + service. Review: https://reviewboard.asterisk.org/r/1442/ (closes + issue ASTERISK-17541) (closes issue ASTERISK-18063) (issue + ASTERISK-14384) (issue ASTERISK-17502) (issue ASTERISK-18325) + (issue ASTERISK-18422) ........ ................ + +2011-09-21 21:26 +0000 [r337343-337385] Tilghman Lesher + + * /, apps/app_voicemail.c: More silly spacing changes ..... Merged + revisions 337353 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ..... Merged + revisions 337380 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /, apps/app_voicemail.c: ................ ........ Dumb little + spacing fix. ........ Merged revisions 337344 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ Merged revisions 337345 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * funcs/func_curl.c, /: ................ ........ Escape commas in + keys and values, when keys and values are enumerated by commas. + Review: https://reviewboard.asterisk.org/r/1433 ........ Merged + revisions 337325 from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ Merged revisions 337342 from + https://origsvn.digium.com/svn/asterisk/branches/10 + +2011-09-21 11:21 +0000 [r337262-337283] Gregory Nietsky + + * /, configs/sip.conf.sample: Merged revisions 337263 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r337263 | irroot | 2011-09-21 13:15:48 +0200 (Wed, 21 Sep 2011) | + 1 line Whitespace fixup from SRTP patch ........ + + * /, apps/app_originate.c, CHANGES: Merged revisions 337261 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ........ r337261 | irroot | 2011-09-21 12:42:06 +0200 (Wed, 21 + Sep 2011) | 10 lines Adds a timeout argument to app_originate the + default is 30s this will be used if the timout supplied is + invalid or no timeout is supplied. Contributed by: jacco (thank + you for the work) Review: + https://reviewboard.asterisk.org/r/1310/ ........ + +2011-09-21 09:39 +0000 [r337179-337220] Olle Johansson + + * main/pbx.c, /, CHANGES, configs/extensions.conf.sample: Merged + revisions 337219 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13 + lines Make ast_pbx_run() not default to s@default if extension is + not found Review: https://reviewboard.asterisk.org/r/1446/ This + is a bug - or architecture mistake - that has been in Asterisk + for a very long time. It was exposed by the AMI originate action + and possibly some other applications. Most channel drivers checks + if an extension exists BEFORE starting a pbx on an inbound call, + so most calls will not depend on this issue. Thanks everyone + involved in the review and on IRC and the mailing list for a + quick review and all the feedback. (closes issue ASTERISK-18578) + ........ + + * res/res_rtp_asterisk.c, /, configs/rtp.conf.sample, CHANGES: + Merged revisions 337178 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r337178 | oej | 2011-09-21 10:51:41 +0200 (Ons, 21 Sep 2011) | 14 + lines Change strictrtp option to default to yes in the RTP module + Suggested by Kapejod on Facebook Review: + https://reviewboard.asterisk.org/r/1448/ (closes issue + ASTERISK-18587) Thanks for quick feedback to kpfleming and + Tilghman --Denna och nedanstående rader kommer inte med i + loggmeddelandet-- M CHANGES M configs/rtp.conf.sample M + res/res_rtp_asterisk.c ........ + +2011-09-20 23:02 +0000 [r337124] Matthew Jordan + + * apps/app_dial.c, include/asterisk/app.h, apps/app_meetme.c, + apps/app_minivm.c, main/app.c, apps/app_confbridge.c, + apps/app_followme.c, apps/app_voicemail.c: Merged revisions + 337120 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r337120 | mjordan | 2011-09-20 17:49:36 -0500 + (Tue, 20 Sep 2011) | 28 lines Merged revisions 337118 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337118 | mjordan | 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) + | 21 lines Fix for incorrect voicemail duration in external + notifications This patch fixes an issue where the voicemail + duration was being reported with a duration significantly less + than the actual sound file duration. Voicemails that contained + mostly silence were reporting the duration of only the sound in + the file, as opposed to the duration of the file with the + silence. This patch fixes this by having two durations reported + in the __ast_play_and_record family of functions - the + sound_duration and the actual duration of the file. The + sound_duration, which is optional, now reports the duration of + the sound in the file, while the actual full duration of the file + is reported in the duration parameter. This allows the voicemail + applications to use the sound_duration for minimum duration + checking, while reporting the full duration to external parties + if the voicemail is kept. (issue ASTERISK-2234) (closes issue + ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad + House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: + https://reviewboard.asterisk.org/r/1443 ........ ................ + +2011-09-20 22:54 +0000 [r337121-337123] Richard Mudgett + + * /, funcs/func_strings.c: Merged revisions 337119 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r337119 | rmudgett | 2011-09-20 17:47:45 -0500 (Tue, 20 Sep 2011) + | 16 lines Fix crash with STRREPLACE function. The + ast_func_read() function calls the .read2 callback with the len + parameter set to zero indicating no size restrictions on the + supplied ast_str buffer. The value was used to dimension a local + starts[] array with the array subsequently used. * Reworked the + strreplace() function to perform the string replacement in a + straight forward manner. Eliminated the need for the starts[] + array. (closes issue ASTERISK-18545) Reported by: Federico Alves + Patches: jira_asterisk_18545_v10.patch (license #5621) patch + uploaded by rmudgett Tested by: rmudgett, Federico Alves ........ + + * /: Updated 10 merge property. + + * /: Restore branch-10 merge properties. + +2011-09-20 22:29 +0000 [r337117] Leif Madsen + + * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 337115 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337115 | lmadsen | 2011-09-20 17:18:25 -0500 (Tue, 20 Sep 2011) + | 7 lines Update RedHat Init script to work with Heartbeat. The + current RedHat init script was not LSB compatible. This change + will make it LSB compatible so that it can work correctly with + Heartbeat. (Closes issue ASTERISK-18253) Reported by: c0rnoTa + ........ + +2011-09-20 21:05 +0000 [r337063] Kinsey Moore + + * main/pbx.c, /, tests/test_pbx.c: Merged revisions 337062 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r337062 | kmoore | 2011-09-20 16:05:01 -0500 + (Tue, 20 Sep 2011) | 18 lines Merged revisions 337061 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337061 | kmoore | 2011-09-20 16:04:11 -0500 (Tue, 20 Sep 2011) | + 11 lines Make CANMATCH with the new pattern match engine behave + more like the old one When checking an extension for E_CANMATCH + using the new extension matching algorithm, an exact match was + not returned as a possible match resulting in the queue failing + to allow a caller to exit on DTMF. This removes the requirement + that an extension be longer than acquired digits for an + E_CANMATCH operation to succeed. (closes issue ASTERISK-18044) + Review: https://reviewboard.asterisk.org/r/1367/ ........ + ................ + +2011-09-20 19:13 +0000 [r336988-337009] Richard Mudgett + + * /, channels/sig_ss7.c: Merged revisions 337008 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r337008 | rmudgett | 2011-09-20 14:12:24 -0500 + (Tue, 20 Sep 2011) | 22 lines Merged revisions 337007 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) + | 15 lines Check if a channel was created before using the + pointer in sig_ss7_new_ast_channel(). Fixes the crash in + ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing + libss7 access lock protection. * Prevent cancelling the + ss7_linkset() thread at inoportune times just like the + pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M + Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621) + patch uploaded by rmudgett (attached to related ASTERISK-17966) + ........ ................ + + * /, channels/sig_ss7.c: Merged revisions 336978 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336978 | rmudgett | 2011-09-20 13:14:40 -0500 + (Tue, 20 Sep 2011) | 28 lines Merged revisions 336977 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) + | 21 lines Fix deadlock from not releasing SS7 linkset lock. + sig_ss7_hangup() failed to release the SS7 linkset lock if the + call had the alreadyhungup flag set. * Made unlock the SS7 + linkset lock in sig_ss7_hangup() if the alreadyhungup flag is + set. * Made ss7_start_call() not hold any locks while creating + the channel for an incoming call to prevent deadlock. * Made + ss7_grab() a void function, since it could never fail, to + simplify calling code. * Made obtain the channel lock to do + softhangup in some places. Patches: jira_ast_668_v1.8.patch + (license #5621) patch uploaded by rmudgett JIRA AST-668 ........ + ................ + +2011-09-20 16:56 +0000 [r336937] Gregory Nietsky + + * channels/sip/sdp_crypto.c, /, channels/chan_sip.c, + channels/sip/include/sdp_crypto.h, channels/sip/include/srtp.h, + configs/sip.conf.sample, CHANGES, channels/sip/include/sip.h: + Merged revisions 336936 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r336936 | irroot | 2011-09-20 18:51:59 +0200 (Tue, 20 Sep 2011) | + 14 lines Allow Setting Auth Tag Bit length Based on invite or + config option Update the SIP SRTP API to allow use of 32 or 80 + bit taglen. Curently only 80 bit is supported. The outgoing + invite will use the taglen of the incoming invite preventing + one-way audio. (Closes issue ASTERISK-17895) Review: + https://reviewboard.asterisk.org/r/1173/ ........ + +2011-09-20 01:11 +0000 [r336879] Russell Bryant + + * res/res_rtp_asterisk.c, /: Merged revisions 336878 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336878 | russell | 2011-09-19 20:03:55 -0500 + (Mon, 19 Sep 2011) | 43 lines Merged revisions 336877 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336877 | russell | 2011-09-19 19:56:20 -0500 (Mon, 19 Sep 2011) + | 36 lines Fix crashes in ast_rtcp_write(). This patch addresses + crashes related to RTCP handling. The backtraces just show a + crash in ast_rtcp_write() where it appears that the RTP instance + is no longer valid. There is a race condition with scheduled RTCP + transmissions and the destruction of the RTP instance. This patch + utilizes the fact that ast_rtp_instance is a reference counted + object and ensures that it will not get destroyed while a + reference is still around due to scheduled RTCP transmissions. + RTCP transmissions are scheduled and executed from the chan_sip + scheduler context. This scheduler context is processed in the SIP + monitor thread. The destruction of an RTP instance occurs when + the associated sip_pvt gets destroyed (which happens when the + sip_pvt reference count reaches 0). However, the SIP monitor + thread is not the only thread that can cause a sip_pvt to get + destroyed. The sip_hangup function, executed from a channel + thread, also decrements the reference count on a sip_pvt and + could cause it to get destroyed. While this is being changed + anyway, the patch also removes calling ast_sched_del() from + within the RTCP scheduler callback. It's not helpful. Simply + returning 0 prevents the callback from being rescheduled. (closes + issue ASTERISK-18570) Related issues that look like they are the + same problem: (issue ASTERISK-17560) (issue ASTERISK-15406) + (issue ASTERISK-15257) (issue ASTERISK-13334) (issue + ASTERISK-9977) (issue ASTERISK-9716) Review: + https://reviewboard.asterisk.org/r/1444/ ........ + ................ + +2011-09-19 22:28 +0000 [r336837] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 336792 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336792 | twilson | 2011-09-19 17:13:34 -0500 + (Mon, 19 Sep 2011) | 9 lines Merged revisions 336791 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r336791 | twilson | 2011-09-19 17:07:58 -0500 (Mon, 19 + Sep 2011) | 2 lines Don't interfere with T.38 reinvites This is + an update to the fix for ASTERISK-18340 and ASTERISK-17725 + ........ ................ + +2011-09-19 21:42 +0000 [r336735-336790] Tilghman Lesher + + * /, funcs/func_strings.c: Merged revisions 336789 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r336789 | tilghman | 2011-09-19 16:41:16 -0500 (Mon, 19 Sep 2011) + | 2 lines Ensure substring will not be found in the previous + match. ........ + + * Makefile, /, configure, include/asterisk/autoconfig.h.in, + main/Makefile, codecs/gsm/Makefile, configure.ac, Makefile.rules, + include/asterisk/optional_api.h: Merged revisions 336734 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336734 | tilghman | 2011-09-19 15:29:40 -0500 + (Mon, 19 Sep 2011) | 18 lines Merged revisions 336733 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011) + | 11 lines Various changes to allow 1.8 to compile on Mac OS X + Lion (10.7) * Makefile workaround for 10.6 extended to work on + 10.7 and later. * Now uses the 'weak' symbol for Lion systems, + which no longer support 'weak_import' Closes ASTERISK-17612. + Closes ASTERISK-18213. Tested by: tilghman, oej. ........ + ................ + +2011-09-19 20:23 +0000 [r336732] Jonathan Rose + + * /, apps/app_echo.c, apps/app_saycounted.c, apps/app_mp3.c, + apps/app_morsecode.c, res/res_musiconhold.c, apps/app_queue.c, + apps/app_mixmonitor.c: Merged revisions 336717 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336717 | jrose | 2011-09-19 15:16:23 -0500 + (Mon, 19 Sep 2011) | 14 lines Merged revisions 336716 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336716 | jrose | 2011-09-19 15:07:36 -0500 (Mon, 19 Sep 2011) | + 7 lines Document applications that play audio and do not answer + unanswered calls. This patch is part of an effort to document + early media and its usage. If you are interested in contributing + to this documentation effort, there are probably other + applications worth documenting as well as an Asterisk wiki + article at + https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application + ........ ................ + +2011-09-19 19:03 +0000 [r336660-336662] Richard Mudgett + + * apps/app_dial.c, /, UPGRADE-1.8.txt: Merged revisions 336659 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336659 | rmudgett | 2011-09-19 13:51:19 -0500 + (Mon, 19 Sep 2011) | 38 lines Merged revisions 336658 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) + | 31 lines Made Dial d and H options no longer immediately + auto-answer the calling leg. The Dial d and H options break DTMF + attended transfer atxferdropcall option. 1) Party A calls party + B. 2) Party B does a DTMF attended transfer to Party C. If the + dialplan uses the Dial d or H options to call Party C then the + Dial application answers the call immediately before initiating + the call leg to Party C. The premature answer causes the transfer + code to not invoke the atxferdropcall=no behavior for a blonde + transfer since Party C has "answered". The transfer code thinks + that Party B has "consulted" with Party C when Party B hangs up + and completes the transfer to Party A. Party A now hears ringback + until Party C actually answers. ASTERISK-13294 Dial d option. + ASTERISK-11067 Dial H option to disconnect before answer. The + referenced issues made Dial answer with the d and H options + because many SIP and ISDN phones cannot send DTMF before the call + is connected. * Made require the dialplan to control when or if + the call needs to be answered to use the Dial application d and H + options. (The call is no longer surprise answered when using the + Dial d or H options.) Review: + https://reviewboard.asterisk.org/r/1381/ JIRA AST-623 JIRA + AST-666 ........ ................ + + * /: Update merge 10 branch merge propterty. + + * /: Restore 10 branch merge properties. + +2011-09-19 16:22 +0000 [r336600] Jason Parker + + * cel/cel_odbc.c, configs/cel_odbc.conf.sample, sounds/Makefile: + Remove weird mergeinfo props that make merges annoying sometimes. + +2011-09-19 15:48 +0000 [r336574] Leif Madsen + + * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 336572 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336572 | lmadsen | 2011-09-19 10:41:16 -0500 (Mon, 19 Sep 2011) + | 7 lines Update get_ilbc_source.sh script to work again. + Recently iLBC support in Asterisk has changed after the + acquisition of GIPS by Google. More information about how this + may affect you is available in a blog post at: + http://blogs.asterisk.org/2011/09/19/ilbc-support-in-asterisk-after-googles-acquisition-of-gips/ + ........ + +2011-09-19 15:36 +0000 [r336571] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 336570 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336570 | rmudgett | 2011-09-19 10:32:00 -0500 + (Mon, 19 Sep 2011) | 11 lines Merged revisions 336569 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336569 | rmudgett | 2011-09-19 10:25:34 -0500 (Mon, 19 Sep 2011) + | 4 lines Rework sig_pri_hangup() to be simpler and clearer. JIRA + AST-675 ........ ................ + +2011-09-19 13:57 +0000 [r336505] Olle Johansson + + * /, channels/chan_sip.c: Merged revisions 336502 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336502 | oej | 2011-09-19 15:38:53 +0200 (Mån, + 19 Sep 2011) | 12 lines Merged revisions 336501 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336501 | oej | 2011-09-19 15:33:50 +0200 (Mån, 19 Sep 2011) | 5 + lines Add diversion header to a 302 redirect response if we have + diversion data (closes issue ASTERISK-18143) patch by oej + ........ ................ + +2011-09-19 13:41 +0000 [r336503] Gregory Nietsky + + * /, channels/chan_h323.c: Merged revisions 336500 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336500 | irroot | 2011-09-19 15:31:50 +0200 + (Mon, 19 Sep 2011) | 19 lines Merged revisions 336499 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336499 | irroot | 2011-09-19 15:27:52 +0200 (Mon, 19 Sep 2011) | + 13 lines A long time ago in a galaxy far far away a IPv6 update + was made, chan_h323 was not updated causeing all to flee to + chan_ooh323. the brave Jedi [asterisk developers] pondered this + miscarrige of justice and restored order to the force for the + sake of closing out 2 old issues. (closes issue ASTERISK-17278) + (closes issue ASTERISK-17500) Reported by: dread, sybasesql + Tested by: irroot Reviewed by: IRC (russellb, kpfleming) ........ + ................ + +2011-09-19 12:20 +0000 [r336382-336453] Olle Johansson + + * main/manager.c, /: Merged revisions 336441 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336441 | oej | 2011-09-19 14:15:06 +0200 (Mån, + 19 Sep 2011) | 9 lines Merged revisions 336440 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336440 | oej | 2011-09-19 14:06:48 +0200 (Mån, 19 Sep 2011) | 2 + lines Make sure manager_debug option is reset at reload ........ + ................ + + * /, channels/chan_sip.c: Merged revisions 336381 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336381 | oej | 2011-09-19 12:05:00 +0200 (Mån, + 19 Sep 2011) | 16 lines Merged revisions 336378 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336378 | oej | 2011-09-19 11:40:44 +0200 (Mån, 19 Sep 2011) | 9 + lines Add missing unlock at MWI message sending time (closes + issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041) + by Gregory Hinton Nietsky Thanks to irrot for the reminder, to + Gregory for the patch! ........ ................ + +2011-09-16 22:12 +0000 [r336315-336317] Terry Wilson + + * /, funcs/func_frame_trace.c: Merged revisions 336316 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336316 | twilson | 2011-09-16 17:11:39 -0500 + (Fri, 16 Sep 2011) | 9 lines Merged revisions 336314 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r336314 | twilson | 2011-09-16 17:10:56 -0500 (Fri, 16 + Sep 2011) | 2 lines Whitespace fix ........ ................ + + * /, funcs/func_frame_trace.c: Merged revisions 336313 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336313 | twilson | 2011-09-16 17:07:00 -0500 + (Fri, 16 Sep 2011) | 12 lines Merged revisions 336312 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336312 | twilson | 2011-09-16 17:04:25 -0500 (Fri, 16 Sep 2011) + | 5 lines Add missing frame types to func_frame_trace Also casts + control frames to the proper enum so that the compile will catch + new additions. ........ ................ + +2011-09-16 21:20 +0000 [r336311] Jonathan Rose + + * main/channel.c, main/rtp_engine.c, /, channels/chan_sip.c, + include/asterisk/frame.h: Merged revisions 336307 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336307 | jrose | 2011-09-16 16:09:20 -0500 + (Fri, 16 Sep 2011) | 20 lines Merged revisions 336294 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336294 | jrose | 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) | + 13 lines Fix bad RTP media bridges in directmedia calls on peers + separated by multiple Asterisk nodes. In a situation involving + devices on separate Asterisk trunks, the remote RTP bridge would + break when starting a call with directmedia. This patch queues a + new type of control frame so that our RTP bridge loop can + properly detect when these situations occur and check to see if + peers need to be updated in order to send their media to the + proper location. (Closes issue ASTERISK-18340) Reported by: + Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk + Tested by: twilson, jrose ........ ................ + +2011-09-16 19:11 +0000 [r336236] Sean Bright + + * /, UPGRADE-1.8.txt: Merged revisions 336235 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336235 | seanbright | 2011-09-16 15:10:39 -0400 + (Fri, 16 Sep 2011) | 9 lines Merged revisions 336234 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r336234 | seanbright | 2011-09-16 15:06:27 -0400 (Fri, + 16 Sep 2011) | 2 lines Make a note that inotify won't work with + an NFS mounted spooler directory. ........ ................ + +2011-09-16 10:16 +0000 [r336095-336168] Gregory Nietsky + + * channels/chan_misdn.c, /: Merged revisions 336167 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336167 | irroot | 2011-09-16 12:12:03 +0200 + (Fri, 16 Sep 2011) | 22 lines Merged revisions 336166 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336166 | irroot | 2011-09-16 12:09:17 +0200 (Fri, 16 Sep 2011) | + 16 lines The round robin routing routine in chan_misdn.c is + broken. it rotates between ports but never checks the channels in + the ports. i have extensivly tested it and verified it works on 1 + upto 4 ports. before the patch only 1 out of each port was used + now all are used as expected. (closes issue ASTERISK-18413) + Reported by: irroot Tested by: irroot Reviewed by: irroot Review: + https://reviewboard.asterisk.org/r/1410/ ........ + ................ + + * /, apps/app_queue.c: Merged revisions 336094 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r336094 | irroot | 2011-09-15 17:54:46 +0200 + (Thu, 15 Sep 2011) | 26 lines Merged revisions 336093 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r336093 | irroot | 2011-09-15 17:46:21 +0200 (Thu, 15 Sep 2011) | + 20 lines Locking order in app_queue.c causes deadlocks. a channel + lock must never be held with the queues container lock held. the + deadlock occured on masquerade. the queues container lock is a + relic of the past the old queue module lock. with ao2 there is no + need to hold this lock when dealing with members this patch + removes unneeded locks. (closes issue ASTERISK-18101) (closes + issue ASTERISK-18487) Reported by: Paul Rolfe, Jason Legault + Tested by: irroot, Jason Legault, Paul Rolfe Reviewed by: Matthew + Nicholson Review: https://reviewboard.asterisk.org/r/1402/ + ........ ................ + +2011-09-15 15:19 +0000 [r336092] David Vossel + + * /, main/format_cap.c: Merged revisions 336091 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r336091 | dvossel | 2011-09-15 10:19:10 -0500 (Thu, 15 Sep 2011) + | 2 lines Removes some no-op code found in format_cap.c. ........ + +2011-09-15 12:50 +0000 [r336043] Olle Johansson + + * CREDITS, /, apps/app_meetme.c, CHANGES: Merged revisions 336042 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r336042 | oej | 2011-09-15 14:46:38 +0200 (Tor, 15 Sep 2011) | 12 + lines Meetme: Introducing a new option "k" to kill a conference + if there's only a single member left. When using Meetme as a + modular call bridge from third party applications, it's handy to + make it behave like a normal call bridge. When the second to last + person exists, the last person will be kicked out of the + conference when this option is enabled. (closes issue + ASTERISK-18234) Review: https://reviewboard.asterisk.org/r/1376/ + Patch by oej, sponsored by ClearIT, Solna, Sweden ........ + +2011-09-15 08:40 +0000 [r335993] Gregory Nietsky + + * /, channels/chan_agent.c: Merged revisions 335991 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335991 | irroot | 2011-09-15 10:29:12 +0200 + (Thu, 15 Sep 2011) | 17 lines Merged revisions 335978 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335978 | irroot | 2011-09-15 10:15:22 +0200 (Thu, 15 Sep 2011) | + 11 lines lock the channel before calling ast_bridged_channel() to + prevent a seg fault. AMI agents list called on shutdown causes a + segfault, introducing proper locking will prevent this. (closes + issue ASTERISK-18092) Reported by: agustina Patches: + chan_agent.patch (License #5041) patch uploaded by irroot + ........ ................ + +2011-09-14 18:38 +0000 [r335853-335913] Richard Mudgett + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 335912 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335912 | rmudgett | 2011-09-14 13:31:15 -0500 + (Wed, 14 Sep 2011) | 20 lines Merged revisions 335911 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) + | 13 lines Remove unnecessary libpri dependency checks in the + configure script. Using the --with-pri option with the configure + script generated an error about not having PRI_L2_PERSISTENCE if + you did not have the absolute latest libpri SVN checkout + installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the + configure.ac script seems to be for libraries that are dependent + upon other libraries and not necessarily for optional/added + features within a library. (closes issue ASTERISK-18535) Reported + by: Michael Keuter ........ ................ + + * channels/chan_dahdi.c, /: Merged revisions 335852 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335852 | rmudgett | 2011-09-14 11:00:37 -0500 + (Wed, 14 Sep 2011) | 18 lines Merged revisions 335851 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335851 | rmudgett | 2011-09-14 10:53:25 -0500 (Wed, 14 Sep 2011) + | 11 lines Fixed cut-n-paste regression using the wrong variable. + Fixes the missing DAHDI channels when using the newer + chan_dahdi.conf sections for channel configuration. (closes issue + ASTERISK-18496) Reported by: Sean Darcy Patches: + jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by + rmudgett Tested by: Sean Darcy, rmudgett ........ + ................ + +2011-09-14 13:29 +0000 [r335792] Matthew Nicholson + + * main/manager.c, /: Merged revisions 335791 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335791 | mnicholson | 2011-09-14 08:28:50 -0500 + (Wed, 14 Sep 2011) | 11 lines Merged revisions 335790 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335790 | mnicholson | 2011-09-14 08:28:16 -0500 (Wed, 14 Sep + 2011) | 4 lines The tech and data members of + fast_originate_helper are not string fields. ASTERISK-17709 + ........ ................ + +2011-09-13 22:11 +0000 [r335722] Richard Mudgett + + * /, apps/app_directed_pickup.c: Merged revisions 335721 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335721 | rmudgett | 2011-09-13 17:10:44 -0500 + (Tue, 13 Sep 2011) | 9 lines Merged revisions 335720 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r335720 | rmudgett | 2011-09-13 17:10:15 -0500 (Tue, 13 + Sep 2011) | 1 line Remove obsolete todo comment about + PICKUPRESULT. ........ ................ + +2011-09-13 21:52 +0000 [r335719] Paul Belanger + + * main/dnsmgr.c: Additional updates for parsing dnsmgr.conf Review: + https://reviewboard.asterisk.org/r/1432/ + +2011-09-13 21:40 +0000 [r335718] Tzafrir Cohen + + * main/asterisk.c: do parse defaultlanguage from asterisk.conf Do + parse the option "defaultlanguage" from the [options] section of + asterisk.conf, as in the sample config file. Otherwise the + build-time default language (normally "en") is always the default + one. Review: https://reviewboard.asterisk.org/r/1342/ + Signed-off-by: Tzafrir Cohen (License #5035) + Original-Commit: + http://svn.digium.com/svn/asterisk/branches/1.8@335716 + Original-Commit: + http://svn.digium.com/svn/asterisk/branches/10@335717 + +2011-09-13 18:56 +0000 [r335657] Tilghman Lesher + + * /, configure, configure.ac: Merged revisions 335656 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335656 | tilghman | 2011-09-13 13:55:33 -0500 + (Tue, 13 Sep 2011) | 11 lines Merged revisions 335655 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335655 | tilghman | 2011-09-13 13:52:38 -0500 (Tue, 13 Sep 2011) + | 4 lines Move mandatory checks closer to the beginning of the + file. If these are going to fail, they should fail as quickly as + possible. ........ ................ + +2011-09-13 18:49 +0000 [r335654] Matthew Nicholson + + * main/pbx.c, main/manager.c, /: Merged revisions 335653 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335653 | mnicholson | 2011-09-13 13:47:57 -0500 + (Tue, 13 Sep 2011) | 12 lines Merged revisions 335618 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335618 | mnicholson | 2011-09-13 13:20:52 -0500 (Tue, 13 Sep + 2011) | 5 lines Don't limit the size of appdata for manager + originate actions. ASTERISK-17709 Patch by: tilghman (with + modifications) ........ ................ + +2011-09-13 18:11 +0000 [r335555-335603] Paul Belanger + + * UPGRADE.txt, main/dsp.c: Clean up dsp.conf parsing Review: + https://reviewboard.asterisk.org/r/1434/ + + * UPGRADE.txt, cdr/cdr_csv.c: Clean up cdr.conf parsing for [csv] + section Review: https://reviewboard.asterisk.org/r/1427/ + + * main/dnsmgr.c, UPGRADE.txt: Clean up dnsmgr.conf parsing Review: + https://reviewboard.asterisk.org/r/1432/ + +2011-09-13 07:35 +0000 [r335511] Russell Bryant + + * include/asterisk/event.h, /, res/ais/evt.c, main/event.c: Merged + revisions 335510 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335510 | russell | 2011-09-13 02:24:34 -0500 + (Tue, 13 Sep 2011) | 22 lines Merged revisions 335497 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335497 | russell | 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) + | 15 lines Fix a crash in res_ais. This patch resolves a crash + observed in a load testing environment that involved the use of + the res_ais module. I observed some crashes where the event + delivery callback would get called, but the length parameter + incidcating how much data there was to read was 0. The code + assumed (with good reason I would think) that if this callback + got called, there was an event available to read. However, if the + rare case that there's nothing there, catch it and return instead + of blowing up. More specifically, the change always ensure that + the size of the received event in the cluster is always big + enough to be a real ast_event. Review: + https://reviewboard.asterisk.org/r/1423/ ........ + ................ + +2011-09-12 15:56 +0000 [r335435] Matthew Nicholson + + * main/channel.c, /: Merged revisions 335434 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335434 | mnicholson | 2011-09-12 10:55:48 -0500 + (Mon, 12 Sep 2011) | 13 lines Merged revisions 335433 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335433 | mnicholson | 2011-09-12 10:54:41 -0500 (Mon, 12 Sep + 2011) | 6 lines Properly set caller_warning and callee_warning + before we try to use them. ASTERISK-18199 Patch by: elguero + Testing by: rtang ........ ................ + +2011-09-12 14:33 +0000 [r335385] Olle Johansson + + * channels/chan_sip.c: Documentation updates + +2011-09-12 14:24 +0000 [r335354] Kinsey Moore + + * apps/app_dial.c, /: Merged revisions 335346 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335346 | kmoore | 2011-09-12 09:22:15 -0500 + (Mon, 12 Sep 2011) | 17 lines Merged revisions 335341 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335341 | kmoore | 2011-09-12 09:21:17 -0500 (Mon, 12 Sep 2011) | + 10 lines Ensure frames are not written to dialed channel if + ringback is requested When a single channel was dialed and there + was media to be forwarded to the calling channel, the media was + written without regard for ringback causing silence to be heard + in some circumstances. This regression was introduced when the + meaning of "single" changed to mean only the number of channels + dialed. (closes issue ASTERISK-18083) ........ ................ + +2011-09-12 14:22 +0000 [r335324-335349] Olle Johansson + + * channels/chan_sip.c: Small documentation updates + + * CREDITS, channels/chan_sip.c, include/asterisk/indications.h, + UPGRADE.txt, configs/sip.conf.sample, channels/sip/include/sip.h: + New sip.conf option for setting default tonezone for channel or + individual devices Review: + https://reviewboard.asterisk.org/r/1429/ (closes issue + ASTERISK-18497) Thanks to russellb for peer review. + + * /, channels/chan_sip.c: Merged revisions 335323 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335323 | oej | 2011-09-12 15:47:13 +0200 (Mån, + 12 Sep 2011) | 19 lines Merged revisions 335319 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335319 | oej | 2011-09-12 15:25:30 +0200 (Mån, 12 Sep 2011) | 12 + lines Lock the peer->mvipvt to avoid crashes with SIP history + enabled After the launch of 1.6 event-based MWI we have two + threads handling the peer->mwipvt, which cause issues with SIP + history additions in combination with the max limit for number of + history entries. Review: https://reviewboard.asterisk.org/r/1373/ + (closes issue ASTERISK-18288) Thanks to irrot for peer review. + Work with this bug funded by IPvision AS ........ + ................ + +2011-09-12 13:27 +0000 [r335322] Kinsey Moore + + * /, channels/chan_iax2.c: Merged revisions 335321 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335321 | kmoore | 2011-09-12 08:27:04 -0500 + (Mon, 12 Sep 2011) | 16 lines Merged revisions 335320 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335320 | kmoore | 2011-09-12 08:25:42 -0500 (Mon, 12 Sep 2011) | + 9 lines Prevent IAX2 from getting IPv6 addresses via DNS IAX2 + does not support IPv6 and getting such addresses from DNS can + cause error messages on the remote end involving bad IPv4 address + casts in the presence of IPv6/IPv4 tunnels. This patch ensures + that IAX2 will not encounter IPv6 addresses via DNS queries. + (closes issue ASTERISK-18090) ........ ................ + +2011-09-12 11:15 +0000 [r335261] Stefan Schmidt + + * /, channels/chan_sip.c: Merged revisions 335260 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335260 | schmidts | 2011-09-12 11:11:45 +0000 + (Mon, 12 Sep 2011) | 12 lines Merged revisions 335259 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335259 | schmidts | 2011-09-12 11:09:19 +0000 (Mon, 12 Sep 2011) + | 6 lines build_peer doesnt unlink a peer object from peers_by_ip + container which leads to a wrong refcounter value. adding an + ao2_unlink from the peers_by_ip container fix it. Review: + https://reviewboard.asterisk.org/r/1428/ ........ + ................ + +2011-09-12 03:10 +0000 [r335170-335212] Paul Belanger + + * UPGRADE.txt: Be more specific on which section has changed. + + * main/cdr.c, UPGRADE.txt: Iterate though cdr.conf setting Review: + https://reviewboard.asterisk.org/r/1426/ + +2011-09-11 17:09 +0000 [r335129] Terry Wilson + + * configs/res_config_sqlite3.conf.sample (added), + res/res_config_sqlite3.c (added): Add SQLite 3 realtime support + +2011-09-09 16:28 +0000 [r335079] Matthew Jordan + + * channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, + addons/chan_ooh323.c, channels/chan_sip.c, + channels/chan_console.c, channels/sig_pri.c, channels/chan_oss.c, + main/channel.c, channels/chan_usbradio.c, main/dial.c, + channels/chan_dahdi.c, channels/chan_misdn.c, + channels/chan_skinny.c, funcs/func_frame_trace.c, + main/features.c, channels/chan_h323.c, channels/chan_alsa.c, + include/asterisk/frame.h, channels/sig_ss7.c, + channels/chan_mgcp.c: Merged revisions 335078 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500 + (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) + | 23 lines Updated SIP 484 handling; added Incomplete control + frame When a SIP phone uses the dial application and receives a + 484 Address Incomplete response, if overlapped dialing is enabled + for SIP, then the 484 Address Incomplete is forwarded back to the + SIP phone and the HANGUPCAUSE channel variable is set to 28. + Previously, the Incomplete application dialplan logic was + automatically triggered; now, explicit dialplan usage of the + application is required. Additionally, this patch adds a new + AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel + driver receives this control frame, it is an indication that the + dialplan expects more digits back from the device. If the device + supports overlap dialing it should attempt to notify the device + that the dialplan is waiting for more digits; otherwise, it can + handle the frame in a manner appropriate to the channel driver. + (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested + by: Matthew Jordan Review: + https://reviewboard.asterisk.org/r/1416/ ........ + ................ + +2011-09-09 07:28 +0000 [r335015] Gregory Nietsky + + * funcs/func_dialplan.c, /, apps/app_readexten.c, CHANGES: Merged + revisions 335014 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r335014 | irroot | 2011-09-09 09:23:53 +0200 (Fri, 09 Sep 2011) | + 9 lines Move code for VALID_EXTEN from app_readexten to + func_dialplan Mark VALID_EXTEN deprecated. Review: + https://reviewboard.asterisk.org/r/1396/ ........ + +2011-09-08 22:30 +0000 [r334955] Richard Mudgett + + * /, main/logger.c: Merged revisions 334954 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334954 | rmudgett | 2011-09-08 17:28:56 -0500 + (Thu, 08 Sep 2011) | 17 lines Merged revisions 334953 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334953 | rmudgett | 2011-09-08 17:27:40 -0500 (Thu, 08 Sep 2011) + | 10 lines Fix crash with res_fax when MALLOC_DEBUG and "core + stop gracefully" are used. Asterisk crashes if MALLOC_DEBUG is + enabled when res_fax tries to unregister its logger level. * Make + ast_logger_unregister_level() use ast_free() instead of free(). + When MALLOC_DEBUG is enabled, ast_free() does not degenerate into + a call to free(). Therefore, if you allocated memory with a form + of ast_malloc you must free it with ast_free. ........ + ................ + +2011-09-08 13:36 +0000 [r334907] Jonathan Rose + + * main/cdr.c, main/pbx.c: Removes colorful verb statements + erroneously commited with r332760 + +2011-09-07 19:38 +0000 [r334845] Paul Belanger + + * /, channels/chan_iax2.c: Merged revisions 334844 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334844 | pabelanger | 2011-09-07 15:37:24 -0400 + (Wed, 07 Sep 2011) | 11 lines Merged revisions 334843 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334843 | pabelanger | 2011-09-07 15:35:52 -0400 (Wed, 07 Sep + 2011) | 4 lines Cleanup chan_iax2.c log messages Review: + https://code.asterisk.org/code/cru/CR-AST-11 ........ + ................ + +2011-09-07 19:35 +0000 [r334842] Richard Mudgett + + * /, main/features.c: Merged revisions 334841 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334841 | rmudgett | 2011-09-07 14:33:38 -0500 + (Wed, 07 Sep 2011) | 17 lines Merged revisions 334840 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334840 | rmudgett | 2011-09-07 14:31:44 -0500 (Wed, 07 Sep 2011) + | 10 lines Fix AMI action Park crash. * Made AMI action Park not + say anything to the parker channel (AMI header Channel2) since + the AMI action is a third party parking the call. (This is a + change in behavior that cannot be preserved without a lot of + effort.) * Made not play pbx-parkingfailed if the Park 's' option + is used. JIRA AST-660 ........ ................ + +2011-09-07 15:37 +0000 [r334683-334792] Stefan Schmidt + + * /, main/features.c: Merged revisions 334747 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334747 | schmidts | 2011-09-07 15:10:37 +0000 + (Wed, 07 Sep 2011) | 9 lines Merged revisions 334682 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 + Sep 2011) | 3 lines Adding the Feature to sent a Reason Header in + a SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE + before doing a masquerade in the pickup function. ........ + ................ + + * main/features.c: clean up wrong merged stuff + + * /, main/features.c: Merged revisions 334682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r334682 | schmidts | 2011-09-07 13:26:50 +0000 (Wed, 07 Sep 2011) + | 3 lines Adding the Feature to sent a Reason Header in a SIP + Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE before + doing a masquerade in the pickup function. ........ + + * main/features.c: Adding the Feature to sent a Reason Header in a + SIP Cancel message by set the flag AST_FLAG_ANSWERED_ELSEWHERE + before doing a masquerade in the pickup function. + +2011-09-07 08:17 +0000 [r334618-334623] Alec L Davis + + * /, CHANGES, apps/app_queue.c: Merged revisions 334621 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334621 | alecdavis | 2011-09-07 20:14:50 +1200 + (Wed, 07 Sep 2011) | 9 lines Merged revisions 334620 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r334620 | alecdavis | 2011-09-07 20:12:49 +1200 (Wed, 07 + Sep 2011) | 2 lines peroid typo ........ ................ + + * main/logger.c: log Asterisk Version number, Build etc into each + log file Allow tracking of previous versions through log file + records to be tracked. Each time log file is created or opened, + log Asterisk Version, Buildinfo. etc. alecdavis (license 585) + Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/1409/ + + * main/pbx.c, /: Merged revisions 334617 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334617 | alecdavis | 2011-09-07 19:45:00 +1200 + (Wed, 07 Sep 2011) | 17 lines Merged revisions 334616 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334616 | alecdavis | 2011-09-07 19:33:39 +1200 (Wed, 07 Sep + 2011) | 10 lines Prevent segfault if call arrives before Asterisk + is fully booted. Prevent ast_pbx_start and ast_run_start from + starting a new thread unless asterisk is fully booted. alecdavis + (license 585) Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/1407/ ........ + ................ + +2011-09-07 00:54 +0000 [r334574] Tilghman Lesher + + * main/frame.c, contrib/realtime/mysql/iaxfriends.sql, + contrib/realtime/postgresql/realtime.sql, + configs/sip.conf.sample, CHANGES, + contrib/realtime/mysql/sipfriends.sql: Implement the '!' negation + element to negate codecs directly in the allow keyword. This + permits the list of codecs to be specified in one configuration + line, instead of two or more, generally with the aim of either + allowing all codecs with the exception of a few or disallowing + most but permitting a few. Review: + https://reviewboard.asterisk.org/r/1411/ + +2011-09-06 16:15 +0000 [r334519] Gregory Nietsky + + * /, apps/app_voicemail.c: Merged revisions 334455 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334455 | irroot | 2011-09-06 15:58:56 +0200 + (Tue, 06 Sep 2011) | 18 lines Merged revisions 334453 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | + 13 lines Make SQL query in app_voicemail.c portable LIMIT is not + portable. Regression from r312212 (closes issue ASTERISK-18255) + Reported by: Leif Madsen Tested by: Leif Madsen Review: + https://reviewboard.asterisk.org/r/1415/ ........ + ................ + +2011-09-06 16:08 +0000 [r334517] Paul Belanger + + * configs/iax.conf.sample, /, CHANGES, channels/chan_iax2.c: Merged + revisions 334514 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r334514 | pabelanger | 2011-09-06 11:47:59 -0400 (Tue, 06 Sep + 2011) | 6 lines authdebug is now disabled by default To enable + this functionaility again set authdebug = yes in iax.conf Review: + https://reviewboard.asterisk.org/r/1414/ ........ + +2011-09-06 16:04 +0000 [r334472-334515] Gregory Nietsky + + * /, apps/app_voicemail.c: Revert r334472 due to properties going + missing + + * /, apps/app_voicemail.c: Merged revisions 334455 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334455 | irroot | 2011-09-06 15:58:56 +0200 + (Tue, 06 Sep 2011) | 18 lines Merged revisions 334453 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334453 | irroot | 2011-09-06 15:48:03 +0200 (Tue, 06 Sep 2011) | + 13 lines Make SQL query in app_voicemail.c portable LIMIT is not + portable. Regression from r312212 (closes issue ASTERISK-18255) + Reported by: Leif Madsen Tested by: Leif Madsen Review: + https://reviewboard.asterisk.org/r/1415/ ........ + ................ + +2011-09-02 21:09 +0000 [r334304-334358] Richard Mudgett + + * /, res/res_musiconhold.c: Merged revisions 334357 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334357 | rmudgett | 2011-09-02 16:08:16 -0500 + (Fri, 02 Sep 2011) | 26 lines Merged revisions 334355 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334355 | rmudgett | 2011-09-02 15:59:49 -0500 (Fri, 02 Sep 2011) + | 19 lines MusicOnHold has extra unref which may lead to memory + corruption and crash. The problem happens when a call is + disconnected and you had started a MOH class that does not use + the files mode. If you define REF_DEBUG and recreate the problem, + it will announce itself with the following warning: Attempt to + unref mohclass 0xb70722e0 (default) when only 1 ref remained, and + class is still in a container! * Fixed moh_alloc() and + moh_release() functions not handling the state->class reference + consistently. (closes issue ASTERISK-18346) Reported by: Mark + Murawski Patches: jira_asterisk_18346_v1.8.patch (license #5621) + patch uploaded by rmudgett Tested by: rmudgett, Mark Murawski + Review: https://reviewboard.asterisk.org/r/1404/ ........ + ................ + + * /, include/asterisk/config.h, main/config.c: Merged revisions + 334297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334297 | rmudgett | 2011-09-02 12:15:08 -0500 + (Fri, 02 Sep 2011) | 46 lines Merged revisions 334296 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334296 | rmudgett | 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011) + | 39 lines Fix potential memory allocation failure crashes in + config.c. * Added required checks to the returned memory + allocation pointers to prevent crashes. * Made + ast_include_rename() create a replacement ast_variable list node + if the new filename is longer than the available space. Fixes + potential crash and memory leak. * Factored out + ast_variable_move() from ast_variable_update() so + ast_include_rename() can also use it when creating a replacement + ast_variable list node. * Made the filename stuffed at the end of + the struct a minimum allocated size in ast_variable_new() in case + ast_include_rename() changes the stored filename. * Constify + struct char pointers pointing to strings stuffed at the end of + the struct for: ast_variable, cache_file_mtime, and + ast_config_map. * Factored out cfmtime_new() to remove inlined + code and allow some struct pointers to become const. * Removed + the list lock from struct cache_file_mtime that was never used. * + Added doxygen comments to several structure elements and better + documented what strings are stuffed at the struct end char array. + * Reworked ast_config_text_file_save() and set_fn() to handle + allocation failure of the include file scratch pad object + tracking blank lines. * Made ast_config_text_file_save() fn[] + declared with PATH_MAX to ensure it is long enough for any + filename with path. Also reduced the number of container fileset + buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review: + https://reviewboard.asterisk.org/r/1378/ ........ + ................ + +2011-09-01 17:41 +0000 [r334231-334236] Tilghman Lesher + + * main/pbx.c, /: Merged revisions 334235 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334235 | tilghman | 2011-09-01 12:39:32 -0500 + (Thu, 01 Sep 2011) | 9 lines Merged revisions 334234 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r334234 | tilghman | 2011-09-01 12:38:33 -0500 (Thu, 01 + Sep 2011) | 2 lines Remove 1.6 compatibility documentation from + 1.8, as it no longer applies. ........ ................ + + * res/res_config_odbc.c, /: Merged revisions 334230 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334230 | tilghman | 2011-09-01 12:30:19 -0500 + (Thu, 01 Sep 2011) | 25 lines Merged revisions 334229 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334229 | tilghman | 2011-09-01 12:28:09 -0500 (Thu, 01 Sep 2011) + | 18 lines Create a local alias for ast_odbc_clear_cache. As a + function pointer, the reference has to be resolved at load time + irrespective of the RTLD_LAZY flag. Creating a local alias solves + this problem, because the structure is initialized with that + local function pointer, while the actual function can remain + lazily linked until runtime. The reason why this is important is + because we lazily load function references during the module + loading process, in order to obtain priority values for each + module, ensuring that modules are loaded in the correct order. + Previous to this change, when this module was initially loaded, + the module loader would emit a symbol resolution error, because + of the above requirement. Closes ASTERISK-18399 (reported by + Mikael Carlsson, fix suggested by Walter Doekes, patch by me) + ........ ................ + +2011-08-31 18:54 +0000 [r334158] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 334157 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334157 | mnicholson | 2011-08-31 13:53:40 -0500 + (Wed, 31 Aug 2011) | 11 lines Merged revisions 334156 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334156 | mnicholson | 2011-08-31 13:50:33 -0500 (Wed, 31 Aug + 2011) | 4 lines Disable T.38 when we get a invite with image + media port set to 0 ASTERISK-17678 ........ ................ + +2011-08-31 18:11 +0000 [r334115] Richard Mudgett + + * channels/chan_sip.c: Optimize chan_sip.c check_rtp_timeout() + function. * Make check_rtp_timeout() remember the values returned + by ast_rtp_instance_get_timeout(), + ast_rtp_instance_get_hold_timeout(), and + ast_rtp_instance_get_keepalive() instead of repeatedly calling + them. (closes issue ASTERISK-18319) Reported by: Rob Gagnon + Patches: issue-18319-trunk-r333066.diff (License #6159) patch + uploaded by Rob Gagnon Review: + https://reviewboard.asterisk.org/r/1377/ + +2011-08-31 16:31 +0000 [r334067] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 334064 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r334064 | mnicholson | 2011-08-31 11:31:00 -0500 (Wed, 31 Aug + 2011) | 4 lines only alter the gateway_timeout when attching the + gateway to a channel ASTERISK-18219 ........ + +2011-08-31 16:02 +0000 [r334011-334014] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 334013 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334013 | rmudgett | 2011-08-31 11:00:49 -0500 + (Wed, 31 Aug 2011) | 30 lines Merged revisions 334012 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334012 | rmudgett | 2011-08-31 10:57:12 -0500 (Wed, 31 Aug 2011) + | 23 lines No DAHDI channel available for conference, user + introduction disabled. The following error will consistently + occur when trying to dial into a MeetMe conference when the + server does not have DAHDI hardware installed: app_meetme.c: No + DAHDI channel available for conference, user introduction + disabled (is chan_dahdi loaded?) While chan_dahdi is loaded + correctly during compilation and install of Asterisk/Dahdi, + including associated modules, etc., a chan_dahdi.conf + configuration file in /etc/asterisk is not created by FreePBX if + hardware does not exist, causing MeetMe to be unable to open a + DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo + channel when there is no chan_dahdi.conf file to load. (closes + issue ASTERISK-17398) Reported by: Preston Edwards Patches: + jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by + rmudgett Tested by: rmudgett ........ ................ + + * main/channel.c, /, channels/chan_agent.c: Merged revisions 334010 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334010 | rmudgett | 2011-08-31 10:23:11 -0500 + (Wed, 31 Aug 2011) | 50 lines Merged revisions 334009 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334009 | rmudgett | 2011-08-31 10:20:31 -0500 (Wed, 31 Aug 2011) + | 43 lines Call pickup race leaves orphaned channels or crashes. + Multiple users attempting to pickup a call that has been forked + to multiple extensions either crashes or fails a masquerade with + a "bad things may happen" message. This is the scenario that is + causing all the grief: 1) Pickup target is selected 2) target is + marked as being picked up in ast_do_pickup() 3) target is + unlocked by ast_do_pickup() 4) app dial or queue gets a chance to + hang up losing calls and calls ast_hangup() on target 5) SINCE A + MASQUERADE HAS NOT BEEN SETUP YET BY ast_do_pickup() with + ast_channel_masquerade(), ast_hangup() completes successfully and + the channel is no longer in the channels container. 6) + ast_do_pickup() then calls ast_channel_masquerade() to schedule + the masquerade on the dead channel. 7) ast_do_pickup() then calls + ast_do_masquerade() on the dead channel 8) bad things happen + while doing the masquerade and in the process ast_do_masquerade() + puts the dead channel back into the channels container 9) The + "orphaned" channel is visible in the channels list if a crash + does not happen. This patch does the following: * Made + ast_hangup() set AST_FLAG_ZOMBIE on a successfully hung-up + channel and not release the channel lock until that has happened. + * Made __ast_channel_masquerade() not setup a masquerade if + either channel has AST_FLAG_ZOMBIE set. * Fix chan_agent misuse + of AST_FLAG_ZOMBIE since it would no longer work. (closes issue + ASTERISK-18222) Reported by: Alec Davis Tested by: rmudgett, Alec + Davis, irroot, Karsten Wemheuer (closes issue ASTERISK-18273) + Reported by: Karsten Wemheuer Tested by: rmudgett, Alec Davis, + irroot, Karsten Wemheuer Review: + https://reviewboard.asterisk.org/r/1400/ ........ + ................ + +2011-08-31 15:20 +0000 [r334008] Kinsey Moore + + * /, channels/chan_sip.c: Merged revisions 334007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r334007 | kmoore | 2011-08-31 10:19:30 -0500 + (Wed, 31 Aug 2011) | 14 lines Merged revisions 334006 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r334006 | kmoore | 2011-08-31 10:18:37 -0500 (Wed, 31 Aug 2011) | + 7 lines Correct an AMI protocol violation with SIPshowpeer The + response of SIPshowpeer ends with "\r\n\r\n". Since other + commands are ended by using \r\n this confuses any interfacing + script. (closes issue ASTERISK-17486) ........ ................ + +2011-08-30 22:16 +0000 [r333963] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, /, + addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/oochannels.c, + addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c: Merged + revisions 333961-333962 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r333961 | may | 2011-08-31 01:21:53 +0400 (Wed, + 31 Aug 2011) | 11 lines Merged revisions 333947 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r333947 | may | 2011-08-31 01:16:30 +0400 (Wed, 31 Aug 2011) | 5 + lines cleanups in ACF/ARJ GK replies processing fixed long (24 + sec) pause if acf/arj proccessed before ast_cond_wait called to + wait this ........ ................ r333962 | may | 2011-08-31 + 01:53:42 +0400 (Wed, 31 Aug 2011) | 3 lines security fix. really + drop call if signalling addr is not same as socket addr + ................ + +2011-08-30 14:03 +0000 [r333896] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 333895 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r333895 | mnicholson | 2011-08-30 09:01:31 -0500 (Tue, 30 Aug + 2011) | 6 lines Replaced FAXOPT(gwtimeout) with a second + parameter to FAXOPT(gateway). Patch by: irroot Review: + https://reviewboard.asterisk.org/r/1385/ ASTERISK-18219 ........ + +2011-08-29 21:43 +0000 [r333838] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 333837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r333837 | twilson | 2011-08-29 16:41:13 -0500 + (Mon, 29 Aug 2011) | 22 lines Merged revisions 333836 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r333836 | twilson | 2011-08-29 16:38:31 -0500 (Mon, 29 Aug 2011) + | 15 lines Refresh peer address if DNS unavailable at peer + creation If Asterisk starts and no DNS is available, outbound + registrations will fail indefinitely. This patch copies the + address from the sip_registry struct, which will be updated, to + the peer->addr when necessary. If dnsmgr is enabled, the + registration fails without the patch because even though the + address on the registry is updated via dnsmgr, the address is + just copied on the first try. Since we use ast_sockaddr_copy, + dnsmgr can't update the address that is copied to the sip_pvt or + peers. Closes issue ASTERISK-18000 Review: + https://reviewboard.asterisk.org/r/1335/ ........ + ................ + +2011-08-29 21:17 +0000 [r333789] Richard Mudgett + + * /, include/asterisk/channel.h, addons/chan_mobile.c: Merged + revisions 333786 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r333786 | rmudgett | 2011-08-29 16:12:29 -0500 + (Mon, 29 Aug 2011) | 13 lines Merged revisions 333784-333785 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r333784 | rmudgett | 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011) + | 2 lines Fix deadlock potential of + chan_mobile.c:mbl_ast_hangup(). ........ r333785 | rmudgett | + 2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) | 1 line Add some do + not hold locks notes to channel.h ........ ................ + +2011-08-29 18:28 +0000 [r333736] Matthew Nicholson + + * /, res/res_fax_spandsp.c: Merged revisions 333716 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r333716 | mnicholson | 2011-08-29 13:22:58 -0500 (Mon, 29 Aug + 2011) | 5 lines It is possible for the gateway to be attached + when the channel is still negotiating T.38. This change handles + that case. ASTERISK-18329 ........ + +2011-08-29 17:31 +0000 [r333689] Terry Wilson + + * main/channel.c, /, CHANGES: Merged revisions 333681 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r333681 | twilson | 2011-08-29 12:28:59 -0500 (Mon, 29 Aug 2011) + | 7 lines Use realtime text when it is negotiated This patch make + use of wirte_text() realtime text instead of send_text() if T.140 + is in native formats. ASTERISK-17937 Review: + https://reviewboard.asterisk.org/r/1356/ ........ + +2011-08-29 17:14 +0000 [r333632] Matthew Jordan + + * apps/app_voicemail.c: Merged revisions 333631 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r333631 | mjordan | 2011-08-29 12:12:55 -0500 + (Mon, 29 Aug 2011) | 9 lines Merged revisions 333630 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r333630 | mjordan | 2011-08-29 12:11:15 -0500 (Mon, 29 + Aug 2011) | 1 line Fixed improperly formatted TestEvent AMI + message in app_voicemail ........ ................ + +2011-08-29 15:58 +0000 [r333571] Jonathan Rose + + * /, res/res_jabber.c: Merged revisions 333570 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r333570 | jrose | 2011-08-29 10:56:56 -0500 + (Mon, 29 Aug 2011) | 11 lines Merged revisions 333569 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r333569 | jrose | 2011-08-29 10:55:34 -0500 (Mon, 29 Aug 2011) | + 4 lines Accidental use of variable client->status instead of + client->state in from ASTERISK-18078 (issue ASTERISK-18078) + ........ ................ + +2011-08-28 09:57 +0000 [r333509] Tzafrir Cohen + + * channels/chan_vpb.cc: chan_vpb: remove unused variables (gcc4.6) + GCC 4.6 detects variables that get assined to, but never used + later. Also removes some remmed-out lines that become invalid. + (closes issue ASTERISK-18336) Signed-off-by: Tzafrir Cohen + (License #5035) , + +2011-08-26 16:38 +0000 [r333428] Jonathan Rose + + * /, res/res_jabber.c: Merged revisions 333410 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r333410 | jrose | 2011-08-26 11:28:03 -0500 + (Fri, 26 Aug 2011) | 19 lines Merged revisions 333378 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r333378 | jrose | 2011-08-26 11:19:07 -0500 (Fri, 26 Aug 2011) | + 13 lines [patch] Buddies are always auto-registered when + processing the roster Reporter said autoregister flag was ignored + for registering 'buddies' which had a subscription to us. + Verified that this was the case and observed how the patch + addressed this and made sure it didn't break anything. (closes + issue ASTERISK-14233) Reported by: Simon Arlott Patches: + asterisk-0015229.patch (license #5756) patch uploaded by Simon + Arlott Tested by: Jonathan Rose ........ ................ + +2011-08-26 16:12 +0000 [r333371] Matthew Jordan + + * /, apps/app_voicemail.c: Merged revisions 333370 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r333370 | mjordan | 2011-08-26 10:58:37 -0500 + (Fri, 26 Aug 2011) | 26 lines Merged revisions 333339 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r333339 | mjordan | 2011-08-26 08:36:36 -0500 (Fri, 26 Aug 2011) + | 20 lines Bug fixes for voicemail user emailsubject / emailbody. + This code change fixes a few issues with the voicemail user + override of emailbody and emailsubject, including escaping the + strings, potential memory leaks, and not overriding the voicemail + defaults. Revision 325877 fixed this for ASTERISK-16795, but did + not fix it for ASTERISK-16781. A subsequent check-in prevented + 325877 from being applied to 10. This check-in resolves both + issues, and applies the changes to 1.8, 10, and trunk. (closes + issue ASTERISK-16781) Reported by: Sebastien Couture Tested by: + mjordan (closes issue ASTERISK-16795) Reported by: mdeneen Tested + by: mjordan Review: https://reviewboard.asterisk.org/r/1374 + ........ ................ + +2011-08-25 19:13 +0000 [r333276] Jonathan Rose + + * /, res/res_jabber.c: Merged revisions 333266 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r333266 | jrose | 2011-08-25 14:00:05 -0500 + (Thu, 25 Aug 2011) | 20 lines Merged revisions 333265 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r333265 | jrose | 2011-08-25 13:47:42 -0500 (Thu, 25 Aug 2011) | + 14 lines Segfault when publishing device states via XMPP and not + connected When using publishing device state with res_jabber, + Asterisk will attempt to send a device state using the + unconnected client using iks_send_raw and crash. This patch + checks the validity of the connection before attempting to send + the device state. (closes issue ASTERISK-18078) Reported by: + Michael L. Young Patches: + res_jabber-segfault-pubsub-not-connected2.patch (license #5026) + patch uploaded by Michael L. Young Tested by: Jonathan Rose + ........ ................ + +2011-08-25 19:01 +0000 [r333159-333269] Jason Parker + + * Makefile, /: Merged revisions 333268 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r333268 | qwell | 2011-08-25 14:01:18 -0500 + (Thu, 25 Aug 2011) | 9 lines Merged revisions 333267 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r333267 | qwell | 2011-08-25 14:00:55 -0500 (Thu, 25 Aug + 2011) | 2 lines Fix for DESTDIR spaces patch. ........ + ................ + + * Makefile, build_tools/mkpkgconfig, /, configure, configure.ac, + makeopts.in, sounds/Makefile: Merged revisions 333203 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r333203 | qwell | 2011-08-25 10:29:56 -0500 + (Thu, 25 Aug 2011) | 15 lines Merged revisions 333201 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r333201 | qwell | 2011-08-25 10:27:06 -0500 (Thu, 25 Aug 2011) | + 8 lines Fix installation into directories containing spaces. This + also vastly simplifies the logic in sounds/Makefile (Closes issue + ASTERISK-18290) Reported by: Paul Belanger Review: + https://reviewboard.asterisk.org/r/1379/ ........ + ................ + + * channels/chan_local.c: Fix typo from r333070 + +2011-08-24 16:52 +0000 [r333117] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 333115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r333115 | mnicholson | 2011-08-24 11:51:42 -0500 (Wed, 24 Aug + 2011) | 4 lines Changed the "timeout" option to "gwtimeout". + ASTERISK-18219 ........ + +2011-08-24 09:17 +0000 [r333070-333075] Olle Johansson + + * channels/chan_local.c: Formatting changes - Removing some red + white space and adding some curly brackets. + + * CHANGES: Add documentation for new manager event in chan_local + AST-17623 + + * channels/chan_local.c: Add manager event for local channel + semi-bridge (issue AST-17623) Review: + https://reviewboard.asterisk.org/r/1154 + +2011-08-23 18:17 +0000 [r332881-333014] Richard Mudgett + + * /, apps/app_queue.c: Merged revisions 333011 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r333011 | rmudgett | 2011-08-23 13:15:49 -0500 + (Tue, 23 Aug 2011) | 19 lines Merged revisions 333010 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r333010 | rmudgett | 2011-08-23 13:14:01 -0500 (Tue, 23 Aug 2011) + | 12 lines Memory Leak in app_queue The patch that was committed + in the 1.6.x versions of Asterisk for ASTERISK-15862 actually + fixed two issues. One was not applicable to 1.8 but the other is. + queue_leak.patch fixes the portion applicable to 1.8. (closes + issue ASTERISK-18265) Reported by: Fred Schroeder Patches: + queue_leak.patch (license #5049) patch uploaded by mmichelson + Tested by: Thomas Arimont ........ ................ + + * /, main/config.c: Merged revisions 332940 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332940 | rmudgett | 2011-08-22 16:23:40 -0500 + (Mon, 22 Aug 2011) | 14 lines Merged revisions 332939 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332939 | rmudgett | 2011-08-22 16:22:24 -0500 (Mon, 22 Aug 2011) + | 7 lines Minor code optimizations. * Simplify + ast_category_browse() logic for easier understanding. * Remove + dead code in ast_variable_delete() and simplify some of its + logic. ........ ................ + + * /, apps/app_queue.c: Merged revisions 332875,332878 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332875 | rmudgett | 2011-08-22 14:41:03 -0500 + (Mon, 22 Aug 2011) | 1 line Fix merge property. ................ + r332878 | rmudgett | 2011-08-22 14:46:25 -0500 (Mon, 22 Aug 2011) + | 25 lines Merged revisions 332874 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332874 | rmudgett | 2011-08-22 14:32:19 -0500 (Mon, 22 Aug 2011) + | 18 lines Reference leaks in app_queue. * Fixed + load_realtime_queue() leaking a queue reference when it + overwrites q when processing a realtime queue. (issue + ASTERISK-18265) * Make join_queue() unreference the queue + returned by load_realtime_queue() when it is done with the + pointer. The load_realtime_queue() returns a reference to the + just loaded realtime queue. * Fixed queues container reference + leak in queues_data_provider_get(). * queue_unref() should not + return q that was just unreferenced. * Made logic in + __queues_show() and queues_data_provider_get() when calling + load_realtime_queue() easier to understand. ........ + ................ + +2011-08-22 19:56 +0000 [r332880] Paul Belanger + + * /, channels/chan_gtalk.c: Merged revisions 332877 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332877 | pabelanger | 2011-08-22 15:43:33 -0400 + (Mon, 22 Aug 2011) | 13 lines Merged revisions 332876 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332876 | pabelanger | 2011-08-22 15:41:24 -0400 (Mon, 22 Aug + 2011) | 6 lines Revert previous commit It seems google is still + making changes to the protocol. (issue ASTERISK-18301) ........ + ................ + +2011-08-22 19:52 +0000 [r332879] Richard Mudgett + + * /: Fix merge 10 branch merge properties. + +2011-08-22 19:19 +0000 [r332844] Matthew Jordan + + * include/asterisk/test.h, main/manager.c, /, main/file.c, + main/test.c, main/app.c, configs/manager.conf.sample, + include/asterisk/manager.h, apps/app_voicemail.c: Merged + revisions 332817 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332817 | mjordan | 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) + | 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This + update adds a new AMI event, TestEvent, which is enabled when the + TEST_FRAMEWORK compiler flag is defined. It also adds initial + usage of this event to app_voicemail. The TestEvent AMI event is + used extensively by the voicemail tests in the Asterisk Test + Suite. ........ + +2011-08-22 18:33 +0000 [r332762-332831] Richard Mudgett + + * res/res_config_pgsql.c, res/res_config_odbc.c, /: Merged + revisions 332830 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332830 | rmudgett | 2011-08-22 13:32:09 -0500 + (Mon, 22 Aug 2011) | 15 lines Merged revisions 332816 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332816 | rmudgett | 2011-08-22 13:14:59 -0500 (Mon, 22 Aug 2011) + | 8 lines Memory leaks in realtime_multi_xxx() when database + access returns error. * Fix realtime_multi_pgsql() configuration + memory leak when the database access returns an error. * Fix + realtime_multi_odbc() configuration category use after free when + the database access returns an error. ........ ................ + + * /, main/config.c: Merged revisions 332761 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332761 | rmudgett | 2011-08-22 12:05:35 -0500 + (Mon, 22 Aug 2011) | 22 lines Merged revisions 332759 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332759 | rmudgett | 2011-08-22 12:00:03 -0500 (Mon, 22 Aug 2011) + | 15 lines Memory leak reading realtime database variable list. + Calling ast_load_realtime() can leak the last list node if the + read list only contains empty variable value items. * Fixed list + filter loop in ast_load_realtime() to delete the list node + immediately instead of the next time through the loop. The next + time through the loop may not happen if the node to delete is the + last in the list. (issue ASTERISK-18277) (issue ASTERISK-18265) + Patches: jira_asterisk_18265_v1.8_config.patch (license #5621) + patch uploaded by rmudgett ........ ................ + +2011-08-22 17:05 +0000 [r332760] Jonathan Rose + + * main/cdr.c, main/pbx.c, configs/cdr.conf.sample, + include/asterisk/cdr.h, CHANGES: Add option for logging congested + calls as CONGESTION instead of NO_ANSWER in CDR This patch adds a + CDR option to cdr.conf that will allow CDR files to log calls + ending with congestion in a way that is unique from other + unanswered calls. (closes issue ASTERISK-14842) Reported by: Alec + Davis Patches: cdr_congestion.diff.txt (License #5546) patch + uploaded by Alec Davis + +2011-08-22 16:31 +0000 [r332757] Matthew Nicholson + + * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions + 332756 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r332756 | mnicholson | 2011-08-22 11:29:45 -0500 (Mon, 22 Aug + 2011) | 4 lines add a way to disable and/or modify the gateway + timeout ASTERISK-18219 ........ + +2011-08-21 14:34 +0000 [r332701] Paul Belanger + + * /, channels/chan_gtalk.c: Merged revisions 332700 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332700 | pabelanger | 2011-08-21 10:33:23 -0400 + (Sun, 21 Aug 2011) | 12 lines Merged revisions 332699 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332699 | pabelanger | 2011-08-21 10:31:31 -0400 (Sun, 21 Aug + 2011) | 5 lines Fix outgoing calls in chan_gtalk (closes issue + ASTERISK-18301) Reported by: az1324 ........ ................ + +2011-08-19 20:00 +0000 [r332655] Kinsey Moore + + * /, apps/app_confbridge.c: Merged revisions 332654 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r332654 | kmoore | 2011-08-19 14:59:34 -0500 (Fri, 19 Aug 2011) | + 8 lines Make CONFBRIDGE_INFO behave more nicely CONFBRIDGE_INFO + doesn't behave as well in edge cases as MEETME_INFO. With this + patch, CONFBRIDGE_INFO should behave in a much more reasonable + manner when presented with invalid conferences and keywords. + Review: https://reviewboard.asterisk.org/r/1359/ ........ + +2011-08-19 17:24 +0000 [r332615] Richard Mudgett + + * res/res_config_ldap.c: Fix infinite loop releasing the same + memory in ldap_loadentry(). * Fixed memory leak of vars in + ldap_loadentry(). * Fixed potential NULL ptr dereference of vars + in ldap_loadentry(). + +2011-08-18 21:39 +0000 [r332561] Terry Wilson + + * main/netsock2.c, /: Merged revisions 332560 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332560 | twilson | 2011-08-18 16:34:04 -0500 + (Thu, 18 Aug 2011) | 12 lines Merged revisions 332559 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332559 | twilson | 2011-08-18 16:26:01 -0500 (Thu, 18 Aug 2011) + | 5 lines Fix possible error on stringification of IPv4-mapped + addrs The FreeBSD netsock2 test has been failing for a while. We + were pasing sa->len to getnameinfo instead of sa_tmp->len. + ASTERISK-18289 ........ ................ + +2011-08-18 19:30 +0000 [r332505] Kinsey Moore + + * channels/chan_dahdi.c, /: Merged revisions 332504 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332504 | kmoore | 2011-08-18 14:29:15 -0500 + (Thu, 18 Aug 2011) | 15 lines Merged revisions 332503 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332503 | kmoore | 2011-08-18 14:28:00 -0500 (Thu, 18 Aug 2011) | + 8 lines CRC4 in "dahdi show status" gives wrong impression to T1 + users Change CRC4 to CRC in the output of "dahdi show status" so + that it can apply in more situations without confusing users, + especially since T1 lines use CRC6 instead of CRC4. (closes issue + AST-471) ........ ................ + +2011-08-18 14:49 +0000 [r332388-332448] Tilghman Lesher + + * build_tools/cflags.xml, build_tools/cflags-devmode.xml, /: Merged + revisions 332447 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332447 | tilghman | 2011-08-18 09:48:40 -0500 + (Thu, 18 Aug 2011) | 9 lines Merged revisions 332446 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r332446 | tilghman | 2011-08-18 09:46:54 -0500 (Thu, 18 + Aug 2011) | 2 lines Move BETTER_BACKTRACES out of development + mode, as it's useful when DEBUG_THREADS is enabled. ........ + ................ + + * Makefile, agi/Makefile, utils/Makefile, /, configure, + include/asterisk/autoconfig.h.in, configure.ac, + Makefile.moddir_rules, makeopts.in, sounds/Makefile: Merged + revisions 332369 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332369 | tilghman | 2011-08-17 14:24:59 -0500 + (Wed, 17 Aug 2011) | 17 lines Merged revisions 332355 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) + | 10 lines Re-add support for spaces in pathnames, including now + spaces in DESTDIR. This was initially added to 1.8 prior to + release, primarily to support the standard paths on Mac OS X, but + was partially reverted recently in Subversion, due to the lack of + support for spaces in DESTDIR. This commit restores support for + the standard paths on Mac OS X, and also includes support for + spaces in DESTDIR. (closes issue ASTERISK-18290) Reported by: + pabelanger Review: https://reviewboard.asterisk.org/r/1326/ + ........ ................ + +2011-08-17 18:31 +0000 [r332337] Terry Wilson + + * /, res/res_timing_timerfd.c: Merged revisions 332321 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332321 | twilson | 2011-08-17 13:09:49 -0500 + (Wed, 17 Aug 2011) | 17 lines Merged revisions 332320 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332320 | twilson | 2011-08-17 12:35:27 -0500 (Wed, 17 Aug 2011) + | 10 lines Don't read from a disarmed or invalid timerfd Numerous + isues have been reported for deadlocks that are caused by a + blocking read in res_timing_timerfd on a file descriptor that + will never be written to. This patch adds some checks to make + sure that the timerfd is both valid and armed before calling + read(). Should fix: ASTERISK-18142, ASTERISK-18166, + ASTERISK-18197, AST-486, AST-495, AST-507 and possibly others. + Review: https://reviewboard.asterisk.org/r/1361/ ........ + ................ + +2011-08-17 16:18 +0000 [r332270] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, /, configure, + include/asterisk/autoconfig.h.in, configure.ac, + channels/sig_pri.c: Merged revisions 332265 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332265 | rmudgett | 2011-08-17 11:01:29 -0500 + (Wed, 17 Aug 2011) | 33 lines Merged revisions 332264 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) + | 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with + HA8, HB8, and B410P cards. France Telecom brings layer 2 and + layer 1 down on BRI lines when the line is idle. When layer 1 + goes down Asterisk cannot make outgoing calls and the HA8 and HB8 + cards also get IRQ misses. The inability to make outgoing calls + is because the line is in red alarm and Asterisk will not make + calls over a line it considers unavailable. The IRQ misses for + the HA8 and HB8 card are because the hardware is switching clock + sources from the line which just brought layer 1 down to internal + timing. There is a DAHDI option for the B410P card to not tell + Asterisk that layer 1 went down so Asterisk will allow outgoing + calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI + option for the HA8 and HB8 cards: "modprobe wctdm24xxp + bri_teignored=1". Unfortunately that will not clear up the IRQ + misses when the telco brings layer 1 down. * Add layer 2 + persistence option to customize the layer 2 behavior on BRI PTMP + lines. The new option has three settings: 1) Use libpri default + layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when + the peer brings it down. 3) Leave layer 2 down when the peer + brings it down. Layer 2 will be brought up as needed for outgoing + calls. JIRA AST-598 ........ ................ + +2011-08-16 20:15 +0000 [r332178] Paul Belanger + + * tests/test_substitution.c, tests/test_heap.c, /, + tests/test_expr.c, tests/test_ast_format_str_reduce.c, + tests/test_logger.c, tests/test_gosub.c, tests/test_app.c, + tests/test_dlinklists.c, tests/test_event.c, tests/test_db.c, + tests/test_linkedlists.c, tests/test_sched.c, + tests/test_netsock2.c, tests/test_strings.c, tests/test_pbx.c, + tests/test_func_file.c, tests/test_security_events.c, + tests/test_stringfields.c, tests/test_time.c, tests/test_skel.c, + tests/test_acl.c, tests/test_locale.c, tests/test_utils.c, + tests/test_devicestate.c, tests/test_aoc.c, tests/test_astobj2.c, + tests/test_poll.c, tests/test_amihooks.c: Merged revisions 332177 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332177 | pabelanger | 2011-08-16 16:11:49 -0400 + (Tue, 16 Aug 2011) | 11 lines Merged revisions 332176 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332176 | pabelanger | 2011-08-16 16:10:13 -0400 (Tue, 16 Aug + 2011) | 4 lines Flag test modules as 'core' Review: + https://reviewboard.asterisk.org/r/1369/ ........ + ................ + +2011-08-16 17:53 +0000 [r332120] Jonathan Rose + + * /, channels/chan_sip.c: Merged revisions 332119 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332119 | jrose | 2011-08-16 12:45:38 -0500 + (Tue, 16 Aug 2011) | 23 lines Merged revisions 332118 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332118 | jrose | 2011-08-16 12:38:19 -0500 (Tue, 16 Aug 2011) | + 16 lines ASTERISK-18067 ASTERISK-15479 - White Space affects + mailbox value, multiple MWI subs Before, having multiple + subscriptions to mailboxes on a sip peer set via the mailbox + setting in sip.conf would only result in updates being sent on + whichever mailbox triggered the mwi event. Now all of them get + counted regardless. Also fixes a bug involving parsing of the + mailbox option in sip.conf so that trailing and leading spaces + before/after commas are trimmed. (closes issue ASTERISK-18067) + Reported by: aragon (closes issue ASTERISK-15479) Reported by: + Ben Winslow Patches: + chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) + patch uploaded by Ben Winslow ........ ................ + +2011-08-16 17:23 +0000 [r332117] Richard Mudgett + + * /, main/features.c, CHANGES, configs/features.conf.sample, + main/asterisk.c: Merged revisions 332101 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332101 | rmudgett | 2011-08-16 12:17:28 -0500 + (Tue, 16 Aug 2011) | 140 lines Merged revisions 332100 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332100 | rmudgett | 2011-08-16 11:31:36 -0500 (Tue, 16 Aug 2011) + | 133 lines Fix multiple parking issues. JIRA ASTERISK-17183 + Multi-parkinglot directs calls to wrong parkinglot. JIRA + ASTERISK-17870 Cannot retrieve parked calls. JIRA ASTERISK-17430 + ParkedCall() with no extension should pickup first available call + and does not. JIRA AST-576 Issues with parking lots * Removed + searching for parking lots by extension. Parking lots can only be + found by the parking lot name since parking lot access extensions + and spaces are not guaranteed to be unique. * Added + parking_lot_name option to the Park and ParkedCall applications. + Updated documentation for Park and ParkedCall applications. * Add + parkext_exclusive configuration option to make parking entry + extensions specify which parking lot they access. (closes issue + ASTERISK-17183) Reported by: David Cabrejos Tested by: rmudgett, + David Cabrejos (closes issue ASTERISK-17870) Reported by: Remi + Quezada (closes issue ASTERISK-17430) Reported by: Philippe + Lindheimer JIRA ASTERISK-17452 Parking_offset not used JIRA + AST-624 'next' setting for findslot does nothing * Reimplemented + since findslot feature option broken by -r114655. (closes issue + ASTERISK-17452) Reported by: David Woolley Tested by: rmudgett + JIRA ASTERISK-15792 Dialplan continues execution after transfer + to park. This happens for DTMF attended transfer, DTMF blind + transfer, and DTMF one-touch-parking if the party initiating + these features also initiated the call. * Fixed the return code + from the affected builtin features when parking a call. (closes + issue ASTERISK-15792) Reported by: Mat Murdock Tested by: + rmudgett, twilson JIRA AST-607 The courtesytone is not playing to + the expected call when picking up a parked call. This is mostly a + documentation problem. However, the option is not reset to the + default when features.conf is reloaded. * Updated + features.conf.sample documentation for courtesytone and + parkedplay options. * Reset the parkedplay option to default when + features.conf is reloaded. JIRA AST-615 AMI Park action followed + by features reload results in orphaned channels in parking lot. * + Reloading features.conf will not touch parking lots that have + calls still parked in them. Reload again at a later time. Misc + additional fixes: * Added unit test for parking lot dialplan + usage checking. * Made update connected line when a parked call + is retrieved from a parking lot. * Made retrieved parked call + stop ringing or MOH depending upon how the call was waiting in + the parking lot. * Made CLI "features show" indicate if the + parking lot is enabled for use. * Added PARKINGDYNEXTEN channel + variable to allow dynamic parking lots to specify the parking lot + access extension. * Made AMI ParkedCalls action ParkedCall events + have a Parkinglot header. * Made AMI ParkedCalls action + ParkedCallsComplete event have a Total header. * Fixed potential + deadlock from AMI Park action holding channel locks while calling + masq_park_call(). * Fixed several places where ast_strdupa() were + used inside of loops. (Mostly fixed by refactoring the loop body + into its own function.) * Fixed copy_parkinglot() copying too + much from the source parking lot. Extracted the parking lot + configuration settings into struct parkinglot_cfg. * Refactored + courtesytone playing code to put the channel not playing the tone + in autoservice. * Fix when pbx-parkingfailed is played that the + other channel is put in autoservice if it exists. * Fixed + parkinglot reference leak in parked_call_exec() error paths. * + Fixed parkinglot_unref() use of parkinglot after it was unreffed. + * Made destroy the struct ast_parkinglot parkings lock when done. + * Refactored the features.conf parking lot configuration code to + eliminate redundancy. * Fixed feature reload to better protect + parking lots. * Fixed parking lot container reference leak in + handle_parkedcalls(). * Fixed the total count in + handle_parkedcalls(). Review: + https://reviewboard.asterisk.org/r/1358/ ........ + ................ + +2011-08-16 15:21 +0000 [r332028-332044] Matthew Nicholson + + * /, channels/sip/include/sip.h: Merged revisions 332042 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ........ r332042 | mnicholson | 2011-08-16 10:20:48 -0500 (Tue, + 16 Aug 2011) | 2 lines fix a code comment AST-580 ........ + + * /, UPGRADE.txt, CHANGES: Merged revisions 332029 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r332029 | mnicholson | 2011-08-16 10:17:16 -0500 (Tue, 16 Aug + 2011) | 2 lines Moved notes about 'storesipcause' to UPGRADE.txt + from CHANGES AST-580 ........ + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged + revisions 332027 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332027 | mnicholson | 2011-08-16 10:08:40 -0500 + (Tue, 16 Aug 2011) | 9 lines Merged revisions 332026 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r332026 | mnicholson | 2011-08-16 10:06:31 -0500 (Tue, + 16 Aug 2011) | 2 lines use DEFAULT_STORE_SIP_CAUSE to set the + default value for the 'storesipcause' option AST-580 ........ + ................ + +2011-08-16 14:47 +0000 [r332024] Olle Johansson + + * channels/chan_local.c: Formatting changes while working with + DTMF... + +2011-08-16 14:41 +0000 [r332023] Matthew Nicholson + + * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged + revisions 332022 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r332022 | mnicholson | 2011-08-16 09:40:37 -0500 + (Tue, 16 Aug 2011) | 16 lines In 10 and trunk this option is + disabled by default. Merged revisions 332021 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r332021 | mnicholson | 2011-08-16 09:20:43 -0500 (Tue, 16 Aug + 2011) | 7 lines Added the 'storesipcause' option to sip.conf to + allow the user to disable the setting of HASH(SIP_CAUSE,) on the channel. Having chan_sip set HASH(SIP_CAUSE,) on the channel carries a significant performance penalty + because of the usage of the MASTER_CHANNEL() dialplan function. + AST-580 ........ ................ + +2011-08-15 17:36 +0000 [r331957] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 331956 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331956 | rmudgett | 2011-08-15 12:35:03 -0500 + (Mon, 15 Aug 2011) | 20 lines Merged revisions 331955 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331955 | rmudgett | 2011-08-15 12:24:08 -0500 (Mon, 15 Aug 2011) + | 13 lines Fix some minor chan_dahdi config load issues. * + Address chan_dahdi.conf dahdichan option todo item about needing + line number. * Make ignore_failed_channels option also apply to + dahdichan option. * Don't attempt to create a default pseudo + channel if the chan_dahdi.conf channel/channels option is not + allowed. * Add a similar check for dahdichan in normal + chan_dahdi.conf sections as is done in users.conf. ........ + ................ + +2011-08-15 15:24 +0000 [r331903] Paul Belanger + + * main/rtp_engine.c, /: Merged revisions 331894 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331894 | pabelanger | 2011-08-15 11:22:45 -0400 + (Mon, 15 Aug 2011) | 12 lines Merged revisions 331886 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331886 | pabelanger | 2011-08-15 11:21:16 -0400 (Mon, 15 Aug + 2011) | 5 lines Fix noisy message when briding channels (closes + issue ASTERISK-18270) Reported by: Federico Alves ........ + ................ + +2011-08-15 15:15 +0000 [r331869] David Vossel + + * /, channels/chan_sip.c: Merged revisions 331868 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331868 | dvossel | 2011-08-15 10:14:13 -0500 + (Mon, 15 Aug 2011) | 12 lines Merged revisions 331867 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331867 | dvossel | 2011-08-15 10:12:16 -0500 (Mon, 15 Aug 2011) + | 6 lines Fixes locking inversion issues present in the handling + of the sip REFER method. (closes issue ASTERISK-18082) Reported + by: James Van Vleet ........ ................ + +2011-08-15 13:27 +0000 [r331830] Olle Johansson + + * channels/chan_sip.c: Formatting guideline fixes + +2011-08-12 19:06 +0000 [r331776] Matthew Nicholson + + * /, apps/app_queue.c: Merged revisions 331775 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331775 | mnicholson | 2011-08-12 14:03:31 -0500 + (Fri, 12 Aug 2011) | 17 lines Merged revisions 331774 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331774 | mnicholson | 2011-08-12 14:01:27 -0500 (Fri, 12 Aug + 2011) | 11 lines Unlock the channel before calling update_queue. + Holding the channel lock when calling update_queue which attempts + to lock the queue lock can cause a deadlock. This deadlock + involves the following chain: 1. hold chan lock -> wait queue + lock 2. hold queue lock -> wait agent list lock 3. hold agent + list lock -> wait chan list lock 4. hold chan list lock -> wait + chan lock ........ ................ + +2011-08-12 19:01 +0000 [r331773] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 331772 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331772 | rmudgett | 2011-08-12 13:59:45 -0500 + (Fri, 12 Aug 2011) | 15 lines Merged revisions 331771 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331771 | rmudgett | 2011-08-12 13:58:40 -0500 (Fri, 12 Aug 2011) + | 8 lines Suppress warning message when using DAHDITransfer or + DAHDIHangup. * The fake event should only be processed by the + channel that currently owns the private and not the associated + call waiting or 3-way channel. JIRA AST-620 JIRA SWP-3616 + ........ ................ + +2011-08-12 18:03 +0000 [r331717] Jonathan Rose + + * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331644 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331644 | jrose | 2011-08-12 11:18:57 -0500 + (Fri, 12 Aug 2011) | 9 lines Merged revisions 331635 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r331635 | jrose | 2011-08-12 10:49:17 -0500 (Fri, 12 Aug + 2011) | 1 line Fixes 32bit compilation warnings brought on by + 331634 in app_dial and app_meetme ........ ................ + +2011-08-12 17:56 +0000 [r331716] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 331715 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331715 | rmudgett | 2011-08-12 12:54:47 -0500 + (Fri, 12 Aug 2011) | 29 lines Merged revisions 331714 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331714 | rmudgett | 2011-08-12 12:47:57 -0500 (Fri, 12 Aug 2011) + | 22 lines AMI actions DAHDIHangup and DAHDITransfer have no + effect. The AMI actions DAHDIHangup and DAHDITransfer have no + effect on a DAHDI channel. These two AMI actions are highly + specialized to analog channels and appear to make the channel + behave like a jack port for headsets. * Made the faked DAHDI + event get processed before a normal media stream read in + dahdi_read() instead of trying to trigger an exception read by + setting the AST_FLAG_EXCEPTION flag. Apparently a change was made + long ago that changed how AST_FLAG_EXCEPTION is processed in the + core. Unfortunately, the faked DAHDI events no longer worked when + that happened. * Updated the DAHDI AMI action documentation for + the following actions: DAHDITransfer, DAHDIHangup, + DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff, DAHDIShowChannels, and + DAHDIRestart. * Made use sscanf() instead of atoi() for better + error checking of the DAHDIChannel header string. JIRA AST-620 + JIRA SWP-3616 ........ ................ + +2011-08-12 16:32 +0000 [r331660] Terry Wilson + + * /, tests/test_netsock2.c: Merged revisions 331659 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331659 | twilson | 2011-08-12 11:31:21 -0500 + (Fri, 12 Aug 2011) | 11 lines Merged revisions 331658 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331658 | twilson | 2011-08-12 11:30:26 -0500 (Fri, 12 Aug 2011) + | 4 lines Fix netsock2 multiple zero-expansion test Remove + erroneous single bracket. ........ ................ + +2011-08-12 16:22 +0000 [r331657] Kinsey Moore + + * /, main/logger.c: Merged revisions 331654 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331654 | kmoore | 2011-08-12 11:21:37 -0500 + (Fri, 12 Aug 2011) | 19 lines Merged revisions 331649 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331649 | kmoore | 2011-08-12 11:20:25 -0500 (Fri, 12 Aug 2011) | + 12 lines Logger does not warn of failure to open logging channels + Currently, logger only prints an error message to stderr when it + fails to open a logger channel where many users will not see it + because the logger lock is held. The alternative provided by this + patch is to log the error to all attached consoles in the hopes + that it will be easier to see. Additionally, this patch prevents + the failed logger channel from being added to the list where it + would silently fail on each call to the Asterisk logger. (closes + issue ASTERISK-16231) Review: + https://reviewboard.asterisk.org/r/1338 ........ ................ + +2011-08-11 21:55 +0000 [r331580] Jason Parker + + * apps/app_dial.c, /, apps/app_meetme.c: Merged revisions 331579 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331579 | qwell | 2011-08-11 16:54:54 -0500 + (Thu, 11 Aug 2011) | 13 lines Merged revisions 331578 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331578 | qwell | 2011-08-11 16:46:39 -0500 (Thu, 11 Aug 2011) | + 6 lines Use proper values for 64-bit option flags. Also, reusing + bits es no bueno, so change the value of a duplicate. (issue + ASTERISK-18239) ........ ................ + +2011-08-11 21:44 +0000 [r331577] Richard Mudgett + + * /, funcs/func_shell.c: Merged revisions 331576 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331576 | rmudgett | 2011-08-11 16:42:21 -0500 + (Thu, 11 Aug 2011) | 16 lines Merged revisions 331575 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331575 | rmudgett | 2011-08-11 16:39:58 -0500 (Thu, 11 Aug 2011) + | 9 lines Segfault in shell_helper in func_shell.c. The return + value of popen() was not checked for failure to open. (closes + issue ASTERISK-18109) JIRA SWP-3633 Reported by: Michael Myles + Tested by: rmudgett ........ ................ + +2011-08-10 22:24 +0000 [r331519] Kinsey Moore + + * /, channels/chan_sip.c: Merged revisions 331518 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331518 | kmoore | 2011-08-10 17:23:49 -0500 + (Wed, 10 Aug 2011) | 17 lines Merged revisions 331517 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331517 | kmoore | 2011-08-10 17:23:08 -0500 (Wed, 10 Aug 2011) | + 10 lines SIP Notify via AMI or CLI leaks SIP PVTs Any SIP notify + sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing + the additional ref just before the invite and adding an unref + following it corrects the issue as seen via REF_DEBUG. The unref + existed in a distant revision and it appears as though the wrong + ref operation was removed. (closes issue ASTERISK-18091) Review: + https://reviewboard.asterisk.org/r/1332/ ........ + ................ + +2011-08-10 20:51 +0000 [r331419-331463] Richard Mudgett + + * /, main/logger.c: Merged revisions 331462 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331462 | rmudgett | 2011-08-10 15:41:35 -0500 + (Wed, 10 Aug 2011) | 37 lines Merged revisions 331461 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331461 | rmudgett | 2011-08-10 15:29:59 -0500 (Wed, 10 Aug 2011) + | 30 lines Output of queue log not started until logger reloaded. + ASTERISK-15863 caused a regression with queue logging. The output + of the queue log is not started until the logger configuration is + reloaded. * Queue log initialization is completely delayed until + the first message is posted to the queue log system. Including + the initial opening of the queue log file. * Fixed rotate_file() + ROTATE strategy to give the file just rotated out to the + configured exec function after rotate. Just like the other + strategies. * Fixed logger reload to always post the queue reload + entry instead of just if there is a queue log file. * Refactored + some code to eliminate some redundancy and to reduce stack + utilization. (closes issue ASTERISK-17036) JIRA SWP-2952 Reported + by: Juan Carlos Valero Patches: jira_asterisk_17036_v1.8.patch + (license #5621) patch uploaded by rmudgett Tested by: rmudgett + (closes issue ASTERISK-18208) Reported by: Christian Pinedo + Review: https://reviewboard.asterisk.org/r/1333/ ........ + ................ + + * /, main/features.c: Merged revisions 331420 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r331420 | rmudgett | 2011-08-10 14:07:53 -0500 (Wed, 10 Aug 2011) + | 2 lines Make sure feature_request_and_dial() initializes + outstate if passed in. ........ + + * /, main/features.c, CHANGES: Merged revisions 331418 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r331418 | rmudgett | 2011-08-10 13:25:08 -0500 (Wed, 10 Aug 2011) + | 6 lines Revert -r318141. It was a band-aid that only partially + fixed parking. A better fix is on reviewboard review 1358. (issue + ASTERISK-17374) ........ + +2011-08-10 15:45 +0000 [r331371] Jonathan Rose + + * channels/chan_sip.c, CHANGES: SIP display-name needed to be empty + for Avaya IP500 In order to address a compatability issue with + certain features on certain devices which rely on display name + content to change behavior, initreqprep in chan_sip.c has been + changed to no longer substitute cid_number into the display name + when cid_name isn't present. Instead, it will send no display + name in that case. (closes issue ASTERISK-16198) Reported by: + Walter Doekes Review: https://reviewboard.asterisk.org/r/1341/ + +2011-08-10 13:49 +0000 [r331317] Kinsey Moore + + * main/manager.c, /: Merged revisions 331316 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331316 | kmoore | 2011-08-10 08:48:41 -0500 + (Wed, 10 Aug 2011) | 15 lines Merged revisions 331315 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331315 | kmoore | 2011-08-10 08:47:46 -0500 (Wed, 10 Aug 2011) | + 8 lines AMI action ModuleReload returns Error if Module: missing + or empty An empty string was not being checked for properly + causing identification of the module to be reloaded to fail and + return an Error with message "No such module." (closes issue + AST-616) ........ ................ + +2011-08-09 23:17 +0000 [r331266] Richard Mudgett + + * main/pbx.c, /, channels/chan_sip.c, main/features.c, + channels/chan_iax2.c, apps/app_parkandannounce.c: Merged + revisions 331265 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331265 | rmudgett | 2011-08-09 18:12:49 -0500 + (Tue, 09 Aug 2011) | 22 lines Merged revisions 331248 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) + | 15 lines Misc minor items found in code. * Add some reentrancy + protection in pbx.c when creating the contexts_table hash table. + * Fix inverted test in chan_sip.c conditional code. * Fix + uninitialized variable and use of the wrong variable in + chan_iax2.c. * Fix test of return value in app_parkandannounce.c. + Explicitly testing for -1 is bad if the function does not + actually return that value when it fails. * Fixup some comments + and add some curly braces in features.c. ........ + ................ + +2011-08-09 17:12 +0000 [r331202] Alexandr Anikin + + * addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c, /, + addons/ooh323c/src/ooLogChan.c, addons/ooh323c/src/ooq931.c: + Merged revisions 331147,331200 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331147 | may | 2011-08-09 20:16:55 +0400 (Tue, + 09 Aug 2011) | 11 lines Merged revisions 331146 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331146 | may | 2011-08-09 20:13:09 +0400 (Tue, 09 Aug 2011) | 4 + lines move ast_cond_signal for admitted call after all data + filled/freed clear all log channels by pointed number not only + first free allocated callToken in ooh323_answer ........ + ................ r331200 | may | 2011-08-09 20:36:39 +0400 (Tue, + 09 Aug 2011) | 9 lines Setup IP proto version for call in GK mode + Added additional check for IP semantics before parse destination + by ast_parse_args due to it can parse numeric as IP. (closes + issue ASTERISK-18218) Reported by: slesru Patch: + ASTERISK-18218.patch ................ + +2011-08-09 17:08 +0000 [r331201] Kinsey Moore + + * funcs/func_enum.c, UPGRADE.txt, main/enum.c: Allow ENUM query + functions to report lookup errors The ENUM dialplan functions do + not report DNS query errors properly. It is useful to + differentiate between failed query (e.g. non-existent domain) vs. + no data records of the appropriate type. This is required to make + overlapped dialing work. (closes issue ASTERISK-13769) Review: + https://reviewboard.asterisk.org/r/1355/ Patch-by: Timo Teras + +2011-08-09 16:02 +0000 [r331140-331144] Jason Parker + + * /, doc/asterisk.8: Merged revisions 331143 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331143 | qwell | 2011-08-09 10:59:54 -0500 + (Tue, 09 Aug 2011) | 9 lines Merged revisions 331142 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r331142 | qwell | 2011-08-09 10:58:16 -0500 (Tue, 09 Aug + 2011) | 1 line Regenerate asterisk man page from sgml. ........ + ................ + + * /, doc/asterisk.8, configs/asterisk.conf.sample, + configs/voicemail.conf.sample, doc/asterisk.sgml: Merged + revisions 331139 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331139 | qwell | 2011-08-09 10:50:07 -0500 + (Tue, 09 Aug 2011) | 19 lines Merged revisions 306999 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r306999 | lathama | 2011-02-08 14:22:35 -0600 (Tue, 08 Feb 2011) + | 12 lines Documentation Updates Note default polling setting in + voicemail.conf Add missing config to asterisk.conf Update manpage + (issue #16505) Reported by: tzafrir Patches: + asterisk_sgml_fixes_demo.diff uploaded by tzafrir (license 46) + Tested by: lathama, tzafrir ........ ................ + + * doc/asterisk.8, configs/asterisk.conf.sample, + configs/voicemail.conf.sample, doc/asterisk.sgml: Merged + revisions 331138 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r331138 | qwell | 2011-08-09 10:47:20 -0500 (Tue, 09 Aug 2011) | + 1 line Revert merge of r306999, due to merge conflict. ........ + +2011-08-08 22:59 +0000 [r331042-331098] Terry Wilson + + * /, UPGRADE.txt, CHANGES, include/asterisk/manager.h: Merged + revisions 331097 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r331097 | twilson | 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) + | 5 lines Bump the AMI protocol version to 1.2 As a result of + converting Unlink events that were missed in the AMI 1.1 update + to Bridge events, the AMI protocol version is being incremented. + ........ + + * main/channel.c, /, CHANGES: Merged revisions 331041 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011) + | 6 lines Replace AMI Unlink events with Bridge events A previous + update converted some of the Link and Unlink events to Bridge + events, but a couple of Unlink events were missed. This patch + rectifies the situation. (closes issues ASTERISK-17455) ........ + +2011-08-08 20:54 +0000 [r331000-331040] Kinsey Moore + + * /, res/res_musiconhold.c: Merged revisions 331039 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r331039 | kmoore | 2011-08-08 15:53:30 -0500 + (Mon, 08 Aug 2011) | 18 lines Merged revisions 331038 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r331038 | kmoore | 2011-08-08 15:52:45 -0500 (Mon, 08 Aug 2011) | + 11 lines In-queue MOH stops after a periodic announcement If the + seek value is past the end of file when resuming G.722 MOH, MOH + will cease to function for the duration of the MOH session + through all starts and stops until saved state is cleared. + Adjusting the code to guarantee a single valid read (which is + already assumed) fixes the bug. (closes issue ASTERISK-18077) + Review: https://reviewboard.asterisk.org/r/1328/ Tested-by: + Jonathan Rose ........ ................ + + * configs/queues.conf.sample, apps/app_queue.c: Log queue member + name when state_interface is set for ADDMEMBER and REMOVEMEMBER + events app_queue logs the events ADDMEMBER and REMOVEMEMBER with + the agent field set to the interface value rather than the + membername value when a member is added with a state_interface + value set. However all other member related queue events are + logged with the membername when a state_interface is set. This + patch makes these fields optionally more consistent and correct. + (closes issue ASTERISK-14769) Review: + https://reviewboard.asterisk.org/r/1286 Patch-by: Jamuel Starkey + Tested-by: Kinsey Moore + + * apps/app_queue.c: app_queue: Add StateInterface to output of + "queue show" and "QueueStatus" This patch adds the + state_interface of the queue member struct to the output of + "queue show" (CLI command) and "QueueStatus" (AMI action) when + displaying relevant queue member information. For the AMI event + message the variable StateInterface has been added. (closes issue + ASTERISK-18071) Review: https://reviewboard.asterisk.org/r/1300/ + Patch-by: Jamuel Starkey + +2011-08-05 15:57 +0000 [r330941] David Vossel + + * /, codecs/codec_resample.c: Merged revisions 330940 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r330940 | dvossel | 2011-08-05 10:53:49 -0500 (Fri, 05 Aug 2011) + | 2 lines The slin resampler is no longer dependent on an + external library, but the dependency was not removed correctly. + ........ + +2011-08-05 08:47 +0000 [r330903] Alexandr Anikin + + * addons/ooh323c/src/ooGkClient.c, /, + addons/ooh323c/src/ooCmdChannel.c: Merged revisions 330899 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330899 | may | 2011-08-05 11:38:28 +0400 (Fri, + 05 Aug 2011) | 11 lines Merged revisions 330827 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r330827 | may | 2011-08-04 23:37:16 +0400 (Thu, 04 Aug 2011) | 4 + lines change gk client behaivour on rrq/grq failures to setup + timers and next tries after timeout instead of complete failure + in the ooh323 stack ........ ................ + +2011-08-04 20:53 +0000 [r330845] Terry Wilson + + * /, configure, configure.ac: Merged revisions 330844 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330844 | twilson | 2011-08-04 15:51:23 -0500 + (Thu, 04 Aug 2011) | 11 lines Merged revisions 330843 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r330843 | twilson | 2011-08-04 15:29:19 -0500 (Thu, 04 Aug 2011) + | 4 lines Make libsrtp instructions more explicit when linking + fails (closes issue ASTERISK-18139) ........ ................ + +2011-08-03 15:16 +0000 [r330707-330764] Kinsey Moore + + * /, main/Makefile: Merged revisions 330763 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330763 | kmoore | 2011-08-03 10:15:26 -0500 + (Wed, 03 Aug 2011) | 16 lines Merged revisions 330762 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r330762 | kmoore | 2011-08-03 10:14:36 -0500 (Wed, 03 Aug 2011) | + 9 lines editing files in main/editline does not ensure rebuild of + libedit.a When editing a source file in main/editline, the build + system does not rebuild libedit.a and uses the already existing + one instead. Adding a PHONY to CHECK_SUBDIR fixes this problem. + (closes issue ASTERISK-16221) Patch-by: Walter Doekes ........ + ................ + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 330706 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330706 | kmoore | 2011-08-03 08:39:06 -0500 + (Wed, 03 Aug 2011) | 17 lines Merged revisions 330705 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r330705 | kmoore | 2011-08-03 08:38:17 -0500 (Wed, 03 Aug 2011) | + 10 lines Call pickup broken for DAHDI channels when beginning + with # The call pickup feature did not work on DAHDI devices for + anything other than feature codes beginning with * since all + feature codes in chan_dahdi were originally hard-coded to begin + with *. This patch is also applied to chan_dahdi.c to fix this + bug with radio modes. (closes issue AST-621) Review: + https://reviewboard.asterisk.org/r/1336/ ........ + ................ + +2011-08-02 20:54 +0000 [r330650] Kevin P. Fleming + + * /, res/res_jabber.c: Merged revisions 330649 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330649 | kpfleming | 2011-08-02 15:52:44 -0500 + (Tue, 02 Aug 2011) | 9 lines Merged revisions 330648 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r330648 | kpfleming | 2011-08-02 15:51:56 -0500 (Tue, 02 + Aug 2011) | 2 lines Convert an error message to actually be + helpful. ........ ................ + +2011-08-02 16:19 +0000 [r330577-330593] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 330586 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330586 | dvossel | 2011-08-02 11:17:59 -0500 + (Tue, 02 Aug 2011) | 15 lines Merged revisions 330581 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r330581 | dvossel | 2011-08-02 11:15:08 -0500 (Tue, 02 Aug 2011) + | 8 lines Fixes crash in chan_iax2. Fixes crash in chan_iax2 + resulting from an edge case in the way control frames are queued + during calltoken negotiation is complete. (closes issue + ASTERISK-17610) Reported by: mgrobecker ........ ................ + + * /, channels/chan_sip.c: Merged revisions 330579 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330579 | dvossel | 2011-08-02 11:08:57 -0500 + (Tue, 02 Aug 2011) | 9 lines Merged revisions 330578 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r330578 | dvossel | 2011-08-02 11:07:02 -0500 (Tue, 02 + Aug 2011) | 2 lines Optimization to buffer initialization fix. + ........ ................ + + * /, channels/chan_sip.c: Merged revisions 330576 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330576 | dvossel | 2011-08-02 10:55:36 -0500 + (Tue, 02 Aug 2011) | 12 lines Merged revisions 330575 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r330575 | dvossel | 2011-08-02 10:53:21 -0500 (Tue, 02 Aug 2011) + | 5 lines Fixes uninitialized string buffer in log message. + (closes issue ASTERISK-17200) Reported by: lmadsen ........ + ................ + +2011-08-01 15:24 +0000 [r330435] Kinsey Moore + + * /, main/say.c: Merged revisions 330434 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330434 | kmoore | 2011-08-01 10:23:29 -0500 + (Mon, 01 Aug 2011) | 16 lines Merged revisions 330433 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r330433 | kmoore | 2011-08-01 10:22:10 -0500 (Mon, 01 Aug 2011) | + 9 lines Incorrect playback for Spanish in some circumstances When + you say the time in spanish and it is 01:00 - 01:59 or 13:00 - + 13:59 you must use female pronunciation "1F". The function + "say_date_with_format_es" does not take this in account. (closes + ASTERISK-15016) Patch-by: Luis Jimenez ........ ................ + +2011-07-31 00:19 +0000 [r330370-330379] Richard Mudgett + + * main/astobj2.c: Fixed compiler warning and a couple prototype + mismatches. + + * main/channel.c, /: Merged revisions 330369 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330369 | rmudgett | 2011-07-30 18:57:56 -0500 + (Sat, 30 Jul 2011) | 11 lines Merged revisions 330368 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011) + | 4 lines Remove some redundant locking code in + ast_do_masquerade(). Also updated some comments. ........ + ................ + +2011-07-30 15:54 +0000 [r330313] Gregory Nietsky + + * main/channel.c, /: Merged revisions 330312 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330312 | irroot | 2011-07-30 17:34:41 +0200 + (Sat, 30 Jul 2011) | 15 lines Merged revisions 330311 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) | + 9 lines prevent double masqurading channels when one is been hung + up and deadlock avoidance is used. There is a race condition in + ast_do_masquerade / ast_hangup (at least) Reported by me signed + off by schmidts with input from David Vossel Review: + https://reviewboard.asterisk.org/r/1323/ ........ + ................ + +2011-07-29 19:34 +0000 [r330273] Russell Bryant + + * include/asterisk/astobj2.h, tests/test_astobj2.c, + channels/chan_iax2.c, main/astobj2.c: astobj2: Avoid using + temporary objects + ao2_find() with OBJ_POINTER. There is a + fairly common pattern making its way through the code base where + we put a temporary object on the stack so we can call ao2_find() + with OBJ_POINTER. The purpose is so that it can be passed into + the object hash function. However, this really seems like a hack + and potentially error prone. This patch is a first stab at + approach to avoid having to do that. It adds a new flag, OBJ_KEY, + which can be used instead of OBJ_POINTER in these situations. + Then, the hash function can know whether it was given an object + or some custom data to hash. The patch also changes some uses of + ao2_find() for iax2_user and iax2_peer objects to reflect how + OBJ_KEY would be used. So long, and thanks for all the fish. + Review: https://reviewboard.asterisk.org/r/1184/ + +2011-07-29 17:20 +0000 [r330205-330221] Sean Bright + + * /, formats/format_wav.c: Merged revisions 330217 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330217 | seanbright | 2011-07-29 13:19:42 -0400 + (Fri, 29 Jul 2011) | 9 lines Merged revisions 330213 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r330213 | seanbright | 2011-07-29 13:18:56 -0400 (Fri, + 29 Jul 2011) | 2 lines Correct the check for O_RDONLY. ........ + ................ + + * /, formats/format_wav.c: Merged revisions 330204 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330204 | seanbright | 2011-07-29 12:58:40 -0400 + (Fri, 29 Jul 2011) | 9 lines Merged revisions 330203 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r330203 | seanbright | 2011-07-29 12:58:08 -0400 (Fri, + 29 Jul 2011) | 2 lines Only write to wav files that were opened + to be written to. ........ ................ + +2011-07-29 05:27 +0000 [r330163] Paul Belanger + + * /, apps/app_confbridge.c: Merged revisions 330162 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r330162 | pabelanger | 2011-07-29 01:25:18 -0400 (Fri, 29 Jul + 2011) | 4 lines Fix typo pointed out on #asterisk Thanks notten + ........ + +2011-07-28 21:46 +0000 [r330109] Terry Wilson + + * /, main/term.c: Merged revisions 330108 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330108 | twilson | 2011-07-28 16:44:31 -0500 + (Thu, 28 Jul 2011) | 9 lines Merged revisions 330107 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r330107 | twilson | 2011-07-28 16:42:41 -0500 (Thu, 28 + Jul 2011) | 2 lines Make console colors work for + TERM=xterm-256color ........ ................ + +2011-07-28 17:16 +0000 [r330052] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 330051 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r330051 | rmudgett | 2011-07-28 12:10:37 -0500 + (Thu, 28 Jul 2011) | 29 lines Merged revisions 330050 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r330050 | rmudgett | 2011-07-28 12:04:24 -0500 + (Thu, 28 Jul 2011) | 22 lines Merged revisions 330033 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu, + 28 Jul 2011) | 15 lines Datacalls with B410P fail. Incoming and + outgoing call legs of a data call are using different formats: + a-law, u-law. When the call is bridged, the media stream is run + through translation to convert the media formats. The translation + is bad for data calls. * Make incoming call that does not + explicitly specify u-law or a-law use the DAHDI channel's default + law. The outgoing call always uses the default law from the DAHDI + channel. (closes issue ABE-2800) Patches: + jira_abe_2800_companding.patch (license #5621) patch uploaded by + rmudgett .......... ................ ................ + +2011-07-28 15:46 +0000 [r329996] Jason Parker + + * /, channels/chan_sip.c: Merged revisions 329995 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329995 | qwell | 2011-07-28 10:45:49 -0500 + (Thu, 28 Jul 2011) | 13 lines Merged revisions 329994 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329994 | qwell | 2011-07-28 10:45:24 -0500 (Thu, 28 Jul 2011) | + 6 lines Fix a SIP transfer deadlock. The locking in this function + is very scary. There are like 6 structs involved. (closes issue + AST-470) ........ ................ + +2011-07-28 15:30 +0000 [r329993] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 329992 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329992 | mnicholson | 2011-07-28 10:28:21 -0500 + (Thu, 28 Jul 2011) | 13 lines Merged revisions 329991 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329991 | mnicholson | 2011-07-28 10:26:56 -0500 (Thu, 28 Jul + 2011) | 6 lines check for CONFIG_STATUS_FILE_INVALID when loading + the res_fax config file Patch by: tzafrir Reported by: tzafrir + (closes issue ASTERISK-18161) ........ ................ + +2011-07-28 13:04 +0000 [r329897-329953] Sean Bright + + * configs/confbridge.conf.sample, /: Merged revisions 329952 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ........ r329952 | seanbright | 2011-07-28 09:03:58 -0400 (Thu, + 28 Jul 2011) | 4 lines The default conf-usermenu says that '8' + can be used to leave the conference, so put that in the sample + user menu. '5' is supposed to extend the conference, but there + doesn't appear to be a concept of that in the menu actions. + ........ + + * /, apps/app_confbridge.c: Merged revisions 329950 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r329950 | seanbright | 2011-07-28 08:43:55 -0400 (Thu, 28 Jul + 2011) | 1 line Correct the spelling of 'conference.' ........ + + * /, channels/chan_sip.c: Merged revisions 329896 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329896 | seanbright | 2011-07-28 07:35:27 -0400 + (Thu, 28 Jul 2011) | 9 lines Merged revisions 329895 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r329895 | seanbright | 2011-07-28 07:34:33 -0400 (Thu, + 28 Jul 2011) | 2 lines Make the output of Externhost in 'sip show + settings' more consistent. ........ ................ + +2011-07-27 21:22 +0000 [r329835-329856] Jonathan Rose + + * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: + reverting 329840 due to failing tests. Going to change this + feature to be purely optional. + + * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Adds cdr + logging of calls resulting in CONGESTION Applies a patch made a + long time ago by alecdavis which adds a CDR feature for logging + calls that failed due to congestion. (closes issue #15907) + Reported by: alecdavis Patches: cdr_congestion.diff.txt uploaded + by alecdavis (license #5546) Review: + https://reviewboard.asterisk.org/r/454/ + +2011-07-27 19:19 +0000 [r329775] Sean Bright + + * /, Makefile.moddir_rules: Merged revisions 329771 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329771 | seanbright | 2011-07-27 15:18:47 -0400 + (Wed, 27 Jul 2011) | 15 lines Merged revisions 329767 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329767 | seanbright | 2011-07-27 15:17:46 -0400 (Wed, 27 Jul + 2011) | 8 lines Explicitly sort the module list so that the + menuselect lists are sorted. (closes ASTERISK-18141) Reported by: + Richard Miller Patches: sort-order.diff uploaded by seanbright + (License #5060) Tested by: leifmadsen ........ ................ + +2011-07-27 18:12 +0000 [r329711] Jonathan Rose + + * /, configs/indications.conf.sample: Merged revisions 329710 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329710 | jrose | 2011-07-27 13:11:07 -0500 + (Wed, 27 Jul 2011) | 14 lines Merged revisions 329709 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329709 | jrose | 2011-07-27 13:10:30 -0500 (Wed, 27 Jul 2011) | + 8 lines Fix New Zealand indications profile based on + http://www.telepermit.co.nz/TNA102.pdf (closes issue + ASTERISK-16263) Reported by: richardf Patches: + nz-indications.patch uploaded by richardf (License #6015) + ........ ................ + +2011-07-27 15:26 +0000 [r329671] Sean Bright + + * /, main/loader.c: Merged revisions 329670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r329670 | seanbright | 2011-07-27 11:25:53 -0400 (Wed, 27 Jul + 2011) | 2 lines Sort the module list so that 'module show' is + alphabetical. ........ + +2011-07-27 04:27 +0000 [r329615] Tilghman Lesher + + * /, cdr/cdr_odbc.c: Merged revisions 329614 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329614 | tilghman | 2011-07-26 23:25:26 -0500 + (Tue, 26 Jul 2011) | 13 lines Merged revisions 329613 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329613 | tilghman | 2011-07-26 23:23:46 -0500 (Tue, 26 Jul 2011) + | 6 lines Duration and billsec are swapped in high resolution + time. Closes ASTERISK-18024 Patches: + 20110726__ASTERISK-18024.diff by Tilghman Lesher (License 5003) + ........ ................ + +2011-07-26 14:27 +0000 [r329530-329564] Jonathan Rose + + * /, apps/app_voicemail.c: Merged revisions 329538 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329538 | jrose | 2011-07-26 09:19:34 -0500 + (Tue, 26 Jul 2011) | 11 lines Merged revisions 329529 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329529 | jrose | 2011-07-26 09:04:55 -0500 (Tue, 26 Jul 2011) | + 5 lines Changes sound file for prepend "then-press-pound" to + "vm-then-pound" which is the same prompt, only it turned out + "then-press-pound" was part of extra sounds. Also, vm is more + appropriate anyway. ........ ................ + + * include/asterisk/app.h, /, configs/voicemail.conf.sample, + main/app.c, apps/app_voicemail.c: Merged revisions 329528 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 + (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | + 17 lines Fixes some voicemail forwarding behavior based around + prepend mode. Formerly, prepend forwarding would have the user + record a message with no useful prompt and an expectation for the + user to push a button on the phone when finished recording. If a + length of silence was detected instead, the recording would be + canceled and the user would re-enter the voicemail forwarding + menu. Subsequent time-outs in prepend recording would also bug + out in the sense that they would write over the original message + and get sent to the recipient regardless of whether they timed + out or were accepted. This patch fixes this issue and adds a + prompt which will be played after a timeout informing the user + that they needed to press a button. Currently, the sound files + that we have are somewhat inadquate for this, so after the call + we simply have Allison say "Please try again. Then press pound." + which actually relies on two separate sound files. Just one would + be more appropriate. reporter: Vlad Povorozniuc Review: + https://reviewboard.asterisk.org/r/1327/ ........ + ................ + +2011-07-25 19:57 +0000 [r329473] Paul Belanger + + * /, main/enum.c: Merged revisions 329472 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329472 | pabelanger | 2011-07-25 15:55:33 -0400 + (Mon, 25 Jul 2011) | 9 lines Merged revisions 329471 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r329471 | pabelanger | 2011-07-25 15:49:40 -0400 (Mon, + 25 Jul 2011) | 2 lines Decrease verbose messages to debug, to + help clean up CLI. ........ ................ + +2011-07-25 14:07 +0000 [r329391-329432] Gregory Nietsky + + * include/asterisk/dsp.h, main/dsp.c: dsp_process was enhanced to + work with alaw and ulaw in addition to slin. noticed that some + functions could be refactored here it is. Reported by: irroot + Tested by: irroot, mnicholson Review: + https://reviewboard.asterisk.org/r/1304/ + + * channels/chan_sip.c, channels/sip/include/sip.h: Remove + lastmsgssent from sip it has not been working since 1.6 Clean up + the return values to be consistant not currently used Add doxygen + returns MWI Event is sent on Register (closes issue + ASTERISK-17866) Reported by: one47 Tested by: irroot, mvanbaak + Review: https://reviewboard.asterisk.org/r/1172/ + +2011-07-22 21:15 +0000 [r329332-329335] Richard Mudgett + + * main/pbx.c, /: Merged revisions 329334 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 ........ + r329334 | rmudgett | 2011-07-22 16:14:22 -0500 (Fri, 22 Jul 2011) + | 1 line Make use less redundant loop construct for iterating + over hints. ........ + + * main/pbx.c, /: Merged revisions 329331 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329331 | rmudgett | 2011-07-22 15:43:07 -0500 + (Fri, 22 Jul 2011) | 55 lines Merged revisions 329299 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011) + | 48 lines Deadlocks dealing with dialplan hints during reload. + There are two remaining different deadlocks reported dealing with + dialplan hints. The deadlock in ASTERISK-17666 is caused by + invalid locking order in ast_remove_hint(). The hints container + must be locked before the hint object. The deadlock in + ASTERISK-17760 is caused by a catch-22 situation in + handle_statechange(). The deadlock is caused by not having the + conlock before calling the watcher callbacks. Unfortunately, + having that lock causes a different deadlock as reported in + ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made + handle_statechange() no longer call the watcher callbacks holding + any locks that matter. * Made hint ao2 destructor do the watcher + callbacks for extension deactivation to guarantee that they get + called. * Fixed hint reference leak in ast_add_hint() if the + callback container constructor failed. * Fixed hint reference + leak in complete_core_show_hint() for every hint it found for CLI + tab completion. * Adjusted locking in + ast_merge_contexts_and_delete() for safety. * Added + context_merge_lock to prevent ast_merge_contexts_and_delete() and + handle_statechange() from interfering with each other. * Fixed + ast_change_hint() not taking into account that the extension is + used for the hash key. (closes issue ASTERISK-17666) Reported by: + irroot Tested by: irroot JIRA SWP-3318 (closes issue + ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA + SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/ + ........ ................ + +2011-07-21 20:26 +0000 [r329258] Russell Bryant + + * channels/chan_dahdi.c, /, main/features.c, + include/asterisk/netsock2.h, CHANGES, channels/sig_pri.c, + include/asterisk/rtp_engine.h: Merged revisions 329257 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ........ r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 + Jul 2011) | 2 lines s/1.10/10.0/ ........ + +2011-07-21 18:06 +0000 [r329146-329205] Richard Mudgett + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged + revisions 329204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329204 | rmudgett | 2011-07-21 13:05:18 -0500 + (Thu, 21 Jul 2011) | 13 lines Merged revisions 329203 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329203 | rmudgett | 2011-07-21 13:04:09 -0500 (Thu, 21 Jul 2011) + | 6 lines Document parkinglot in chan_dahdi.conf.sample. * + Document existing feature in chan_dahdi.conf.sample. * Remove + some dead code related to the parkinglot option. ........ + ................ + + * /, apps/app_directed_pickup.c: Merged revisions 329200 via + svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329200 | rmudgett | 2011-07-21 12:32:02 -0500 + (Thu, 21 Jul 2011) | 24 lines Merged revisions 329199 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329199 | rmudgett | 2011-07-21 12:30:57 -0500 (Thu, 21 Jul 2011) + | 17 lines Update PickupChan documentation. The PickupChan uses + the ampersand as the argument separator. Was documented as: + PickupChan(channel[,channel2[,...][,options]]) Fixed + documentation to: + PickupChan(Technology/Resource[&Technology2/Resource2[&...]][,options]) + This is a continuation of ASTERISK-17494 for v1.8 and later. + (closes issue ASTERISK-18144) Reported by: Erik Smith Patches: + pickupchan_ducumentation-v2.patch (License #6263) patch uploaded + by Erik Smith Tested by: Erik Smith ........ ................ + + * /, main/features.c: Merged revisions 329145 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/10 + ................ r329145 | rmudgett | 2011-07-21 11:52:17 -0500 + (Thu, 21 Jul 2011) | 16 lines Merged revisions 329144 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) + | 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed + more times than we've locked! This appears to be a leftover from + when ast_channel was converted to ao2 objects. Simply removed the + extraneous unlock. (closes issue ASTERISK-17772) ........ + ................ + +2011-07-21 16:22 +0000 [r329106-329130] Jason Parker + + * UPGRADE-1.10.txt (removed), UPGRADE-10.txt (added), UPGRADE.txt: + Fix UPGRADE.txt files for Asterisk 10. + + * /: Remove another 2.0 property. + +2011-07-21 16:05 +0000 [r329105] Russell Bryant + + * /: Fix merge properties to reflect Asterisk 10 branch +