From 148fd0c0deb3f4d98bef971b525d21313db1aaa5 Mon Sep 17 00:00:00 2001 From: Asterisk Autobuilder Date: Mon, 10 Dec 2012 01:56:12 +0000 Subject: [PATCH] Importing files for 10.12.0-digiumphones-rc1 release. git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/10.12.0-digiumphones-rc1@377522 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- .lastclean | 1 + .version | 1 + ChangeLog | 30080 +++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 30082 insertions(+) create mode 100644 .lastclean create mode 100644 .version create mode 100644 ChangeLog 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..7a6ba3611f --- /dev/null +++ b/.version @@ -0,0 +1 @@ +10.12.0-digiumphones-rc1 diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 0000000000..3f1a49f1da --- /dev/null +++ b/ChangeLog @@ -0,0 +1,30080 @@ +2012-12-10 Asterisk Development Team + + * Asterisk 10.12.0-digiumphones-rc1 Released. + +2012-12-03 20:25 +0000 [r377068-377134] Automerge script + + * /: automerge cancel + + * main/cli.c, main/cdr.c, /: Merged revisions 377070,377074 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r377070 | rmudgett | 2012-12-03 12:41:28 -0600 + (Mon, 03 Dec 2012) | 15 lines 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 + ................ r377074 | rmudgett | 2012-12-03 13:14:14 -0600 + (Mon, 03 Dec 2012) | 12 lines 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 + ................ + + * /, main/ccss.c: Merged revisions 377038 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r377038 | rmudgett | 2012-12-03 11:06:44 -0600 + (Mon, 03 Dec 2012) | 10 lines 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 + ................ + +2012-11-30 22:25 +0000 [r376656-376982] Automerge script + + * channels/misdn/isdn_lib.c, /: Merged revisions 376951 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r376951 | rmudgett | 2012-11-30 15:33:38 -0600 + (Fri, 30 Nov 2012) | 18 lines 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 + ................ + + * /, channels/chan_sip.c, funcs/func_volume.c: Merged revisions + 376916,376920 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376916 | mmichelson | 2012-11-30 10:23:46 -0600 + (Fri, 30 Nov 2012) | 23 lines 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 + ................ r376920 | seanbright | 2012-11-30 11:06:21 -0600 + (Fri, 30 Nov 2012) | 5 lines Minor spelling fix to the VOLUME + documentation. ........ Merged revisions 376919 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * channels/chan_local.c, /, channels/chan_sip.c: Merged revisions + 376865,376869 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376865 | rmudgett | 2012-11-29 16:30:26 -0600 + (Thu, 29 Nov 2012) | 7 lines Fix compile error. (issue + ASTERISK-20724) ........ Merged revisions 376864 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ r376869 | rmudgett | 2012-11-29 16:58:28 -0600 + (Thu, 29 Nov 2012) | 7 lines chan_local: Fix local_pvt ref leak + in local_devicestate(). Regression introduced by ASTERISK-20390 + fix. ........ Merged revisions 376868 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * /, channels/chan_sip.c: Merged revisions 376835 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376835 | elguero | 2012-11-29 15:51:50 -0600 + (Thu, 29 Nov 2012) | 19 lines 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 + ................ + + * main/astmm.c, main/asterisk.c, /: Merged revisions 376789 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r376789 | rmudgett | 2012-11-28 18:45:11 -0600 + (Wed, 28 Nov 2012) | 26 lines 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 + ................ + + * main/astmm.c, /: Merged revisions 376759 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376759 | rmudgett | 2012-11-28 18:05:25 -0600 + (Wed, 28 Nov 2012) | 19 lines 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 + ................ + + * main/manager.c, /: Merged revisions 376726 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376726 | jrose | 2012-11-28 10:30:27 -0600 + (Wed, 28 Nov 2012) | 16 lines 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 + ................ + + * UPGRADE.txt, main/pbx.c, /: Merged revisions 376689 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r376689 | rmudgett | 2012-11-27 17:58:23 -0600 + (Tue, 27 Nov 2012) | 33 lines 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 + ................ + + * apps/app_celgenuserevent.c, pbx/pbx_dundi.c, + addons/res_config_mysql.c, /: Merged revisions 376658 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r376658 | rmudgett | 2012-11-27 14:36:45 -0600 + (Tue, 27 Nov 2012) | 21 lines 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 + ................ + + * include/asterisk/linkedlists.h, /: Merged revisions 376628 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r376628 | rmudgett | 2012-11-27 11:35:54 -0600 + (Tue, 27 Nov 2012) | 7 lines 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 + ................ + +2012-11-23 00:27 +0000 [r376614] Automerge script + + * main/logger.c, include/asterisk/lock.h, main/lock.c, /: Merged + revisions 376587 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376587 | mjordan | 2012-11-22 17:56:00 -0600 + (Thu, 22 Nov 2012) | 23 lines 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 + ................ + +2012-11-20 17:24 +0000 [r376497-376539] Automerge script + + * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged + revisions 376522 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376522 | mmichelson | 2012-11-20 11:01:04 -0600 + (Tue, 20 Nov 2012) | 14 lines 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 + ................ + + * main/indications.c, /, channels/chan_sip.c, + main/security_events.c: Merged revisions 376470 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376470 | wdoekes | 2012-11-19 13:44:58 -0600 + (Mon, 19 Nov 2012) | 11 lines 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 + ................ + +2012-11-18 20:23 +0000 [r376427-376446] Automerge script + + * main/utils.c, main/stdtime/localtime.c, main/asterisk.c, /: + Merged revisions 376431 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376431 | mjordan | 2012-11-18 14:18:24 -0600 + (Sun, 18 Nov 2012) | 49 lines 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 + ................ + + * apps/confbridge/conf_state.c, /: Merged revisions 376414 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ........ r376414 | mjordan | 2012-11-18 08:22:39 -0600 (Sun, 18 + Nov 2012) | 8 lines 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. ........ + +2012-11-16 20:23 +0000 [r376338-376408] Automerge script + + * res/res_monitor.c, /: Merged revisions 376390 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376390 | jrose | 2012-11-16 13:41:55 -0600 + (Fri, 16 Nov 2012) | 17 lines 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 + ................ + + * utils/extconf.c, /: Merged revisions 376342 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376342 | dlee | 2012-11-15 18:08:56 -0600 (Thu, + 15 Nov 2012) | 9 lines 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 + ................ + + * utils/hashtest.c (removed), tests/test_hashtab_thrash.c (added), + utils/hashtest2.c (removed), include/asterisk/hashtab.h, + utils/Makefile, tests/test_astobj2_thrash.c (added), + utils/utils.xml, /, apps/app_meetme.c: Merged revisions + 376308,376315 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376308 | jrose | 2012-11-15 16:55:04 -0600 + (Thu, 15 Nov 2012) | 17 lines 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 + ................ r376315 | dlee | 2012-11-15 17:17:54 -0600 (Thu, + 15 Nov 2012) | 28 lines 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 + ................ + +2012-11-15 23:06 +0000 [r376309-376311] David M. Lee + + * utils/hashtest.c (added), tests/test_hashtab_thrash.c (removed), + utils/hashtest2.c (added), include/asterisk/hashtab.h, + utils/Makefile, tests/test_astobj2_thrash.c (removed), + utils/utils.xml, /: Reverted r376309; merged to wrong branch + + * utils/hashtest.c (removed), tests/test_hashtab_thrash.c (added), + utils/hashtest2.c (removed), include/asterisk/hashtab.h, + utils/Makefile, tests/test_astobj2_thrash.c (added), + utils/utils.xml, /: 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 + +2012-11-15 02:22 +0000 [r376260-376281] Automerge script + + * apps/app_voicemail.c, /: Merged revisions 376263 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r376263 | newtonr | 2012-11-14 19:50:54 -0600 + (Wed, 14 Nov 2012) | 10 lines (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 + ................ + + * pbx/pbx_spool.c, /: Merged revisions 376233 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376233 | rmudgett | 2012-11-14 13:50:52 -0600 + (Wed, 14 Nov 2012) | 19 lines 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 + ................ + +2012-11-13 18:22 +0000 [r376165-376216] Automerge script + + * main/channel.c, /: Merged revisions 376208 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376208 | beagles | 2012-11-13 12:20:13 -0600 + (Tue, 13 Nov 2012) | 14 lines 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 + ................ + + * main/pbx.c, /: Merged revisions 376167 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376167 | file | 2012-11-12 14:44:56 -0600 (Mon, + 12 Nov 2012) | 14 lines 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 + ................ + + * main/pbx.c, /: Merged revisions 376143 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376143 | elguero | 2012-11-12 14:15:27 -0600 + (Mon, 12 Nov 2012) | 20 lines 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 + ................ + +2012-11-08 22:25 +0000 [r375992-376117] Automerge script + + * /, res/res_fax.c: Merged revisions 376088 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376088 | mmichelson | 2012-11-08 15:59:13 -0600 + (Thu, 08 Nov 2012) | 12 lines 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 + ................ + + * channels/sig_ss7.c, /: Merged revisions 376059 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376059 | rmudgett | 2012-11-08 15:07:09 -0600 + (Thu, 08 Nov 2012) | 16 lines 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 + ................ + + * main/utils.c, main/astmm.c, main/asterisk.c, + include/asterisk/utils.h, include/asterisk/astmm.h, /: Merged + revisions 376030 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r376030 | rmudgett | 2012-11-08 11:08:39 -0600 + (Thu, 08 Nov 2012) | 35 lines 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 + ................ + + * 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, + channels/sig_analog.c, apps/app_waitforring.c, + 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: Merged revisions 375995 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375995 | mmichelson | 2012-11-07 11:16:24 -0600 + (Wed, 07 Nov 2012) | 41 lines 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 + ................ + + * main/features.c, include/asterisk/channel.h, .cleancount, + include/asterisk/features.h, main/channel.c, /: Merged revisions + 375965 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375965 | rmudgett | 2012-11-06 12:27:19 -0600 + (Tue, 06 Nov 2012) | 21 lines 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 + ................ + +2012-12-10 Asterisk Development Team + + * Asterisk 10.11.0-digiumphones Released. + +2012-12-06 Asterisk Development Team + + * Asterisk 10.11.0-digiumphones-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 10.11.0-digiumphones-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 10.11.0-digiumphones-rc1 Released. + +2012-11-05 23:26 +0000 [r375858-375921] Automerge script + + * 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, main/timing.c, main/channel.c, /, + res/res_timing_pthread.c: Merged revisions 375894 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r375894 | mjordan | 2012-11-05 17:00:32 -0600 + (Mon, 05 Nov 2012) | 28 lines 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 + ................ + + * main/loader.c, /: Merged revisions 375863 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375863 | rmudgett | 2012-11-05 15:39:00 -0600 + (Mon, 05 Nov 2012) | 10 lines 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 + ................ + + * UPGRADE.txt, /: Merged revisions 375846 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r375846 | jrose | 2012-11-05 11:55:13 -0600 (Mon, 05 Nov 2012) | + 9 lines 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/ ........ + +2012-11-04 03:25 +0000 [r375612-375828] Automerge script + + * main/manager.c, /, res/res_fax.c: Merged revisions + 375794,375797,375801 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375794 | mjordan | 2012-11-03 21:30:30 -0500 + (Sat, 03 Nov 2012) | 15 lines 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 + ................ r375797 | mjordan | 2012-11-03 21:42:43 -0500 + (Sat, 03 Nov 2012) | 9 lines 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) ................ r375801 | mjordan | 2012-11-03 22:08:12 + -0500 (Sat, 03 Nov 2012) | 17 lines 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 + ................ + + * main/db.c, main/xmldoc.c, /: Merged revisions 375759,375761 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r375759 | mjordan | 2012-11-03 19:55:19 -0500 + (Sat, 03 Nov 2012) | 18 lines Fix memory leak when unloading XML + documentation This patch is a modified version of a patch + originally committed for the Asterisk 11 branch in r375756. A + portion of that patch, that fixed the memory leak during + unloading XML documentation, applies to branches 1.8 and 10 as + well. The patch for this issue was modified for these two + branches. (issue ASTERISK-20648) Reported by: Corey Farrell + Tested by: mjordan patches: xmldoc-memory_leak.patch uploaded by + Corey Farrell (license 5909) ........ Merged revisions 375758 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ r375761 | mjordan | 2012-11-03 20:13:37 -0500 + (Sat, 03 Nov 2012) | 15 lines 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) ................ + + * main/cdr.c, /: Merged revisions 375728 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375728 | mjordan | 2012-11-03 18:51:43 -0500 + (Sat, 03 Nov 2012) | 16 lines 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 + ................ + + * README, include/asterisk/doxyref.h, /: Merged revisions 375699 + via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375699 | lathama | 2012-11-02 22:15:30 -0500 + (Fri, 02 Nov 2012) | 9 lines 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 + ................ + + * main/format_pref.c, main/channel.c, channels/chan_misdn.c, /, + main/ccss.c: Merged revisions 375659 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375659 | rmudgett | 2012-11-02 15:53:53 -0500 + (Fri, 02 Nov 2012) | 5 lines Things don't need to be that const. + ........ Merged revisions 375658 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged + revisions 375626 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375626 | rmudgett | 2012-11-02 13:42:23 -0500 + (Fri, 02 Nov 2012) | 127 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 375601 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375601 | elguero | 2012-11-02 12:19:33 -0500 + (Fri, 02 Nov 2012) | 14 lines 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 + ................ + +2012-10-31 15:26 +0000 [r375387-375558] Automerge script + + * res/res_calendar_ews.c, /: Merged revisions 375531 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r375531 | mjordan | 2012-10-31 09:34:42 -0500 + (Wed, 31 Oct 2012) | 24 lines 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 + ................ + + * bridges/bridge_softmix.c, apps/app_mixmonitor.c, /: Merged + revisions 375485,375496 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375485 | jrose | 2012-10-30 13:55:58 -0500 + (Tue, 30 Oct 2012) | 8 lines 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 + ................ r375496 | rmudgett | 2012-10-30 14:20:28 -0500 + (Tue, 30 Oct 2012) | 8 lines 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. ................ + + * apps/app_confbridge.c, /: Merged revisions 375470 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 ........ + r375470 | jrose | 2012-10-30 09:42:29 -0500 (Tue, 30 Oct 2012) | + 7 lines confbridge: Fix a bug which made conferences not record + with AMI/CLI commands (closes issue ASTERISK-20601) Reported by: + Vilius Patches: confbridge_mixmonitor.diff uploaded by Jonathan + Rose (license 6182) ........ + + * apps/app_queue.c, /: Merged revisions 375451 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375451 | mjordan | 2012-10-29 21:14:01 -0500 + (Mon, 29 Oct 2012) | 14 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 375417 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375417 | mmichelson | 2012-10-29 16:09:18 -0500 + (Mon, 29 Oct 2012) | 23 lines 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 + ................ + + * main/features.c, /: Merged revisions 375389 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375389 | rmudgett | 2012-10-29 14:28:38 -0500 + (Mon, 29 Oct 2012) | 16 lines 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 + ................ + + * channels/chan_dahdi.c, /: Merged revisions 375362 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r375362 | rmudgett | 2012-10-29 10:51:24 -0500 + (Mon, 29 Oct 2012) | 15 lines 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 + ................ + +2012-10-23 17:25 +0000 [r375290-375350] Automerge script + + * contrib/scripts/ast_tls_cert, /: Merged revisions 375326 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r375326 | jrose | 2012-10-23 11:21:22 -0500 + (Tue, 23 Oct 2012) | 10 lines 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 + ................ + + * main/app.c, /: Merged revisions 375300 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375300 | jrose | 2012-10-22 14:56:20 -0500 + (Mon, 22 Oct 2012) | 14 lines 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 + ................ + + * codecs/gsm/src/code.c, /: Merged revisions 375273 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r375273 | jrose | 2012-10-22 12:08:49 -0500 + (Mon, 22 Oct 2012) | 10 lines 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 + ................ + +2012-10-18 21:25 +0000 [r375214-375241] Automerge script + + * apps/app_queue.c, /: Merged revisions 375217 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375217 | jrose | 2012-10-18 16:09:29 -0500 + (Thu, 18 Oct 2012) | 15 lines 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 + ................ + + * configure.ac, makeopts.in, Makefile, /, build_tools/make_version, + configure: Merged revisions 375190 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375190 | rmudgett | 2012-10-18 14:53:08 -0500 + (Thu, 18 Oct 2012) | 36 lines 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 + ................ + +2012-10-17 19:26 +0000 [r375043-375173] Automerge script + + * main/tcptls.c, /: Merged revisions 375147 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375147 | kmoore | 2012-10-17 13:58:52 -0500 + (Wed, 17 Oct 2012) | 15 lines 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/ ........ Merged + revisions 375146 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * /, channels/chan_sip.c: Merged revisions 375112 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375112 | wdoekes | 2012-10-16 16:43:29 -0500 + (Tue, 16 Oct 2012) | 10 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 375078 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r375078 | wdoekes | 2012-10-16 14:22:44 -0500 + (Tue, 16 Oct 2012) | 7 lines 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 + ................ + + * contrib/scripts/autosupport, /: Merged revisions 375060 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r375060 | tzafrir | 2012-10-16 14:16:43 -0500 + (Tue, 16 Oct 2012) | 10 lines autosupport: fix bashism '==' is + bashism (bashspecific, fails when dash is /bin/sh). Anyway, a + 'case' works better there. (closes issue ASTERISK-20567) ........ + Merged revisions 375059 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * include/asterisk/strings.h, channels/chan_iax2.c, + apps/app_dial.c, /, main/ccss.c: Merged revisions 375026 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r375026 | mmichelson | 2012-10-15 16:06:42 -0500 + (Mon, 15 Oct 2012) | 22 lines 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 + ................ + +2012-10-14 10:20 +0000 [r374994] Automerge script + + * config.guess, config.sub, /: Merged revisions 374991 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r374991 | tzafrir | 2012-10-14 04:40:24 -0500 + (Sun, 14 Oct 2012) | 12 lines 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 + ................ + +2012-10-12 21:56 +0000 [r374931] 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 16:23 +0000 [r374720-374923] Automerge script + + * include/asterisk/tcptls.h, main/tcptls.c, /, channels/chan_sip.c: + Merged revisions 374906 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374906 | mmichelson | 2012-10-12 11:11:30 -0500 + (Fri, 12 Oct 2012) | 28 lines 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 + ................ + + * main/cdr.c, /: Merged revisions 374844 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374844 | mjordan | 2012-10-11 10:43:19 -0500 + (Thu, 11 Oct 2012) | 29 lines 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 + ................ + + * apps/app_queue.c, /: Merged revisions 374803 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374803 | rmudgett | 2012-10-10 15:55:44 -0500 + (Wed, 10 Oct 2012) | 30 lines 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 + ................ + + * main/pbx.c, /: Merged revisions 374763 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374763 | rmudgett | 2012-10-09 17:19:26 -0500 + (Tue, 09 Oct 2012) | 15 lines 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 + ................ + + * configs/chan_dahdi.conf.sample, /: Merged revisions 374728 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r374728 | rmudgett | 2012-10-08 17:29:47 -0500 + (Mon, 08 Oct 2012) | 15 lines dahdi.conf.sample: Add description + for "buffers" setting. This contains an edited version of the + patch originally uploaded 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 + ................ + + * pbx/pbx_spool.c, /: Merged revisions 374695 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374695 | rmudgett | 2012-10-08 16:11:41 -0500 + (Mon, 08 Oct 2012) | 34 lines 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 of nominal paths in init_outgoing() when merging + into the new_outgoing() function dealing with o->capabilities. + ................ + +2012-11-06 Asterisk Development Team + + * Asterisk 10.10.0-digiumphones Released. + +2012-11-05 Asterisk Development Team + + * Asterisk 10.10.0-digiumphones-rc2 Released. + + * Fix a bug which made ConfBridge not record the conference when + the recording was initiated from an AMI/CLI command + + (closes issue ASTERISK-20601) + Reported by: Vilius + +2012-10-08 Asterisk Development Team + + * Asterisk 10.10.0-digiumphones-rc1 Released. + +2012-10-08 14:19 +0000 [r374656] Automerge script + + * 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), + 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: Merged revisions 374652 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ........ r374652 | mjordan | 2012-10-08 08:46:27 -0500 (Mon, 08 + Oct 2012) | 46 lines 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) ........ + +2012-10-05 21:25 +0000 [r374226-374610] Automerge script + + * main/manager.c, /: Merged revisions 374586 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374586 | dlee | 2012-10-05 15:23:14 -0500 (Fri, + 05 Oct 2012) | 34 lines 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 + ................ + + * channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c, + channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged + revisions 374537 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374537 | rmudgett | 2012-10-05 13:25:20 -0500 + (Fri, 05 Oct 2012) | 162 lines 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 + ................ + + * configs/dsp.conf.sample, CHANGES, main/dsp.c, /: Merged revisions + 374476,374481 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374476 | alecdavis | 2012-10-04 15:05:14 -0500 + (Thu, 04 Oct 2012) | 13 lines 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 + ................ r374481 | alecdavis | 2012-10-04 15:17:16 -0500 + (Thu, 04 Oct 2012) | 17 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 374457 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374457 | file | 2012-10-04 12:44:38 -0500 (Thu, + 04 Oct 2012) | 17 lines Fix a regression from direct media ACLs + where the directrtpsetup option no longer works. A check was + added for direct media ACLs that immediately forbid remote + bridging if there was no bridged channel. This caused + directrtpsetup to no longer function as it needs this information + before bridging actually occurs. Logic has now been adjusted so + if there is no bridged channel a remote bridge will still be + attempted. (closes issue ASTERISK-20511) Reported by: kristoff + Review: https://reviewboard.asterisk.org/r/2146/ ........ Merged + revisions 374456 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * res/res_agi.c, main/db.c, /: Merged revisions 374427 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r374427 | dlee | 2012-10-04 10:37:11 -0500 (Thu, + 04 Oct 2012) | 25 lines 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 + ................ + + * configs/dsp.conf.sample, CHANGES, main/dsp.c, /: Merged revisions + 374385 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374385 | alecdavis | 2012-10-03 23:41:19 -0500 + (Wed, 03 Oct 2012) | 36 lines 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 + ................ + + * main/dsp.c, /: Merged revisions 374370 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374370 | alecdavis | 2012-10-03 23:18:44 -0500 + (Wed, 03 Oct 2012) | 15 lines _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 + ................ + + * res/res_jabber.c, /: Merged revisions 374336 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374336 | mjordan | 2012-10-03 21:11:05 -0500 + (Wed, 03 Oct 2012) | 31 lines 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. + 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 + ................ + + * /, main/ccss.c: Merged revisions 374300 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r374300 | mjordan | 2012-10-03 12:25:36 -0500 (Wed, 03 Oct 2012) + | 10 lines 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. ........ + + * main/asterisk.c, /: Merged revisions 374231 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374231 | mjordan | 2012-10-02 16:12:30 -0500 + (Tue, 02 Oct 2012) | 9 lines 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 + ................ + + * main/message.c, /: Merged revisions 374210 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r374210 | mjordan | 2012-10-02 12:10:04 -0500 (Tue, 02 Oct 2012) + | 10 lines 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 + 11+) can use the ao2_t_* variants of the functions to allow the + REF_DEBUG flag to enable/disable their debug counterparts. + ........ + +2012-10-02 16:41 +0000 [r374208-374209] Jason Parker + + * /: Re-enable automerge. + + * channels/chan_agent.c, main/features.c, main/cel.c, + main/format_pref.c, main/indications.c, main/message.c, + main/asterisk.c, main/db.c, main/channel.c, main/format.c, + main/data.c, main/pbx.c, main/manager.c, /, main/ccss.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 + +2012-10-02 01:23 +0000 [r374148-374195] Automerge script + + * /: automerge cancel + + * tests/test_db.c, apps/app_queue.c, main/db.c, + include/asterisk/astdb.h, /: Merged revisions 374132,374135 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r374132 | seanbright | 2012-10-01 12:27:22 -0500 + (Mon, 01 Oct 2012) | 2 lines Use ast_copy_string instead of + strncpy to guarantee a NUL terminated string. ................ + r374135 | seanbright | 2012-10-01 12:52:38 -0500 (Mon, 01 Oct + 2012) | 23 lines 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 + ................ + +2012-09-28 19:25 +0000 [r373498-374058] Automerge script + + * res/res_jabber.c, /: Merged revisions 374045 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r374045 | jrose | 2012-09-28 14:21:10 -0500 + (Fri, 28 Sep 2012) | 12 lines 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 + ................ + + * res/res_agi.c, /: Merged revisions 373990 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373990 | file | 2012-09-28 07:15:48 -0500 (Fri, + 28 Sep 2012) | 8 lines 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 + ................ + + * apps/app_senddtmf.c, /: Merged revisions 373946 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373946 | rmudgett | 2012-09-27 17:12:47 -0500 + (Thu, 27 Sep 2012) | 14 lines 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 + ................ + + * main/loader.c, /: Merged revisions 373910 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373910 | file | 2012-09-27 11:50:46 -0500 (Thu, + 27 Sep 2012) | 24 lines 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 + ................ + + * channels/chan_local.c, /: Merged revisions 373879 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r373879 | file | 2012-09-27 06:32:13 -0500 (Thu, + 27 Sep 2012) | 14 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 373849 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373849 | mmichelson | 2012-09-26 16:11:35 -0500 + (Wed, 26 Sep 2012) | 8 lines 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 + ................ + + * /, apps/app_meetme.c: Merged revisions 373816 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373816 | rmudgett | 2012-09-26 13:15:50 -0500 + (Wed, 26 Sep 2012) | 18 lines 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 + ................ + + * channels/chan_agent.c, configs/agents.conf.sample, /, main/say.c: + Merged revisions 373769,373774 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373769 | mmichelson | 2012-09-25 17:54:13 -0500 + (Tue, 25 Sep 2012) | 11 lines 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 + ................ r373774 | mmichelson | 2012-09-25 18:08:46 -0500 + (Tue, 25 Sep 2012) | 10 lines 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 + ................ + + * apps/app_voicemail.c, /: Merged revisions 373737 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r373737 | mmichelson | 2012-09-25 16:12:40 -0500 + (Tue, 25 Sep 2012) | 11 lines 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 + ................ + + * res/res_rtp_asterisk.c, channels/chan_local.c, /: Merged + revisions 373703,373706 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373703 | kmoore | 2012-09-25 14:34:01 -0500 + (Tue, 25 Sep 2012) | 11 lines 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 + ................ r373706 | file | 2012-09-25 15:12:02 -0500 (Tue, + 25 Sep 2012) | 22 lines 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 + ................ + + * configs/sip.conf.sample, apps/app_queue.c, + channels/sip/include/sip.h, /, channels/chan_sip.c: Merged + revisions 373665,373675 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373665 | twilson | 2012-09-25 12:35:30 -0500 + (Tue, 25 Sep 2012) | 21 lines 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 + ................ r373675 | kmoore | 2012-09-25 13:20:04 -0500 + (Tue, 25 Sep 2012) | 13 lines "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 + ................ + + * codecs/ilbc/iLBC_decode.c, codecs/Makefile, /, + channels/chan_sip.c, codecs/ilbc/iLBC_encode.c: Merged revisions + 373631,373633,373645 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373631 | jrose | 2012-09-25 11:24:34 -0500 + (Tue, 25 Sep 2012) | 10 lines 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 + ................ r373633 | rmudgett | 2012-09-25 11:33:31 -0500 + (Tue, 25 Sep 2012) | 5 lines 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 + ................ r373645 | rmudgett | 2012-09-25 12:19:52 -0500 + (Tue, 25 Sep 2012) | 14 lines 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 + ................ + + * configs/res_odbc.conf.sample, /: Merged revisions 373579 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r373579 | kmoore | 2012-09-25 08:28:20 -0500 + (Tue, 25 Sep 2012) | 11 lines 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 + ................ + + * res/res_rtp_multicast.c, /: Merged revisions 373551 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r373551 | file | 2012-09-25 07:00:23 -0500 (Tue, + 25 Sep 2012) | 15 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 373533 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373533 | file | 2012-09-24 19:11:28 -0500 (Mon, + 24 Sep 2012) | 5 lines Add missing checks that I neglected. The + SIP technology and SIP info technology should be considered + equal. ........ Merged revisions 373532 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions + 373501,373505 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373501 | rmudgett | 2012-09-24 17:11:01 -0500 + (Mon, 24 Sep 2012) | 18 lines 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 + ................ r373505 | mjordan | 2012-09-24 17:17:02 -0500 + (Mon, 24 Sep 2012) | 19 lines 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 + ................ + + * apps/app_mixmonitor.c, funcs/func_audiohookinherit.c, /, + channels/chan_sip.c: Merged revisions 373466,373468 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r373466 | rmudgett | 2012-09-24 15:44:27 -0500 + (Mon, 24 Sep 2012) | 33 lines 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 + ................ r373468 | jrose | 2012-09-24 16:05:44 -0500 + (Mon, 24 Sep 2012) | 13 lines 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 + ................ + +2012-09-24 21:22 +0000 [r373490-373491] Jason Parker + + * /: reenable automerge + + * /, 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 + +2012-09-24 19:22 +0000 [r373455] Automerge script + + * /: automerge cancel + +2012-09-21 19:24 +0000 [r373158-373367] Automerge script + + * channels/iax2-provision.c, /: Merged revisions 373343 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r373343 | jrose | 2012-09-21 14:08:58 -0500 + (Fri, 21 Sep 2012) | 10 lines 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 + ................ + + * apps/app_queue.c, /: Merged revisions 373300 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373300 | jrose | 2012-09-21 10:07:38 -0500 + (Fri, 21 Sep 2012) | 12 lines 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 + ................ + + * res/res_rtp_asterisk.c, /, apps/app_meetme.c: Merged revisions + 373237,373245 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373237 | mjordan | 2012-09-20 13:42:51 -0500 + (Thu, 20 Sep 2012) | 18 lines 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 + ................ r373245 | file | 2012-09-20 14:14:31 -0500 (Thu, + 20 Sep 2012) | 15 lines 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 + ................ + + * apps/confbridge/conf_config_parser.c, /: Merged revisions 373196 + via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r373196 | mjordan | 2012-09-19 21:35:13 -0500 (Wed, 19 Sep 2012) + | 12 lines Ensure that all ConfBridge sounds can be set using + CONFBRIDGE function The CONFBRIDGE function can be used to set + the sounds in a ConfBridge bridge profile. Unfortunately, three + sounds were missed in the portion of the code that applies the + settings passed in from the function: sound_only_one, join, and + leave. This patch makes sure that the sounds passed from the + function are applied to the bridge profile. (closes issue + ASTERISK-20404) Reported by: univ Tested by: mjordan ........ + + * /, channels/chan_sip.c: Merged revisions 373179 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373179 | file | 2012-09-19 12:05:47 -0500 (Wed, + 19 Sep 2012) | 13 lines Fix a regression where direct media was + not permitted for calls using SIP INFO DTMF. A change was + committed to fix direct media ACL support. This change wrongly + assumed that only a single channel technology structure exists + for chan_sip. This is in fact false as a second exists for calls + using SIP INFO DTMF. The code which performs direct media ACL + checking now checks for both the non-INFO DTMF and INFO DTMF + channel technology structures. (closes issue ASTERISK-20409) + Reported by: michele cicciotti privatewave ........ Merged + revisions 373165 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * main/manager.c, /: Merged revisions 373132 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373132 | seanbright | 2012-09-18 15:13:21 -0500 + (Tue, 18 Sep 2012) | 10 lines 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 + ................ + +2012-09-15 00:21 +0000 [r373078-373106] Automerge script + + * channels/sig_ss7.c, /: Merged revisions 373101 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373101 | rmudgett | 2012-09-14 19:20:21 -0500 + (Fri, 14 Sep 2012) | 20 lines 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 + ................ + + * include/asterisk/astobj2.h, main/ssl.c, main/astobj2.c, + main/tcptls.c, /, channels/chan_sip.c: Merged revisions + 373059,373062 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373059 | mjordan | 2012-09-14 13:28:40 -0500 + (Fri, 14 Sep 2012) | 16 lines Constify __ao2_ref_debug in astobj2 + When REF_DEBUG is enabled in certain files - most notably ccss.c + - the 'tag' parameter passed to __ao2_ref_debug will be a const + char *. The function currently expects that parameter to not be + const. This causes a warning when compiling, as the const + qualifier is being discarded. With dev-mode enabled, this + prevents compiling Asterisk. This patch makes __ao2_ref_debug's + tag and file parameters const. (closes issue ASTERISK-20408) + Reported by: mjordan ........ Merged revisions 372959 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ r373062 | mjordan | 2012-09-14 14:12:48 -0500 + (Fri, 14 Sep 2012) | 30 lines 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 + ................ + +2012-09-13 19:23 +0000 [r373045] Automerge script + + * include/asterisk/channel.h, main/channel.c, /: Merged revisions + 373025 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r373025 | dlee | 2012-09-13 13:44:30 -0500 (Thu, + 13 Sep 2012) | 18 lines 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 + ................ + +2012-09-13 Asterisk Development Team + + * Asterisk 10.9.0-digiumphones-rc1 Released. + +2012-09-12 15:26 +0000 [r372753-372958] Automerge script + + * /, channels/chan_sip.c: Merged revisions 372933 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372933 | mmichelson | 2012-09-12 09:53:35 -0500 + (Wed, 12 Sep 2012) | 10 lines 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 + ................ + + * channels/chan_local.c, /: Merged revisions 372916 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r372916 | jrose | 2012-09-11 17:23:20 -0500 + (Tue, 11 Sep 2012) | 13 lines 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 + ................ + + * include/asterisk/_private.h, main/message.c, main/asterisk.c, /: + Merged revisions 372885 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r372885 | mmichelson | 2012-09-11 16:04:36 -0500 (Tue, 11 Sep + 2012) | 18 lines 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 ........ + + * Makefile, /: Merged revisions 372863 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r372863 | dlee | 2012-09-11 12:14:06 -0500 (Tue, 11 Sep 2012) | 4 + lines Corrects the astsbindir setting when installing the sample + asterisk.conf. (closes issue ASTERISK-20406) ........ + + * main/features.c, /: Merged revisions 372841 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372841 | mmichelson | 2012-09-11 10:30:37 -0500 + (Tue, 11 Sep 2012) | 15 lines 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) ........ + Merged revisions 372840 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * channels/chan_iax2.c, /: Merged revisions 372805 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r372805 | kmoore | 2012-09-10 15:56:35 -0500 + (Mon, 10 Sep 2012) | 13 lines 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 + ................ + + * /, apps/app_meetme.c, channels/chan_sip.c: Merged revisions + 372764,372767 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372764 | kmoore | 2012-09-10 13:32:51 -0500 + (Mon, 10 Sep 2012) | 12 lines 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 + ................ r372767 | jrose | 2012-09-10 13:41:45 -0500 + (Mon, 10 Sep 2012) | 8 lines 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 + ................ + + * main/channel.c, /: Merged revisions 372737 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372737 | jrose | 2012-09-10 12:14:46 -0500 + (Mon, 10 Sep 2012) | 15 lines 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 + ................ + +2012-09-09 02:26 +0000 [r372174-372735] Automerge script + + * channels/sip/sdp_crypto.c, /: Merged revisions 372710 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r372710 | mjordan | 2012-09-08 20:24:36 -0500 + (Sat, 08 Sep 2012) | 24 lines 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 + ................ + + * /, main/Makefile: Merged revisions 372695 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372695 | dlee | 2012-09-08 00:21:41 -0500 (Sat, + 08 Sep 2012) | 10 lines 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 + ................ + + * main/astmm.c, /: Merged revisions 372656 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372656 | rmudgett | 2012-09-07 18:06:38 -0500 + (Fri, 07 Sep 2012) | 8 lines 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 + ................ + + * funcs/func_math.c, apps/app_queue.c, apps/app_voicemail.c, /: + Merged revisions 372621,372625,372629 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372621 | rmudgett | 2012-09-07 16:24:39 -0500 + (Fri, 07 Sep 2012) | 18 lines 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 + ................ r372625 | rmudgett | 2012-09-07 16:49:16 -0500 + (Fri, 07 Sep 2012) | 10 lines 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 + ................ r372629 | rmudgett | 2012-09-07 17:07:59 -0500 + (Fri, 07 Sep 2012) | 8 lines 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 + ................ + + * apps/app_minivm.c, /: Merged revisions 372582 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372582 | mjordan | 2012-09-06 21:25:36 -0500 + (Thu, 06 Sep 2012) | 13 lines 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 + ................ + + * apps/app_minivm.c, /: Merged revisions 372555 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372555 | mjordan | 2012-09-06 21:11:46 -0500 + (Thu, 06 Sep 2012) | 22 lines 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 + ................ + + * apps/app_queue.c, channels/sig_pri.c, /: Merged revisions + 372518,372522 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372518 | kmoore | 2012-09-06 16:40:50 -0500 + (Thu, 06 Sep 2012) | 14 lines 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 + ................ r372522 | rmudgett | 2012-09-06 17:10:04 -0500 + (Thu, 06 Sep 2012) | 22 lines 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 + ................ + + * configs/res_ldap.conf.sample, /, channels/chan_sip.c: Merged + revisions 372499 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r372499 | dsessions | 2012-09-06 13:54:54 -0500 (Thu, 06 Sep + 2012) | 16 lines 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/ ........ + + * UPGRADE-1.8.txt, /: Merged revisions 372472 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372472 | jrose | 2012-09-06 10:54:38 -0500 + (Thu, 06 Sep 2012) | 15 lines 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 + ................ + + * apps/app_queue.c, /: Merged revisions 372445 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372445 | kmoore | 2012-09-06 09:29:35 -0500 + (Thu, 06 Sep 2012) | 14 lines 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 + ................ + + * pbx/pbx_dundi.c, /: Merged revisions 372418 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372418 | mjordan | 2012-09-05 21:49:41 -0500 + (Wed, 05 Sep 2012) | 25 lines 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 + ................ + + * apps/app_followme.c, /: Merged revisions 372391 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372391 | mjordan | 2012-09-05 19:56:47 -0500 + (Wed, 05 Sep 2012) | 24 lines 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 + ................ + + * main/dsp.c, /: Merged revisions 372372 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r372372 | rmudgett | 2012-09-05 14:42:17 -0500 (Wed, 05 Sep 2012) + | 1 line Fix compile error. ........ + + * main/dsp.c, main/pbx.c, main/manager.c, /: Merged revisions + 372338,372341,372358 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372338 | kmoore | 2012-09-05 13:30:49 -0500 + (Wed, 05 Sep 2012) | 13 lines 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 + ................ r372341 | alecdavis | 2012-09-05 13:43:12 -0500 + (Wed, 05 Sep 2012) | 7 lines 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 + ................ r372358 | kmoore | 2012-09-05 14:22:08 -0500 + (Wed, 05 Sep 2012) | 13 lines 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 + ................ + + * apps/app_voicemail.c, /: Merged revisions 372288 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r372288 | mjordan | 2012-09-05 08:42:54 -0500 + (Wed, 05 Sep 2012) | 27 lines 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 + ................ + + * main/dsp.c, /: Merged revisions 372240 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372240 | alecdavis | 2012-09-05 02:37:42 -0500 + (Wed, 05 Sep 2012) | 21 lines 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 + ................ + + * main/dsp.c, /: Merged revisions 372213 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372213 | alecdavis | 2012-09-05 01:47:54 -0500 + (Wed, 05 Sep 2012) | 13 lines 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 + ................ + + * res/res_rtp_asterisk.c, /: Merged revisions 372198 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r372198 | elguero | 2012-09-04 23:47:00 -0500 + (Tue, 04 Sep 2012) | 19 lines 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 + ................ + + * cel/cel_pgsql.c, /: Merged revisions 372165 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372165 | mjordan | 2012-09-04 21:19:25 -0500 + (Tue, 04 Sep 2012) | 18 lines 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 + ................ + +2012-08-30 21:24 +0000 [r371850-372115] Automerge script + + * apps/app_queue.c, /: Merged revisions 372090 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372090 | mmichelson | 2012-08-30 15:53:09 -0500 + (Thu, 30 Aug 2012) | 17 lines 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 + ................ + + * apps/app_queue.c, /: Merged revisions 372049 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r372049 | mmichelson | 2012-08-30 13:33:37 -0500 + (Thu, 30 Aug 2012) | 16 lines 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 + ................ + + * channels/chan_iax2.c, main/manager.c, /, + README-SERIOUSLY.bestpractices.txt: Merged revisions + 371999,372020 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371999 | mjordan | 2012-08-30 11:06:47 -0500 + (Thu, 30 Aug 2012) | 36 lines 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 reported 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 + ................ r372020 | mjordan | 2012-08-30 11:22:54 -0500 + (Thu, 30 Aug 2012) | 17 lines 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 ........ Merged revisions 372015 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * doc/CODING-GUIDELINES (added), /: Merged revisions 371962 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r371962 | mjordan | 2012-08-30 07:48:07 -0500 + (Thu, 30 Aug 2012) | 17 lines 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 + ................ + + * /, apps/app_meetme.c: Merged revisions 371920 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371920 | jrose | 2012-08-29 15:58:21 -0500 + (Wed, 29 Aug 2012) | 5 lines app_meetme: Adding test events for + following activity in MeetMe. ........ Merged revisions 371919 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * main/channel.c, /: Merged revisions 371890 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371890 | rmudgett | 2012-08-29 14:40:20 -0500 + (Wed, 29 Aug 2012) | 13 lines 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 + ................ + + * apps/app_dial.c, /: Merged revisions 371861 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371861 | rmudgett | 2012-08-29 13:24:54 -0500 + (Wed, 29 Aug 2012) | 15 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 371825 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371825 | jrose | 2012-08-29 12:07:35 -0500 + (Wed, 29 Aug 2012) | 8 lines 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 + ................ + +2012-08-27 22:25 +0000 [r371689-371816] Automerge script + + * configs/agents.conf.sample, main/manager.c, /: Merged revisions + 371783,371789 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371783 | mmichelson | 2012-08-27 16:29:29 -0500 + (Mon, 27 Aug 2012) | 9 lines 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 + ................ r371789 | mmichelson | 2012-08-27 16:49:51 -0500 + (Mon, 27 Aug 2012) | 13 lines 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 + ................ + + * configs/queues.conf.sample, /: Merged revisions 371748 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r371748 | mmichelson | 2012-08-27 12:36:43 -0500 + (Mon, 27 Aug 2012) | 10 lines 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 + ................ + + * main/lock.c, /: Merged revisions 371719 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371719 | dlee | 2012-08-27 11:43:09 -0500 (Mon, + 27 Aug 2012) | 15 lines 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 + ................ + + * main/utils.c, /: Merged revisions 371691 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371691 | kmoore | 2012-08-27 08:57:10 -0500 + (Mon, 27 Aug 2012) | 14 lines 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 + ................ + + * main/dsp.c, /: Merged revisions 371663 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371663 | alecdavis | 2012-08-26 18:06:14 -0500 + (Sun, 26 Aug 2012) | 5 lines mf_detect: incorrectly used + DTMF_GSIZE instead of MF_GSIZE ........ Merged revisions 371662 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + +2012-08-21 21:28 +0000 [r371617] Automerge script + + * 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, cdr/cdr_tds.c, main/xmldoc.c, + apps/app_dial.c, channels/chan_dahdi.c, /, channels/chan_sip.c, + funcs/func_odbc.c: Merged revisions 371591 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371591 | mmichelson | 2012-08-21 15:40:18 -0500 + (Tue, 21 Aug 2012) | 22 lines 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 + ................ + +2012-08-20 16:25 +0000 [r371570] Automerge script + + * main/udptl.c, /: Merged revisions 371545 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371545 | kmoore | 2012-08-20 10:27:15 -0500 + (Mon, 20 Aug 2012) | 15 lines 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 + ................ + +2012-08-18 03:19 +0000 [r371254-371534] Automerge script + + * main/http.c, /: Merged revisions 371529 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r371529 | mjordan | 2012-08-17 21:34:10 -0500 (Fri, 17 Aug 2012) + | 7 lines 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/xmldoc.c, /: Merged revisions 371491 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371491 | mjordan | 2012-08-17 15:21:30 -0500 + (Fri, 17 Aug 2012) | 17 lines 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 + ................ + + * main/loader.c, /: Merged revisions 371437 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371437 | kmoore | 2012-08-17 10:51:06 -0500 + (Fri, 17 Aug 2012) | 11 lines 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 + ................ + + * main/config.c, main/loader.c, /: Merged revisions 371394,371398 + via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371394 | kmoore | 2012-08-16 17:42:53 -0500 + (Thu, 16 Aug 2012) | 11 lines 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 + ................ r371398 | twilson | 2012-08-16 17:50:12 -0500 + (Thu, 16 Aug 2012) | 13 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 371358 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371358 | jrose | 2012-08-16 14:05:21 -0500 + (Thu, 16 Aug 2012) | 11 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 371338 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371338 | jrose | 2012-08-16 11:16:04 -0500 + (Thu, 16 Aug 2012) | 14 lines 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 + ................ + + * apps/app_queue.c, /: Merged revisions 371313 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371313 | mmichelson | 2012-08-15 18:19:09 -0500 + (Wed, 15 Aug 2012) | 20 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 371271 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371271 | kmoore | 2012-08-15 15:15:08 -0500 + (Wed, 15 Aug 2012) | 12 lines 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 371250-371251 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r371250 | elguero | 2012-08-14 20:35:57 -0500 (Tue, 14 Aug 2012) + | 17 lines 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) ........ r371251 | elguero | + 2012-08-14 20:43:23 -0500 (Tue, 14 Aug 2012) | 4 lines Reverting + this change that was meant for branch 11. (issue ASTERISK-20221) + ........ + +2012-08-13 20:25 +0000 [r371226] Automerge script + + * main/loader.c, /, apps/app_meetme.c, channels/chan_sip.c: Merged + revisions 371199,371203 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371199 | mmichelson | 2012-08-13 14:51:19 -0500 + (Mon, 13 Aug 2012) | 5 lines Fix problem where incorrect pointer + was checked for nullity. ........ Merged revisions 371198 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ r371203 | kmoore | 2012-08-13 15:04:15 -0500 + (Mon, 13 Aug 2012) | 13 lines 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 + ................ + +2012-08-10 22:23 +0000 [r370922-371168] Automerge script + + * apps/app_queue.c, /: Merged revisions 371142 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371142 | mmichelson | 2012-08-10 16:23:52 -0500 + (Fri, 10 Aug 2012) | 18 lines 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 + ................ + + * addons/chan_ooh323.c, /: Merged revisions 371090 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r371090 | may | 2012-08-10 11:46:38 -0500 (Fri, + 10 Aug 2012) | 12 lines 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 + ................ + + * addons/ooh323c/src/ooGkClient.c, /: Merged revisions 371061 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r371061 | may | 2012-08-10 10:13:10 -0500 (Fri, + 10 Aug 2012) | 10 lines 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 + ................ + + * configure.ac, channels/sig_pri.c, channels/sig_ss7.c, + addons/ooh323c/src/ooGkClient.c, channels/chan_dahdi.c, /, + configure, include/asterisk/autoconfig.h.in: Merged revisions + 371013,371022 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r371013 | rmudgett | 2012-08-09 14:11:01 -0500 + (Thu, 09 Aug 2012) | 5 lines Use better libss7 detection test and + move libpri compile test. ........ Merged revisions 371012 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ r371022 | may | 2012-08-09 14:20:09 -0500 (Thu, + 09 Aug 2012) | 10 lines 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 + ................ + + * addons/ooh323c/src/ooh323ep.c, /, apps/app_meetme.c: Merged + revisions 370986,370989 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370986 | kmoore | 2012-08-09 12:39:52 -0500 + (Thu, 09 Aug 2012) | 11 lines 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 + ................ r370989 | may | 2012-08-09 13:05:34 -0500 (Thu, + 09 Aug 2012) | 5 lines change opening h323 logfile with append + mode instead of overwrite ........ Merged revisions 370988 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * apps/app_chanspy.c, /: Merged revisions 370954 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370954 | elguero | 2012-08-08 17:42:05 -0500 + (Wed, 08 Aug 2012) | 26 lines 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 + ................ + + * main/channel.c, /: Merged revisions 370924 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370924 | kmoore | 2012-08-08 15:29:16 -0500 + (Wed, 08 Aug 2012) | 9 lines 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 + ................ + + * channels/sig_analog.h, channels/chan_dahdi.c, + channels/sig_analog.c, /: Merged revisions 370901 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r370901 | rmudgett | 2012-08-08 15:04:44 -0500 + (Wed, 08 Aug 2012) | 5 lines Fix the analog dial *0 flash-hook of + bridged peer feature. ........ Merged revisions 370900 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + +2012-08-07 20:23 +0000 [r370880] Automerge script + + * main/channel.c, /: Merged revisions 370858 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370858 | kmoore | 2012-08-07 14:21:54 -0500 + (Tue, 07 Aug 2012) | 5 lines Add missing AST_CAUSE_* -> text + translations ........ Merged revisions 370856 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + +2012-08-06 15:24 +0000 [r370817] Automerge script + + * /, channels/chan_sip.c: Merged revisions 370798 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370798 | mmichelson | 2012-08-06 10:02:04 -0500 + (Mon, 06 Aug 2012) | 7 lines 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 + ................ + +2012-08-03 22:26 +0000 [r370793] Automerge script + + * channels/sip/config_parser.c, channels/sip/include/sip.h, /, + channels/chan_sip.c: Merged revisions 370772 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370772 | mmichelson | 2012-08-03 16:50:29 -0500 + (Fri, 03 Aug 2012) | 41 lines 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 + ................ + +2012-08-01 03:22 +0000 [r370633-370717] Automerge script + + * utils/extconf.c, /: Merged revisions 370698 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370698 | kmoore | 2012-07-31 21:26:09 -0500 + (Tue, 31 Jul 2012) | 8 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 370672 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370672 | mjordan | 2012-07-31 16:19:41 -0500 + (Tue, 31 Jul 2012) | 24 lines 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 + ................ + + * funcs/func_cut.c, tests/test_linkedlists.c, + channels/chan_gtalk.c, cdr/cdr_pgsql.c, main/config.c, + channels/chan_jingle.c, pbx/pbx_spool.c, + apps/app_directed_pickup.c, funcs/func_channel.c, + apps/app_minivm.c, main/features.c, res/res_agi.c, main/http.c, + main/logger.c, pbx/pbx_ael.c, apps/app_macro.c, main/event.c, + apps/app_sms.c, main/astmm.c, include/asterisk/strings.h, + main/db.c, main/dsp.c, apps/app_voicemail.c, addons/app_mysql.c, + channels/chan_sip.c, main/threadstorage.c, main/say.c, + apps/app_dictate.c, apps/app_festival.c, funcs/func_strings.c, + pbx/pbx_lua.c, main/utils.c, funcs/func_logic.c, + apps/app_getcpeid.c, channels/chan_iax2.c, res/res_jabber.c, + funcs/func_global.c, main/channel.c, res/ael/pval.c, + main/tcptls.c, apps/app_osplookup.c, main/manager.c, + main/strcompat.c, main/callerid.c, main/file.c, main/app.c, + channels/chan_alsa.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, + utils/extconf.c, addons/chan_mobile.c, apps/app_mixmonitor.c, + main/asterisk.c, apps/app_while.c, addons/res_config_mysql.c, + res/res_config_pgsql.c, main/pbx.c, include/asterisk/utils.h, /, + apps/app_meetme.c: Merged revisions 370643 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370643 | kmoore | 2012-07-31 14:57:09 -0500 + (Tue, 31 Jul 2012) | 12 lines 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/ ........ Merged + revisions 370642 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * configs/sip.conf.sample, channels/sip/include/sip.h, /, + channels/chan_sip.c: Merged revisions 370619 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370619 | mmichelson | 2012-07-31 10:31:57 -0500 + (Tue, 31 Jul 2012) | 34 lines 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. For those who are having some deja + vu, that's because this patch was originally committed to trunk + since there is a new configuration option added. After seeing a + bug report about audio being slow to set up on SIP calls, it + became apparent that this patch would be the best solution for + resolving the issue. The patch is unintrusive and will have no + effect unless the option is explicitly enabled. (closes issue + AST-896) reported by Thomas Arimont (closes issue ASTERISK-19857) + reported by Matt Jordan ........ Merged revisions 370618 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + +2012-09-13 Asterisk Development Team + + * Asterisk 10.8.0-digiumphones Released. + +2012-09-11 Asterisk Development Team + + * Asterisk 10.8.0-digiumphones-rc2 Released. + + * AST-2012-013: Resolve ACL rules being ignored during calls by some + IAX2 peers + + * AST-2012-012: Resolve AMI User Unauthorized Shell Access through + ExternalIVR + + * r371861: 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 + + * r372710: Only re-create an SRTP session when needed; respond with + correct crypto policy + + 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 + + * r372841: 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) + +2012-07-31 Asterisk Development Team + + * Asterisk 10.8.0-digiumphones-rc1 Released. + +2012-07-30 17:24 +0000 [r370555-370584] Automerge script + + * channels/chan_misdn.c, /: Merged revisions 370564 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r370564 | rmudgett | 2012-07-30 11:49:12 -0500 + (Mon, 30 Jul 2012) | 5 lines Release B channel allocation on + error path in chan_misdn. ........ Merged revisions 370563 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * /, apps/app_meetme.c: Merged revisions 370547 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r370547 | jrose | 2012-07-30 09:50:34 -0500 (Mon, 30 Jul 2012) | + 5 lines app_meetme: Change app_meetme support level to extended + from deprecated (closes issue ASTERISK-20134) Reported by: Leif + Madsen ........ + +2012-07-25 21:22 +0000 [r370509] Automerge script + + * res/res_agi.c, /: Merged revisions 370495 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370495 | jrose | 2012-07-25 16:12:50 -0500 + (Wed, 25 Jul 2012) | 14 lines 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 + ................ + +2012-07-25 03:45 +0000 [r370473] 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 21:08 +0000 [r370465] 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 17:24 +0000 [r370381-370452] Automerge script + + * channels/chan_oss.c, main/frame.c, /: Merged revisions + 370430,370432 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370430 | kpfleming | 2012-07-24 11:54:01 -0500 + (Tue, 24 Jul 2012) | 5 lines 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 + ................ r370432 | tzafrir | 2012-07-24 12:08:40 -0500 + (Tue, 24 Jul 2012) | 4 lines chan_oss: fix "sample rate" error + message Merged revisions 370428 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * /, funcs/func_shell.c: Merged revisions 370384 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370384 | kpfleming | 2012-07-23 16:09:53 -0500 + (Mon, 23 Jul 2012) | 5 lines Improve documentation for the + SHELL() dialplan function. ........ Merged revisions 370383 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * main/channel.c, /: Merged revisions 370361 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370361 | kpfleming | 2012-07-23 09:51:21 -0500 + (Mon, 23 Jul 2012) | 13 lines Free any datastores attached to + dummy channels. Revision 370205 added the use of a datastore + attached to a dummy channel to resolve a memory leak, but + ast_dummy_channel_destructor() in this branch did not free + datastores, resulting in a continued (but slightly smaller) + memory leak. This patch backports the change to free said + datastores from the Asterisk trunk. (related to issue AST-916) + ........ Merged revisions 370360 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + +2012-07-19 22:24 +0000 [r370297] Automerge script + + * main/cel.c, res/res_rtp_asterisk.c, /: Merged revisions + 370271,370274,370277 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370271 | mjordan | 2012-07-19 16:37:09 -0500 + (Thu, 19 Jul 2012) | 49 lines 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/ (issue + ASTERISK-18404) Reporter: Stephane Chazelas Tested by: Matt + Jordan ........ Merged revisions 370252 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ r370274 | mjordan | 2012-07-19 17:01:32 -0500 + (Thu, 19 Jul 2012) | 17 lines 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 + ................ r370277 | rmudgett | 2012-07-19 17:11:48 -0500 + (Thu, 19 Jul 2012) | 7 lines 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 + ................ + +2012-07-18 19:23 +0000 [r370202-370224] Automerge script + + * main/cel.c, /: Merged revisions 370206 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370206 | kpfleming | 2012-07-18 14:14:09 -0500 + (Wed, 18 Jul 2012) | 19 lines 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) Review: + https://reviewboard.asterisk.org/r/2053/ ........ Merged + revisions 370205 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * 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, main/channel.c, addons/app_mysql.c, main/pbx.c, + funcs/func_curl.c, /, main/ccss.c, funcs/func_odbc.c: Merged + revisions 370184 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370184 | kpfleming | 2012-07-18 12:13:07 -0500 + (Wed, 18 Jul 2012) | 10 lines 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 + ................ + +2012-07-16 20:24 +0000 [r370101-370151] Automerge script + + * /, channels/chan_sip.c: Merged revisions 370132 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370132 | wdoekes | 2012-07-16 14:52:45 -0500 + (Mon, 16 Jul 2012) | 11 lines 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 + ................ + + * UPGRADE.txt, CHANGES, UPGRADE-1.8.txt, /: Merged revisions 370082 + via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r370082 | kmoore | 2012-07-16 08:51:57 -0500 + (Mon, 16 Jul 2012) | 8 lines 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 + ................ + +2012-07-12 20:24 +0000 [r369958-370036] Automerge script + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_sip.c: Merged revisions 370015,370025 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r370015 | kmoore | 2012-07-12 15:05:45 -0500 + (Thu, 12 Jul 2012) | 11 lines 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 + ................ r370025 | rmudgett | 2012-07-12 15:20:02 -0500 + (Thu, 12 Jul 2012) | 8 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 369994 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369994 | kmoore | 2012-07-12 13:55:17 -0500 + (Thu, 12 Jul 2012) | 12 lines 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 + ................ + + * funcs/func_math.c, /: Merged revisions 369971 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369971 | elguero | 2012-07-12 09:25:45 -0500 + (Thu, 12 Jul 2012) | 14 lines 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 + ................ + + * funcs/func_realtime.c, /: Merged revisions 369938 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r369938 | tilghman | 2012-07-11 12:12:28 -0500 + (Wed, 11 Jul 2012) | 11 lines 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 + ................ + +2012-07-30 Asterisk Development Team + + * Asterisk 10.7.0-digiumphones Released. + +2012-07-11 Asterisk Development Team + + * Asterisk 10.7.0-digiumphones-rc1 Released. + +2012-07-10 14:22 +0000 [r369889] Automerge script + + * apps/app_stack.c, main/pbx.c, /: Merged revisions 369871 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r369871 | kmoore | 2012-07-10 08:35:30 -0500 + (Tue, 10 Jul 2012) | 12 lines 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 + ................ + +2012-07-09 19:51 +0000 [r369846] Joshua Colp + + * 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) + +2012-07-09 17:22 +0000 [r369810-369836] Automerge script + + * configs/sip_notify.conf.sample, /: Merged revisions 369819 via + svnmerge from file:///srv/subversion/repos/asterisk/branches/10 + ................ r369819 | qwell | 2012-07-09 12:06:40 -0500 + (Mon, 09 Jul 2012) | 9 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 369793 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369793 | jrose | 2012-07-09 09:43:49 -0500 + (Mon, 09 Jul 2012) | 9 lines 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 + ................ + +2012-07-06 21:21 +0000 [r369643-369763] Automerge script + + * /, channels/chan_sip.c: Merged revisions 369751 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369751 | jrose | 2012-07-06 16:02:37 -0500 + (Fri, 06 Jul 2012) | 12 lines 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 + ................ + + * main/tcptls.c, /: Merged revisions 369732 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369732 | mmichelson | 2012-07-06 13:47:05 -0500 + (Fri, 06 Jul 2012) | 21 lines 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 + ................ + + * main/bridging.c, /: Merged revisions 369709 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369709 | mmichelson | 2012-07-06 10:23:28 -0500 + (Fri, 06 Jul 2012) | 14 lines 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 + ................ + + * apps/app_voicemail.c, /: Merged revisions 369653 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r369653 | kmoore | 2012-07-05 14:12:33 -0500 + (Thu, 05 Jul 2012) | 20 lines 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) ........ Merged revisions 369652 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * /, channels/chan_sip.c: Merged revisions 369627 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369627 | mjordan | 2012-07-05 12:02:53 -0500 + (Thu, 05 Jul 2012) | 18 lines 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 + ................ + +2012-07-03 17:23 +0000 [r369578-369598] Automerge script + + * /, channels/chan_sip.c: Merged revisions 369580 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369580 | twilson | 2012-07-03 12:02:18 -0500 + (Tue, 03 Jul 2012) | 11 lines 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 + ................ + + * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged + revisions 369558 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369558 | twilson | 2012-07-03 09:34:22 -0500 + (Tue, 03 Jul 2012) | 14 lines 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 + ................ + +2012-06-29 21:19 +0000 [r369488-369516] Automerge script + + * main/rtp_engine.c, /: Merged revisions 369511 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 ........ + r369511 | mmichelson | 2012-06-29 15:28:10 -0500 (Fri, 29 Jun + 2012) | 3 lines Fix apparent copy and paste error where incorrect + "glue" is used. ........ + + * /, channels/chan_sip.c: Merged revisions 369491 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369491 | file | 2012-06-29 11:54:11 -0500 (Fri, + 29 Jun 2012) | 5 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 369472 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369472 | file | 2012-06-29 10:30:47 -0500 (Fri, + 29 Jun 2012) | 10 lines 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 + ................ + +2012-06-27 21:22 +0000 [r369453] Automerge script + + * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged + revisions 369437 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369437 | twilson | 2012-06-27 16:10:01 -0500 + (Wed, 27 Jun 2012) | 16 lines 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. + (closes issue ASTERISK-19992) ........ Merged revisions 369436 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + +2012-06-26 14:21 +0000 [r369322-369406] Automerge script + + * main/adsi.c, /: Merged revisions 369391 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369391 | mjordan | 2012-06-26 08:22:42 -0500 + (Tue, 26 Jun 2012) | 15 lines 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 + ................ + + * main/cdr.c, /: Merged revisions 369369 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369369 | mjordan | 2012-06-25 14:36:02 -0500 + (Mon, 25 Jun 2012) | 29 lines 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 + ................ + + * channels/sip/include/sip.h, /, channels/chan_sip.c: Merged + revisions 369353 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369353 | mmichelson | 2012-06-25 14:16:52 -0500 + (Mon, 25 Jun 2012) | 14 lines 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 + ................ + + * main/features.c, res/res_adsi.c, main/adsi.c (added), + res/res_adsi.exports.in (removed), include/asterisk/adsi.h, /, + main/Makefile: Merged revisions 369325,369328 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369325 | mmichelson | 2012-06-25 10:52:42 -0500 + (Mon, 25 Jun 2012) | 20 lines 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 + ................ r369328 | rmudgett | 2012-06-25 10:59:28 -0500 + (Mon, 25 Jun 2012) | 15 lines 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 + ................ + + * /, channels/chan_sip.c: Merged revisions 369303 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369303 | mmichelson | 2012-06-25 09:23:16 -0500 + (Mon, 25 Jun 2012) | 14 lines 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 + ................ + +2012-06-23 00:20 +0000 [r369213-369294] Automerge script + + * main/features.c, /: Merged revisions 369283 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369283 | rmudgett | 2012-06-22 19:12:27 -0500 + (Fri, 22 Jun 2012) | 22 lines 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 + ................ + + * apps/app_queue.c, apps/app_dial.c, /: Merged revisions + 369259,369263 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369259 | rmudgett | 2012-06-22 16:37:05 -0500 + (Fri, 22 Jun 2012) | 5 lines 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 + ................ r369263 | rmudgett | 2012-06-22 17:09:29 -0500 + (Fri, 22 Jun 2012) | 5 lines 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 + ................ + + * /, channels/chan_sip.c, main/ccss.c: Merged revisions + 369236,369239 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369236 | rmudgett | 2012-06-22 15:49:33 -0500 + (Fri, 22 Jun 2012) | 5 lines 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 + ................ r369239 | rmudgett | 2012-06-22 16:04:25 -0500 + (Fri, 22 Jun 2012) | 5 lines Check if PBX was started for generic + CCSS recall. ........ Merged revisions 369238 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * /, channels/chan_sip.c: Merged revisions 369215 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369215 | twilson | 2012-06-22 14:34:59 -0500 + (Fri, 22 Jun 2012) | 9 lines 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. + ........ Merged revisions 369214 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + ................ + + * /, channels/chan_sip.c: Merged revisions 369206 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369206 | kmoore | 2012-06-22 12:23:26 -0500 + (Fri, 22 Jun 2012) | 11 lines 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 + ................ + +2012-06-20 18:22 +0000 [r369056-369164] Automerge script + + * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c, /: + Merged revisions 369147 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369147 | may | 2012-06-20 12:36:27 -0500 (Wed, + 20 Jun 2012) | 10 lines 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 + ................ + + * include/asterisk/netsock2.h, main/netsock2.c, /: Merged revisions + 369109 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369109 | elguero | 2012-06-19 21:04:58 -0500 + (Tue, 19 Jun 2012) | 23 lines 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 + ................ + + * addons/chan_ooh323.c, /: Merged revisions 369091 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 ........ + r369091 | may | 2012-06-19 18:32:06 -0500 (Tue, 19 Jun 2012) | 9 + lines 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 ........ + + * /, channels/chan_sip.c: Merged revisions 369067 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369067 | mmichelson | 2012-06-19 10:37:37 -0500 + (Tue, 19 Jun 2012) | 17 lines 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 + ................ + + * main/features.c, /: Merged revisions 369044 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r369044 | rmudgett | 2012-06-18 13:11:30 -0500 + (Mon, 18 Jun 2012) | 12 lines 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 + ................ + +2012-06-15 16:30 +0000 [r369026] 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 + +2012-06-15 16:25 +0000 [r369023] Automerge script + + * main/translate.c, channels/vcodecs.c, + channels/sip/security_events.c, main/jitterbuf.c, + main/autochan.c, pbx/dundi-parser.c, main/aoc.c, main/cel.c, + main/enum.c, channels/iax2-parser.c, main/fskmodem.c, + main/config.c, channels/misdn_config.c, main/netsock.c, + build_tools/find_missing_support_level (added), main/loader.c, + main/ulaw.c, main/dial.c, channels/sig_analog.c, main/srv.c, + main/heap.c, main/privacy.c, channels/misdn/ie.c, res/ais/evt.c, + main/syslog.c, res/snmp/agent.c, main/event.c, main/astmm.c, + channels/sip/config_parser.c, channels/vgrabbers.c, main/db.c, + main/udptl.c, main/lock.c, channels/sip/sdp_crypto.c, + main/stun.c, main/frame.c, channels/sip/srtp.c, + main/threadstorage.c, channels/console_video.c, + channels/iax2-provision.c, main/xml.c, main/astfd.c, + main/taskprocessor.c, utils/astdb2bdb.c, + apps/confbridge/conf_config_parser.c, main/channel.c, main/cdr.c, + res/ael/pval.c, channels/chan_misdn.c, main/framehook.c, + main/tdd.c, main/strcompat.c, channels/console_gui.c, + channels/sip/dialplan_functions.c, main/fixedjitterbuf.c, + main/callerid.c, main/file.c, main/app.c, + main/stdtime/localtime.c, main/dns.c, main/message.c, + main/datastore.c, main/sched.c, main/timing.c, main/netsock2.c, + main/fskmodem_float.c, /, main/slinfactory.c, main/acl.c, + channels/sip/reqresp_parser.c, channels/sig_pri.c, + channels/misdn/isdn_lib.c, main/term.c, main/io.c, + main/hashtab.c, main/format_cap.c, main/abstract_jb.c, + main/fskmodem_int.c, main/logger.c, main/audiohook.c, + main/bridging.c, main/dsp.c, main/global_datastores.c, + main/autoservice.c, main/dnsmgr.c, main/security_events.c, + main/say.c, main/utils.c, channels/misdn/isdn_msg_parser.c, + utils/astdb2sqlite3.c, main/devicestate.c, main/ssl.c, + main/format_pref.c, main/astobj2.c, main/indications.c, + main/chanvars.c, main/cli.c, main/tcptls.c, main/data.c, + main/plc.c, main/test.c, channels/console_board.c, + channels/misdn/portinfo.c, main/image.c, main/alaw.c, + channels/sig_ss7.c, main/asterisk.c, main/xmldoc.c, + main/format.c, main/strings.c, main/pbx.c, main/rtp_engine.c, + main/ccss.c, res/ais/clm.c: Merged revisions 369005 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 + ................ r369005 | kpfleming | 2012-06-15 11:07:08 -0500 + (Fri, 15 Jun 2012) | 22 lines 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 + ................ + +2012-06-15 15:32 +0000 [r368963-368999] 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 + + * apps/app_voicemail.c, /: 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 + + * tests/test_voicemail_api.c, main/app.c, + include/asterisk/app_voicemail.h, apps/app_voicemail.c, + include/asterisk/app.h, /: 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 + +2012-06-14 18:20 +0000 [r368872-368960] Automerge script + + * /, channels/chan_skinny.c: Merged revisions 368947 via svnmerge + from file:///srv/subversion/repos/asterisk/branches/10 ........ + r368947 | mjordan | 2012-06-14 12:31:33 -0500 (Thu, 14 Jun 2012) + | 21 lines 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) ........ + + * /, main/Makefile: Merged revisions 368928 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r368928 | mmichelson | 2012-06-14 10:25:23 -0500 + (Thu, 14 Jun 2012) | 10 lines 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 + ................ + + * res/res_adsi.c, res/res_smdi.c, /, funcs/func_volume.c: Merged + revisions 368895,368899 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r368895 | mjordan | 2012-06-13 15:27:28 -0500 + (Wed, 13 Jun 2012) | 21 lines Mark res_smdi/res_adsi as 'core' + supported modules Recently, various issues surrounding weak + attributes 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 attributes. (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 + ................ r368899 | mmichelson | 2012-06-13 16:13:30 -0500 + (Wed, 13 Jun 2012) | 19 lines 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 + ................ + + * /, main/Makefile: Merged revisions 368885 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r368885 | mmichelson | 2012-06-13 14:36:39 -0500 + (Wed, 13 Jun 2012) | 11 lines 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 + ................ + + * Makefile, /: Merged revisions 368831,368853 via svnmerge from + file:///srv/subversion/repos/asterisk/branches/10 + ................ r368831 | mjordan | 2012-06-12 13:30:06 -0500 + (Tue, 12 Jun 2012) | 24 lines 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 + ................ r368853 | mjordan | 2012-06-13 09:30:34 -0500 + (Wed, 13 Jun 2012) | 11 lines 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 + ................ + +2012-06-12 16:22 +0000 [r368810-368826] Jason Parker + + * /: Let's fix the 1.8-merged prop, to give automerge the best + chance at succeeding. + + * funcs/func_strings.c, channels/sip/reqresp_parser.c, + include/asterisk/channel.h, apps/app_queue.c, + channels/chan_iax2.c, main/loader.c, main/channel.c, + channels/chan_dahdi.c, channels/sig_analog.c, + res/res_config_odbc.c, channels/sip/dialplan_functions.c, + apps/app_directory.c, pbx/pbx_config.c, main/md5.c, + res/res_odbc.c, res/res_speech.c, apps/app_voicemail.c, + main/udptl.c, channels/sip/sdp_crypto.c, channels/chan_sip.c, /, + res/res_fax.c, main/say.c: Multiple revisions + 368721,368739,368760,368808 ........ r368721 | kmoore | + 2012-06-11 09:11:14 -0500 (Mon, 11 Jun 2012) | 8 lines 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 ........ + r368739 | kmoore | 2012-06-11 10:15:07 -0500 (Mon, 11 Jun 2012) | + 10 lines 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 + ........ r368760 | rmudgett | 2012-06-11 12:08:50 -0500 (Mon, 11 + Jun 2012) | 17 lines 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 ........ + r368808 | mmichelson | 2012-06-12 10:37:38 -0500 (Tue, 12 Jun + 2012) | 15 lines 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 368721,368739,368760,368808 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * /: Let's try using an automerge-propname, since we have multiple + heads. + + * /: enable automerge + +2012-07-10 Asterisk Development Team + + * Asterisk 10.6.0-digiumphones Released. + +2012-07-06 Asterisk Development Team + + * Asterisk 10.6.0-digiumphones-rc2 Released. + + * AST-2012-009: Skinny Channel Driver Remote Crash Vulnerability + + * AST-2012-010: Possible Resource Leak on Uncompleted Re-INVITE + Transactions + + * AST-2012-011: Remote Crash Vulnerability in VoiceMail Application + + * Fix 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 is is allowed. + + (closes issue ASTERISK-20040) + + * 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 esnt, 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) + + * 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) + +2012-06-12 Asterisk Development Team + + * 10.6.0-digiumphones-rc1 Released. + +2012-06-12 14:03 +0000 [r368791-368792] Matthew Jordan + + * /: Update merge property info + + * 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) + +2012-06-11 22:49 +0000 [r368781-368783] Jason Parker + + * /: Fix merge prop. + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_sip.c: Multiple revisions 368629,368645 ........ + r368629 | mmichelson | 2012-06-06 14:18:20 -0500 (Wed, 06 Jun + 2012) | 31 lines 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 ........ + r368645 | rmudgett | 2012-06-06 16:32:09 -0500 (Wed, 06 Jun 2012) + | 17 lines 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 368629,368645 from + http://svn.asterisk.org/svn/asterisk/branches/10 + + * apps/app_record.c, include/asterisk/channel.h, + res/res_calendar_caldav.c, pbx/dundi-parser.c, + apps/app_followme.c, main/cel.c, apps/app_queue.c, main/enum.c, + channels/iax2-parser.c, res/res_calendar_ews.c, main/config.c, + main/editline/tokenizer.c, channels/chan_dahdi.c, + channels/sig_analog.c, main/editline/readline.c, main/event.c, + channels/sip/config_parser.c, res/ael/ael.flex, + apps/app_chanspy.c, apps/app_stack.c, res/res_odbc.c, + res/res_calendar.c, channels/chan_sip.c, channels/chan_agent.c, + funcs/func_math.c, channels/iax2-provision.c, UPGRADE.txt, + addons/ooh323c/src/h323/H323-MESSAGES.h, channels/chan_iax2.c, + res/res_monitor.c, main/channel.c, addons/ooh323c/src/ooh323.c, + main/cdr.c, res/ael/pval.c, main/manager.c, main/app.c, + pbx/pbx_dundi.c, addons/ooh323c/src/h323/H323-MESSAGESEnc.c, + addons/ooh323c/src/ooq931.c, main/netsock2.c, + res/res_rtp_asterisk.c, apps/app_meetme.c, /, + channels/sip/reqresp_parser.c, main/acl.c, res/res_musiconhold.c, + include/asterisk/tcptls.h, channels/sig_pri.c, res/res_srtp.c, + res/res_config_odbc.c, funcs/func_odbc.c, funcs/func_cdr.c, + funcs/func_channel.c, apps/app_minivm.c, main/features.c, + apps/app_confbridge.c, codecs/codec_dahdi.c, pbx/pbx_config.c, + apps/app_voicemail.c, apps/app_dial.c, funcs/func_speex.c, + res/res_calendar_exchange.c, funcs/func_dialgroup.c, + apps/app_page.c, include/asterisk/cel.h, main/say.c, + funcs/func_lock.c, apps/app_disa.c, main/devicestate.c, CHANGES, + res/res_jabber.c, main/editline/term.c, main/cli.c, + main/tcptls.c, main/data.c, channels/chan_skinny.c, + funcs/func_aes.c, tests/test_config.c, funcs/func_devstate.c, + channels/sip/include/sip.h, channels/sig_ss7.c, main/asterisk.c, + main/xmldoc.c, res/res_calendar_icalendar.c, main/pbx.c, + channels/chan_local.c, addons/format_mp3.c: Multiple revisions + 365155,365160,365299,365320,365399,365475,365478,365575,365632,365701,365898,365990,366049,366053,366106,366168,366241,366297,366390,366412,366591,366598,366741,366792,366881,366884,366948,367003,367028,367267,367299,367369,367417,367470,367562,367679,367731,367782,367844,367907,367978,367981,368042,368093,368267,368310,368407,368470,368499,368524,368536,368568,368587 + ........ r365155 | may | 2012-05-03 09:27:00 -0500 (Thu, 03 May + 2012) | 11 lines 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 + ........ r365160 | may | 2012-05-03 10:01:14 -0500 (Thu, 03 May + 2012) | 11 lines 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 ........ + r365299 | mmichelson | 2012-05-04 10:51:04 -0500 (Fri, 04 May + 2012) | 12 lines 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 ........ + r365320 | rmudgett | 2012-05-04 11:28:06 -0500 (Fri, 04 May 2012) + | 30 lines 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 ........ + r365399 | kmoore | 2012-05-04 17:15:05 -0500 (Fri, 04 May 2012) | + 13 lines 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 ........ + r365475 | mjordan | 2012-05-07 13:39:10 -0500 (Mon, 07 May 2012) + | 20 lines 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 ........ + r365478 | rmudgett | 2012-05-07 13:43:08 -0500 (Mon, 07 May 2012) + | 5 lines Fix type punned compiler warning in test_config.c + ........ Merged revisions 365476 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r365575 | mmichelson | 2012-05-08 10:51:13 -0500 (Tue, 08 May + 2012) | 22 lines 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 ........ + r365632 | rmudgett | 2012-05-08 13:08:01 -0500 (Tue, 08 May 2012) + | 13 lines * 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 ........ + r365701 | rmudgett | 2012-05-08 15:25:08 -0500 (Tue, 08 May 2012) + | 12 lines * 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 call_followme.moh[] and struct fm_args.namerecloc[]. + This fixes unexpected namerecloc filename length restriction. + ........ Merged revisions 365692 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r365898 | mmichelson | 2012-05-09 11:15:28 -0500 (Wed, 09 May + 2012) | 29 lines 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 ........ + r365990 | jrose | 2012-05-09 14:12:32 -0500 (Wed, 09 May 2012) | + 18 lines 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 ........ + r366049 | jrose | 2012-05-10 10:43:06 -0500 (Thu, 10 May 2012) | + 9 lines 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 ........ + r366053 | mmichelson | 2012-05-10 11:13:06 -0500 (Thu, 10 May + 2012) | 9 lines 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 ........ + r366106 | jrose | 2012-05-10 11:55:22 -0500 (Thu, 10 May 2012) | + 9 lines 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 ........ + r366168 | kmoore | 2012-05-10 15:54:08 -0500 (Thu, 10 May 2012) | + 13 lines 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 ........ + r366241 | rmudgett | 2012-05-10 18:42:43 -0500 (Thu, 10 May 2012) + | 7 lines * 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 + ........ r366297 | russell | 2012-05-11 18:59:35 -0500 (Fri, 11 + May 2012) | 19 lines format_mp3: Fix a possible crash 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 + ........ r366390 | mmichelson | 2012-05-14 14:16:36 -0500 (Mon, + 14 May 2012) | 25 lines 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 + ........ r366412 | mmichelson | 2012-05-14 15:06:58 -0500 (Mon, + 14 May 2012) | 19 lines 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 ........ + r366591 | jrose | 2012-05-15 15:44:59 -0500 (Tue, 15 May 2012) | + 15 lines chan_sip: Check the right channel's host address for + directmediapermit/deny Prior to this patch, when checking the + addresses for directmediapermit and denydirectmediadeny, Asterisk + would check the host address of the channel permit/deny was + specified, which defers from the expectations of both our users + and the development team. Instead, directmediapermit/deny now + checks against the address of the channel that the peer with the + ACL is connected to. (issue AST-876) Review: + https://reviewboard.asterisk.org/r/1899/ ........ Merged + revisions 366547 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r366598 | mmichelson | 2012-05-15 18:39:06 -0500 (Tue, 15 May + 2012) | 8 lines 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 ........ + r366741 | mjordan | 2012-05-17 07:57:30 -0500 (Thu, 17 May 2012) + | 23 lines 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 ........ + r366792 | jrose | 2012-05-17 09:41:13 -0500 (Thu, 17 May 2012) | + 10 lines chan_sip: Fix missed locking of opposing pvt for + directmedia acl from r366547 It also required deadlock avoidance + since two sip_pvts structs needed to be locked simultaneously. + Trunk handles it differently, so this is a 1.8 and 10 patch only. + ........ (issue AST-876) Merged revisions 366791 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r366881 | mjordan | 2012-05-18 09:01:56 -0500 (Fri, 18 May 2012) + | 65 lines 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 + ........ r366884 | kmoore | 2012-05-18 09:18:47 -0500 (Fri, 18 + May 2012) | 9 lines 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 ........ + r366948 | mjordan | 2012-05-18 10:45:42 -0500 (Fri, 18 May 2012) + | 20 lines 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 (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 ........ + r367003 | mmichelson | 2012-05-18 12:00:14 -0500 (Fri, 18 May + 2012) | 19 lines 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 ........ + r367028 | mmichelson | 2012-05-18 12:50:18 -0500 (Fri, 18 May + 2012) | 18 lines 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 ........ + r367267 | twilson | 2012-05-22 11:17:46 -0500 (Tue, 22 May 2012) + | 14 lines 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 ........ + r367299 | twilson | 2012-05-22 12:21:51 -0500 (Tue, 22 May 2012) + | 21 lines 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 ........ + r367369 | mjordan | 2012-05-23 08:25:04 -0500 (Wed, 23 May 2012) + | 26 lines 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 + ........ r367417 | mmichelson | 2012-05-23 15:29:03 -0500 (Wed, + 23 May 2012) | 7 lines 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 ........ + r367470 | rmudgett | 2012-05-23 18:16:49 -0500 (Wed, 23 May 2012) + | 9 lines 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 + ........ r367562 | mjordan | 2012-05-24 08:32:33 -0500 (Thu, 24 + May 2012) | 24 lines 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) ........ r367679 | rmudgett | 2012-05-24 + 17:29:23 -0500 (Thu, 24 May 2012) | 34 lines 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 ........ + r367731 | elguero | 2012-05-24 21:29:26 -0500 (Thu, 24 May 2012) + | 20 lines 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 ........ + r367782 | rmudgett | 2012-05-25 11:30:55 -0500 (Fri, 25 May 2012) + | 18 lines 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 ........ + r367844 | mjordan | 2012-05-29 13:33:20 -0500 (Tue, 29 May 2012) + | 21 lines 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) ........ r367907 | rmudgett | + 2012-05-29 17:28:55 -0500 (Tue, 29 May 2012) | 17 lines 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 ........ + r367978 | rmudgett | 2012-05-30 12:39:24 -0500 (Wed, 30 May 2012) + | 19 lines 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 ........ + r367981 | rmudgett | 2012-05-30 13:07:28 -0500 (Wed, 30 May 2012) + | 7 lines Use the DEADLOCK_AVOIDANCE() macro instead. (issue + ASTERISK-19854) ........ Merged revisions 367980 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r368042 | rmudgett | 2012-05-31 13:20:15 -0500 (Thu, 31 May 2012) + | 10 lines 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 ........ + r368093 | elguero | 2012-05-31 22:28:09 -0500 (Thu, 31 May 2012) + | 17 lines 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 ........ + r368267 | kpfleming | 2012-06-01 15:22:44 -0500 (Fri, 01 Jun + 2012) | 20 lines 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 ........ + r368310 | rmudgett | 2012-06-01 18:24:25 -0500 (Fri, 01 Jun 2012) + | 15 lines 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 ........ + r368407 | rmudgett | 2012-06-04 14:08:52 -0500 (Mon, 04 Jun 2012) + | 23 lines 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 ........ + r368470 | rmudgett | 2012-06-04 16:11:42 -0500 (Mon, 04 Jun 2012) + | 10 lines 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 ........ + r368499 | mmichelson | 2012-06-04 17:02:26 -0500 (Mon, 04 Jun + 2012) | 16 lines 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 ........ + r368524 | kmoore | 2012-06-05 10:19:58 -0500 (Tue, 05 Jun 2012) | + 11 lines 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 ........ + r368536 | kmoore | 2012-06-05 10:27:01 -0500 (Tue, 05 Jun 2012) | + 8 lines 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 ........ + r368568 | rmudgett | 2012-06-05 20:10:10 -0500 (Tue, 05 Jun 2012) + | 15 lines 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 ........ + r368587 | kmoore | 2012-06-06 11:09:10 -0500 (Wed, 06 Jun 2012) | + 12 lines 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 + 365155,365160,365299,365320,365399,365475,365478,365575,365632,365701,365898,365990,366049,366053,366106,366168,366241,366297,366390,366412,366591,366598,366741,366792,366881,366884,366948,367003,367028,367267,367299,367369,367417,367470,367562,367679,367731,367782,367844,367907,367978,367981,368042,368093,368267,368310,368407,368470,368499,368524,368536,368568,368587 + from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-06 17:35 +0000 [r368609] 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". + +2012-05-21 19:16 +0000 [r367162] Mark Michelson + + * main/callerid.c, include/asterisk/callerid.h, + channels/chan_sip.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 + +2012-05-04 21:28 +0000 [r365396] Jason Parker + + * apps/app_mixmonitor.c, apps/app_voicemail.c, /: Add support for + folders in MixMonitor 'm' option. Backport manager actions. The + manager actions are needed, so MixMonitor can be executed on + existing channels. (issue DPMA-68) ........ Merged revisions + 365395 from + http://svn.asterisk.org/svn/asterisk/certified/branches/1.8.11 + +2012-05-03 20:54 +0000 [r365297] Mark Michelson + + * apps/app_mixmonitor.c: Populate file extensions for mixmonitor + recordings properly. + +2012-05-03 20:06 +0000 [r365264] Jason Parker + + * main/jitterbuf.c, configs/queues.conf.sample, + configs/usbradio.conf.sample (removed), + res/res_calendar_caldav.c, apps/rpt_flow.pdf (removed), + apps/app_queue.c, main/cel.c, res/res_config_sqlite.c, + res/res_calendar_ews.c, main/config.c, formats/format_siren7.c, + channels/chan_dahdi.c, formats/format_vox.c, funcs/func_volume.c, + configure, formats/format_h263.c, main/event.c, + apps/app_chanspy.c, formats/format_g719.c, channels/chan_sip.c, + funcs/func_env.c, channels/chan_agent.c, funcs/func_strings.c, + channels/console_video.c, Makefile.rules, main/astfd.c, + formats/format_wav_gsm.c, bridges/bridge_multiplexed.c, + channels/chan_iax2.c, funcs/func_global.c, + apps/confbridge/conf_config_parser.c, res/res_config_curl.c, + build_tools/cflags.xml, main/cdr.c, funcs/func_curl.c, + main/manager.c, main/tdd.c, channels/console_gui.c, + formats/format_pcm.c, main/app.c, main/stdtime/localtime.c, + utils/extconf.c, makeopts.in, main/message.c, + formats/format_gsm.c, res/res_clioriginate.c, + include/asterisk/time.h, res/res_rtp_asterisk.c, + res/res_config_pgsql.c, apps/app_meetme.c, /, + formats/format_wav.c, configure.ac, res/res_musiconhold.c, + channels/chan_gtalk.c, tests/test_linkedlists.c, apps/app_ices.c, + channels/sig_pri.c, res/res_srtp.c, formats/format_ilbc.c, + channels/sig_pri.h, Makefile, apps/app_forkcdr.c, + res/res_config_odbc.c, bridges/bridge_builtin_features.c, + codecs/gsm/src/k6opt.s, build_tools/menuselect-deps.in, + funcs/func_channel.c, apps/app_directed_pickup.c, + main/features.c, res/res_agi.c, main/http.c, main/logger.c, + apps/app_confbridge.c, apps/app_sms.c, main/audiohook.c, + formats/format_h264.c, apps/app_voicemail.c, + codecs/lpc10/Makefile, apps/app_dial.c, formats/format_sln.c, + codecs/gsm/Makefile, funcs/func_sysinfo.c, + formats/format_ogg_vorbis.c, CHANGES, main/astobj2.c, + main/format_pref.c, apps/app_speech_utils.c, + tests/test_security_events.c, main/tcptls.c, + addons/ooh323cDriver.c, formats/format_g723.c, + apps/app_externalivr.c, tests/test_config.c, tests/test_poll.c, + addons/chan_mobile.c, formats/format_siren14.c, + funcs/func_devstate.c, main/asterisk.c, main/xmldoc.c, + channels/chan_mgcp.c, formats/format_g729.c, + channels/chan_unistim.c, configs/chan_dahdi.conf.sample, + main/pbx.c, res/res_calendar_icalendar.c, channels/chan_local.c, + funcs/func_version.c, configs/rpt.conf.sample (removed): Multiple + revisions + 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083 + ........ r361208 | jrose | 2012-04-04 14:30:09 -0500 (Wed, 04 Apr + 2012) | 10 lines 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 + ........ r361211 | jrose | 2012-04-04 15:00:23 -0500 (Wed, 04 Apr + 2012) | 12 lines 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 ........ + r361270 | jrose | 2012-04-05 11:53:35 -0500 (Thu, 05 Apr 2012) | + 10 lines 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 ........ + r361330 | kmoore | 2012-04-06 08:31:51 -0500 (Fri, 06 Apr 2012) | + 11 lines 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 ........ + r361333 | mjordan | 2012-04-06 09:01:33 -0500 (Fri, 06 Apr 2012) + | 11 lines 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 ........ + r361381 | russell | 2012-04-06 10:49:19 -0500 (Fri, 06 Apr 2012) + | 5 lines 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 ........ + r361422 | pabelanger | 2012-04-06 11:31:18 -0500 (Fri, 06 Apr + 2012) | 14 lines 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 ........ + r361472 | kmoore | 2012-04-06 13:13:04 -0500 (Fri, 06 Apr 2012) | + 5 lines Add missing newlines to CLI logging ........ Merged + revisions 361471 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r361522 | rmudgett | 2012-04-06 14:47:29 -0500 (Fri, 06 Apr 2012) + | 8 lines 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. ........ r361560 | mjordan | 2012-04-06 + 15:32:13 -0500 (Fri, 06 Apr 2012) | 13 lines 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 + ........ r361607 | mjordan | 2012-04-06 17:00:11 -0500 (Fri, 06 + Apr 2012) | 12 lines 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 ........ + r361658 | mjordan | 2012-04-09 14:42:53 -0500 (Mon, 09 Apr 2012) + | 15 lines 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 ........ + r361706 | mjordan | 2012-04-09 15:54:55 -0500 (Mon, 09 Apr 2012) + | 17 lines 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 ........ + r361754 | mjordan | 2012-04-09 16:44:30 -0500 (Mon, 09 Apr 2012) + | 12 lines 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 ........ + r361804 | mjordan | 2012-04-10 14:57:30 -0500 (Tue, 10 Apr 2012) + | 10 lines 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 ........ + r361855 | rmudgett | 2012-04-10 16:47:42 -0500 (Tue, 10 Apr 2012) + | 19 lines 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 ........ + r361907 | jrose | 2012-04-11 11:07:50 -0500 (Wed, 11 Apr 2012) | + 10 lines 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/ ........ r361956 | + kmoore | 2012-04-12 10:01:13 -0500 (Thu, 12 Apr 2012) | 13 lines + 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 ........ + r361981 | kmoore | 2012-04-12 11:22:28 -0500 (Thu, 12 Apr 2012) | + 12 lines 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 + ........ r362080 | jrose | 2012-04-13 10:30:22 -0500 (Fri, 13 Apr + 2012) | 10 lines 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 ........ + r362084 | jrose | 2012-04-13 11:04:22 -0500 (Fri, 13 Apr 2012) | + 15 lines 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 ........ + r362152 | mjordan | 2012-04-16 14:39:32 -0500 (Mon, 16 Apr 2012) + | 19 lines 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 ........ + r362202 | mjordan | 2012-04-16 16:40:29 -0500 (Mon, 16 Apr 2012) + | 18 lines 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 ........ + r362205 | mjordan | 2012-04-16 16:57:19 -0500 (Mon, 16 Apr 2012) + | 25 lines 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 ........ + r362264 | elguero | 2012-04-17 09:53:04 -0500 (Tue, 17 Apr 2012) + | 23 lines 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 ........ + r362305 | mjordan | 2012-04-17 13:27:44 -0500 (Tue, 17 Apr 2012) + | 15 lines 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 ........ + r362356 | mjordan | 2012-04-17 15:56:05 -0500 (Tue, 17 Apr 2012) + | 17 lines 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 ........ + r362357 | jrose | 2012-04-17 15:57:36 -0500 (Tue, 17 Apr 2012) | + 12 lines 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 ........ + r362360 | mjordan | 2012-04-17 16:07:29 -0500 (Tue, 17 Apr 2012) + | 24 lines 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 ........ + r362364 | mjordan | 2012-04-17 16:11:25 -0500 (Tue, 17 Apr 2012) + | 29 lines 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 ........ + r362377 | mjordan | 2012-04-17 16:22:37 -0500 (Tue, 17 Apr 2012) + | 13 lines 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/ + ........ r362429 | rmudgett | 2012-04-18 11:27:51 -0500 (Wed, 18 + Apr 2012) | 19 lines 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 + ........ r362496 | mjordan | 2012-04-18 21:27:08 -0500 (Wed, 18 + Apr 2012) | 50 lines 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 ........ + r362537 | twilson | 2012-04-19 09:31:59 -0500 (Thu, 19 Apr 2012) + | 14 lines 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 ........ + r362587 | seanbright | 2012-04-19 11:04:21 -0500 (Thu, 19 Apr + 2012) | 12 lines 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 ........ + r362678 | rmudgett | 2012-04-19 16:00:21 -0500 (Thu, 19 Apr 2012) + | 5 lines Update membermacro and membergosub documentation in + queues.conf.sample. ........ Merged revisions 362677 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r362681 | elguero | 2012-04-19 16:11:35 -0500 (Thu, 19 Apr 2012) + | 9 lines 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 ........ + r362730 | wdoekes | 2012-04-19 16:59:43 -0500 (Thu, 19 Apr 2012) + | 5 lines Fix documentation for ${VERSION(ASTERISK_VERSION_NUM)}. + ........ Merged revisions 362729 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r362816 | twilson | 2012-04-20 09:49:42 -0500 (Fri, 20 Apr 2012) + | 13 lines 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 ........ + r362869 | twilson | 2012-04-20 11:12:34 -0500 (Fri, 20 Apr 2012) + | 11 lines 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 ........ + r362918 | elguero | 2012-04-20 11:47:51 -0500 (Fri, 20 Apr 2012) + | 11 lines 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) ........ r362998 | + rmudgett | 2012-04-20 20:45:13 -0500 (Fri, 20 Apr 2012) | 5 lines + Update app_dial M and U option GOTO return value documentation. + ........ Merged revisions 362997 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r363212 | tilghman | 2012-04-23 11:06:53 -0500 (Mon, 23 Apr 2012) + | 8 lines 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 ........ + r363376 | rmudgett | 2012-04-24 19:01:21 -0500 (Tue, 24 Apr 2012) + | 5 lines Hangup affected channel in error paths of + bridge_call_thread(). ........ Merged revisions 363375 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r363429 | rmudgett | 2012-04-24 20:23:08 -0500 (Tue, 24 Apr 2012) + | 27 lines 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 ........ + r363688 | rmudgett | 2012-04-25 14:47:44 -0500 (Wed, 25 Apr 2012) + | 19 lines 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 ........ + r363734 | rmudgett | 2012-04-25 15:48:22 -0500 (Wed, 25 Apr 2012) + | 18 lines 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 ........ + r363789 | rmudgett | 2012-04-25 17:59:46 -0500 (Wed, 25 Apr 2012) + | 5 lines Update Pickup application documentation. ........ + Merged revisions 363788 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r363876 | rmudgett | 2012-04-25 22:11:45 -0500 (Wed, 25 Apr 2012) + | 5 lines Update Pickup application documentation. (Even better) + ........ Merged revisions 363875 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r363935 | alecdavis | 2012-04-26 04:46:38 -0500 (Thu, 26 Apr + 2012) | 14 lines 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 ........ + r363987 | kmoore | 2012-04-26 08:27:34 -0500 (Thu, 26 Apr 2012) | + 15 lines 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) Reported by: Maciej Krajewski + Tested by: Maciej Karjewski ........ Merged revisions 363986 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r364047 | twilson | 2012-04-26 14:30:55 -0500 (Thu, 26 Apr 2012) + | 8 lines 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 ........ + r364065 | rmudgett | 2012-04-26 15:25:05 -0500 (Thu, 26 Apr 2012) + | 24 lines 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 ........ + r364109 | rmudgett | 2012-04-26 16:10:46 -0500 (Thu, 26 Apr 2012) + | 5 lines Update Pickup application documentation. (With feeling + this time.) ........ Merged revisions 364108 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r364163 | schmidts | 2012-04-27 07:54:19 -0500 (Fri, 27 Apr 2012) + | 3 lines 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. ........ r364204 | mjordan | + 2012-04-27 09:44:13 -0500 (Fri, 27 Apr 2012) | 23 lines 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 + ........ r364259 | kmoore | 2012-04-27 13:58:34 -0500 (Fri, 27 + Apr 2012) | 14 lines 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 ........ + r364285 | mjordan | 2012-04-27 14:30:19 -0500 (Fri, 27 Apr 2012) + | 43 lines 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 ........ + r364342 | mmichelson | 2012-04-27 16:58:06 -0500 (Fri, 27 Apr + 2012) | 10 lines 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 ........ + 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 ........ r364536 | elguero | + 2012-04-28 21:21:10 -0500 (Sat, 28 Apr 2012) | 13 lines 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/ ........ r364579 | + mjordan | 2012-04-29 14:43:53 -0500 (Sun, 29 Apr 2012) | 15 lines + 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 ........ + r364650 | markm | 2012-04-30 11:43:11 -0500 (Mon, 30 Apr 2012) | + 15 lines 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 ........ ........ r364651 | may | 2012-04-30 11:48:57 + -0500 (Mon, 30 Apr 2012) | 10 lines 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 ........ + r364777 | jrose | 2012-05-01 13:23:08 -0500 (Tue, 01 May 2012) | + 13 lines 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 ........ + r364787 | kmoore | 2012-05-01 14:07:09 -0500 (Tue, 01 May 2012) | + 12 lines 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 ........ + r364845 | rmudgett | 2012-05-01 16:50:32 -0500 (Tue, 01 May 2012) + | 7 lines * 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 ........ + r364900 | mmichelson | 2012-05-01 18:10:16 -0500 (Tue, 01 May + 2012) | 16 lines 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 ........ + r364903 | rmudgett | 2012-05-01 18:14:12 -0500 (Tue, 01 May 2012) + | 10 lines 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 ........ + r364965 | mjordan | 2012-05-01 21:44:15 -0500 (Tue, 01 May 2012) + | 11 lines 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. ........ r365014 | elguero | + 2012-05-02 11:16:03 -0500 (Wed, 02 May 2012) | 18 lines 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 (issue ASTERISK-19760) Reported by: + Michael L. Young Tested by: Michael L. Young Patches: + issue-asterisk-19760-branch10.diff uploaded by Michael L. Young + (license 5026) Review: https://reviewboard.asterisk.org/r/1877/ + ........ r365083 | twilson | 2012-05-02 12:29:54 -0500 (Wed, 02 + May 2012) | 33 lines 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 + 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083 + from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-06-04 Asterisk Development Team + + * Asterisk 10.5.0-digiumphones Released. + +2012-05-30 Asterisk Development Team + + * Asterisk 10.5.0-digiumphones-rc2 Released. + + * Resolve crash in subscribing for MWI notifications. + + ASTOBJ_UNREF sets the variable to NULL after unreffing it, so the + variable shoudl 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/ + + * 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 fucntion, + play_sound_helper, to playback both the 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 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) + + * AST-2012-007 + + * AST-2012-008 + +2012-05-03 Asterisk Development Team + + * Asterisk 10.5.0-digiumphones-rc1 Released. + +2012-05-03 20:06 +0000 [r365264] Jason Parker + + * main/jitterbuf.c, configs/queues.conf.sample, + configs/usbradio.conf.sample (removed), + res/res_calendar_caldav.c, apps/rpt_flow.pdf (removed), + apps/app_queue.c, main/cel.c, res/res_config_sqlite.c, + res/res_calendar_ews.c, main/config.c, formats/format_siren7.c, + channels/chan_dahdi.c, formats/format_vox.c, funcs/func_volume.c, + configure, formats/format_h263.c, main/event.c, + apps/app_chanspy.c, formats/format_g719.c, channels/chan_sip.c, + funcs/func_env.c, channels/chan_agent.c, funcs/func_strings.c, + channels/console_video.c, Makefile.rules, main/astfd.c, + formats/format_wav_gsm.c, bridges/bridge_multiplexed.c, + channels/chan_iax2.c, funcs/func_global.c, + apps/confbridge/conf_config_parser.c, res/res_config_curl.c, + build_tools/cflags.xml, main/cdr.c, funcs/func_curl.c, + main/manager.c, main/tdd.c, channels/console_gui.c, + formats/format_pcm.c, main/app.c, main/stdtime/localtime.c, + utils/extconf.c, makeopts.in, main/message.c, + formats/format_gsm.c, res/res_clioriginate.c, + include/asterisk/time.h, res/res_rtp_asterisk.c, + res/res_config_pgsql.c, apps/app_meetme.c, /, + formats/format_wav.c, configure.ac, res/res_musiconhold.c, + channels/chan_gtalk.c, tests/test_linkedlists.c, apps/app_ices.c, + channels/sig_pri.c, res/res_srtp.c, formats/format_ilbc.c, + channels/sig_pri.h, Makefile, apps/app_forkcdr.c, + res/res_config_odbc.c, bridges/bridge_builtin_features.c, + codecs/gsm/src/k6opt.s, build_tools/menuselect-deps.in, + funcs/func_channel.c, apps/app_directed_pickup.c, + main/features.c, res/res_agi.c, main/http.c, main/logger.c, + apps/app_confbridge.c, apps/app_sms.c, main/audiohook.c, + formats/format_h264.c, apps/app_voicemail.c, + codecs/lpc10/Makefile, apps/app_dial.c, formats/format_sln.c, + codecs/gsm/Makefile, funcs/func_sysinfo.c, + formats/format_ogg_vorbis.c, CHANGES, main/astobj2.c, + main/format_pref.c, apps/app_speech_utils.c, + tests/test_security_events.c, main/tcptls.c, + addons/ooh323cDriver.c, formats/format_g723.c, + apps/app_externalivr.c, tests/test_config.c, tests/test_poll.c, + addons/chan_mobile.c, formats/format_siren14.c, + funcs/func_devstate.c, main/asterisk.c, main/xmldoc.c, + channels/chan_mgcp.c, formats/format_g729.c, + channels/chan_unistim.c, configs/chan_dahdi.conf.sample, + main/pbx.c, res/res_calendar_icalendar.c, channels/chan_local.c, + funcs/func_version.c, configs/rpt.conf.sample (removed): Multiple + revisions + 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083 + ........ r361208 | jrose | 2012-04-04 14:30:09 -0500 (Wed, 04 Apr + 2012) | 10 lines 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 + ........ r361211 | jrose | 2012-04-04 15:00:23 -0500 (Wed, 04 Apr + 2012) | 12 lines 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 ........ + r361270 | jrose | 2012-04-05 11:53:35 -0500 (Thu, 05 Apr 2012) | + 10 lines 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 ........ + r361330 | kmoore | 2012-04-06 08:31:51 -0500 (Fri, 06 Apr 2012) | + 11 lines 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 ........ + r361333 | mjordan | 2012-04-06 09:01:33 -0500 (Fri, 06 Apr 2012) + | 11 lines 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 ........ + r361381 | russell | 2012-04-06 10:49:19 -0500 (Fri, 06 Apr 2012) + | 5 lines 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 ........ + r361422 | pabelanger | 2012-04-06 11:31:18 -0500 (Fri, 06 Apr + 2012) | 14 lines 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 ........ + r361472 | kmoore | 2012-04-06 13:13:04 -0500 (Fri, 06 Apr 2012) | + 5 lines Add missing newlines to CLI logging ........ Merged + revisions 361471 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r361522 | rmudgett | 2012-04-06 14:47:29 -0500 (Fri, 06 Apr 2012) + | 8 lines 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. ........ r361560 | mjordan | 2012-04-06 + 15:32:13 -0500 (Fri, 06 Apr 2012) | 13 lines 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 + ........ r361607 | mjordan | 2012-04-06 17:00:11 -0500 (Fri, 06 + Apr 2012) | 12 lines 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 ........ + r361658 | mjordan | 2012-04-09 14:42:53 -0500 (Mon, 09 Apr 2012) + | 15 lines 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 ........ + r361706 | mjordan | 2012-04-09 15:54:55 -0500 (Mon, 09 Apr 2012) + | 17 lines 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 ........ + r361754 | mjordan | 2012-04-09 16:44:30 -0500 (Mon, 09 Apr 2012) + | 12 lines 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 ........ + r361804 | mjordan | 2012-04-10 14:57:30 -0500 (Tue, 10 Apr 2012) + | 10 lines 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 ........ + r361855 | rmudgett | 2012-04-10 16:47:42 -0500 (Tue, 10 Apr 2012) + | 19 lines 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 ........ + r361907 | jrose | 2012-04-11 11:07:50 -0500 (Wed, 11 Apr 2012) | + 10 lines 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/ ........ r361956 | + kmoore | 2012-04-12 10:01:13 -0500 (Thu, 12 Apr 2012) | 13 lines + 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 ........ + r361981 | kmoore | 2012-04-12 11:22:28 -0500 (Thu, 12 Apr 2012) | + 12 lines 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 + ........ r362080 | jrose | 2012-04-13 10:30:22 -0500 (Fri, 13 Apr + 2012) | 10 lines 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 ........ + r362084 | jrose | 2012-04-13 11:04:22 -0500 (Fri, 13 Apr 2012) | + 15 lines 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 ........ + r362152 | mjordan | 2012-04-16 14:39:32 -0500 (Mon, 16 Apr 2012) + | 19 lines 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 ........ + r362202 | mjordan | 2012-04-16 16:40:29 -0500 (Mon, 16 Apr 2012) + | 18 lines 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 ........ + r362205 | mjordan | 2012-04-16 16:57:19 -0500 (Mon, 16 Apr 2012) + | 25 lines 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 ........ + r362264 | elguero | 2012-04-17 09:53:04 -0500 (Tue, 17 Apr 2012) + | 23 lines 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 ........ + r362305 | mjordan | 2012-04-17 13:27:44 -0500 (Tue, 17 Apr 2012) + | 15 lines 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 ........ + r362356 | mjordan | 2012-04-17 15:56:05 -0500 (Tue, 17 Apr 2012) + | 17 lines 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 ........ + r362357 | jrose | 2012-04-17 15:57:36 -0500 (Tue, 17 Apr 2012) | + 12 lines 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 ........ + r362360 | mjordan | 2012-04-17 16:07:29 -0500 (Tue, 17 Apr 2012) + | 24 lines 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 ........ + r362364 | mjordan | 2012-04-17 16:11:25 -0500 (Tue, 17 Apr 2012) + | 29 lines 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 ........ + r362377 | mjordan | 2012-04-17 16:22:37 -0500 (Tue, 17 Apr 2012) + | 13 lines 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/ + ........ r362429 | rmudgett | 2012-04-18 11:27:51 -0500 (Wed, 18 + Apr 2012) | 19 lines 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 + ........ r362496 | mjordan | 2012-04-18 21:27:08 -0500 (Wed, 18 + Apr 2012) | 50 lines 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 ........ + r362537 | twilson | 2012-04-19 09:31:59 -0500 (Thu, 19 Apr 2012) + | 14 lines 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 ........ + r362587 | seanbright | 2012-04-19 11:04:21 -0500 (Thu, 19 Apr + 2012) | 12 lines 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 ........ + r362678 | rmudgett | 2012-04-19 16:00:21 -0500 (Thu, 19 Apr 2012) + | 5 lines Update membermacro and membergosub documentation in + queues.conf.sample. ........ Merged revisions 362677 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r362681 | elguero | 2012-04-19 16:11:35 -0500 (Thu, 19 Apr 2012) + | 9 lines 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 ........ + r362730 | wdoekes | 2012-04-19 16:59:43 -0500 (Thu, 19 Apr 2012) + | 5 lines Fix documentation for ${VERSION(ASTERISK_VERSION_NUM)}. + ........ Merged revisions 362729 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r362816 | twilson | 2012-04-20 09:49:42 -0500 (Fri, 20 Apr 2012) + | 13 lines 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 ........ + r362869 | twilson | 2012-04-20 11:12:34 -0500 (Fri, 20 Apr 2012) + | 11 lines 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 ........ + r362918 | elguero | 2012-04-20 11:47:51 -0500 (Fri, 20 Apr 2012) + | 11 lines 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) ........ r362998 | + rmudgett | 2012-04-20 20:45:13 -0500 (Fri, 20 Apr 2012) | 5 lines + Update app_dial M and U option GOTO return value documentation. + ........ Merged revisions 362997 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r363212 | tilghman | 2012-04-23 11:06:53 -0500 (Mon, 23 Apr 2012) + | 8 lines 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 ........ + r363376 | rmudgett | 2012-04-24 19:01:21 -0500 (Tue, 24 Apr 2012) + | 5 lines Hangup affected channel in error paths of + bridge_call_thread(). ........ Merged revisions 363375 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r363429 | rmudgett | 2012-04-24 20:23:08 -0500 (Tue, 24 Apr 2012) + | 27 lines 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 ........ + r363688 | rmudgett | 2012-04-25 14:47:44 -0500 (Wed, 25 Apr 2012) + | 19 lines 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 ........ + r363734 | rmudgett | 2012-04-25 15:48:22 -0500 (Wed, 25 Apr 2012) + | 18 lines 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 ........ + r363789 | rmudgett | 2012-04-25 17:59:46 -0500 (Wed, 25 Apr 2012) + | 5 lines Update Pickup application documentation. ........ + Merged revisions 363788 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r363876 | rmudgett | 2012-04-25 22:11:45 -0500 (Wed, 25 Apr 2012) + | 5 lines Update Pickup application documentation. (Even better) + ........ Merged revisions 363875 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r363935 | alecdavis | 2012-04-26 04:46:38 -0500 (Thu, 26 Apr + 2012) | 14 lines 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 ........ + r363987 | kmoore | 2012-04-26 08:27:34 -0500 (Thu, 26 Apr 2012) | + 15 lines 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) Reported by: Maciej Krajewski + Tested by: Maciej Karjewski ........ Merged revisions 363986 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r364047 | twilson | 2012-04-26 14:30:55 -0500 (Thu, 26 Apr 2012) + | 8 lines 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 ........ + r364065 | rmudgett | 2012-04-26 15:25:05 -0500 (Thu, 26 Apr 2012) + | 24 lines 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 ........ + r364109 | rmudgett | 2012-04-26 16:10:46 -0500 (Thu, 26 Apr 2012) + | 5 lines Update Pickup application documentation. (With feeling + this time.) ........ Merged revisions 364108 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ + r364163 | schmidts | 2012-04-27 07:54:19 -0500 (Fri, 27 Apr 2012) + | 3 lines 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. ........ r364204 | mjordan | + 2012-04-27 09:44:13 -0500 (Fri, 27 Apr 2012) | 23 lines 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 + ........ r364259 | kmoore | 2012-04-27 13:58:34 -0500 (Fri, 27 + Apr 2012) | 14 lines 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 ........ + r364285 | mjordan | 2012-04-27 14:30:19 -0500 (Fri, 27 Apr 2012) + | 43 lines 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 ........ + r364342 | mmichelson | 2012-04-27 16:58:06 -0500 (Fri, 27 Apr + 2012) | 10 lines 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 ........ + 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 ........ r364536 | elguero | + 2012-04-28 21:21:10 -0500 (Sat, 28 Apr 2012) | 13 lines 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/ ........ r364579 | + mjordan | 2012-04-29 14:43:53 -0500 (Sun, 29 Apr 2012) | 15 lines + 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 ........ + r364650 | markm | 2012-04-30 11:43:11 -0500 (Mon, 30 Apr 2012) | + 15 lines 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 ........ ........ r364651 | may | 2012-04-30 11:48:57 + -0500 (Mon, 30 Apr 2012) | 10 lines 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 ........ + r364777 | jrose | 2012-05-01 13:23:08 -0500 (Tue, 01 May 2012) | + 13 lines 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 ........ + r364787 | kmoore | 2012-05-01 14:07:09 -0500 (Tue, 01 May 2012) | + 12 lines 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 ........ + r364845 | rmudgett | 2012-05-01 16:50:32 -0500 (Tue, 01 May 2012) + | 7 lines * 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 ........ + r364900 | mmichelson | 2012-05-01 18:10:16 -0500 (Tue, 01 May + 2012) | 16 lines 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 ........ + r364903 | rmudgett | 2012-05-01 18:14:12 -0500 (Tue, 01 May 2012) + | 10 lines 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 ........ + r364965 | mjordan | 2012-05-01 21:44:15 -0500 (Tue, 01 May 2012) + | 11 lines 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. ........ r365014 | elguero | + 2012-05-02 11:16:03 -0500 (Wed, 02 May 2012) | 18 lines 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 (issue ASTERISK-19760) Reported by: + Michael L. Young Tested by: Michael L. Young Patches: + issue-asterisk-19760-branch10.diff uploaded by Michael L. Young + (license 5026) Review: https://reviewboard.asterisk.org/r/1877/ + ........ r365083 | twilson | 2012-05-02 12:29:54 -0500 (Wed, 02 + May 2012) | 33 lines 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 + 361208,361211,361270,361330,361333,361381,361422,361472,361522,361560,361607,361658,361706,361754,361804,361855,361907,361956,361981,362080,362084,362152,362202,362205,362264,362305,362356-362357,362360,362364,362377,362429,362496,362537,362587,362678,362681,362730,362816,362869,362918,362998,363212,363376,363429,363688,363734,363789,363876,363935,363987,364047,364065,364109,364163,364204,364259,364285,364342,364365,364369,364536,364579,364650-364651,364777,364787,364845,364900,364903,364965,365014,365083 + from http://svn.asterisk.org/svn/asterisk/branches/10 + +2012-05-02 Asterisk Development Team + + * Asterisk 10.4.0 Released. + +2012-05-01 Asterisk Development Team + + * Asterisk 10.4.0-rc3 Released. + + * 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) + +2012-04-24 Asterisk Development Team + + * Asterisk 10.4.0-rc2 Released. + + * AST-2012-004 + + * AST-2012-005 + + * AST-2012-006 + +2012-04-04 Asterisk Development Team + + * Asterisk 10.4.0-rc1 Released. + +2012-04-04 16:38 +0000 [r361091-361143] Jonathan Rose + + * 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) + + * /, 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 + +2012-04-03 20:08 +0000 [r360993-361041] 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 + + * /, 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 + +2012-03-30 21:29 +0000 [r360934] 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 + +2012-03-29 23:33 +0000 [r360863-360885] 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 + + * /, 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 + +2012-03-28 19:20 +0000 [r360717] Terry Wilson + + * channels/chan_jingle.c, addons/chan_ooh323.c, /, + cdr/cdr_adaptive_odbc.c, addons/cdr_mysql.c, + channels/chan_gtalk.c: Destroy configs when they are no longer + used https://reviewboard.asterisk.org/r/1834/ ........ Merged + revisions 360712 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2012-03-27 18:23 +0000 [r360672] 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 + +2012-03-27 14:35 +0000 [r360489-360575] 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 + + * /, 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 + +2012-03-26 18:41 +0000 [r360472-360476] Paul Belanger + + * /, CHANGES: Update CHANGES for r360471 ........ Merged revisions + 360474 from http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * main/dnsmgr.c, /: Increase verbosity level for ast_verb messages + While this does not fix the issue of the CLI being flooded by + 'doing dnsmgr_lookup' messages, increasing the verbosity level + above 5 should help minimize it. ........ Merged revisions 360471 + from http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2012-03-24 23:47 +0000 [r360358-360414] 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 + + * /, 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 + + * 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 + + * main/ast_expr2.c, /, 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: 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 + +2012-03-24 00:37 +0000 [r360263-360310] 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 + + * /, 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 + +2012-03-21 14:52 +0000 [r360139] 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 + +2012-03-21 14:21 +0000 [r360098] 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 + +2012-03-21 13:28 +0000 [r360088] 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 + +2012-03-20 20:37 +0000 [r360034] 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 + +2012-03-20 18:11 +0000 [r359982] Sean Bright + + * channels/chan_iax2.c: chan_iax2: Emit Port alongside Post in + PeerStatus AMI Event. The PeerStatus event for IAX2 channels + currently includes a header named Post which should have been + Port. So include Port along with Post when emitting the event. + We'll remove Post in trunk. + +2012-03-20 17:25 +0000 [r359980] Richard Mudgett + + * main/manager.c, /, include/asterisk/manager.h: 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 + +2012-03-16 20:20 +0000 [r359898] 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 + +2012-03-16 08:24 +0000 [r359810] 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 + +2012-03-15 19:06 +0000 [r359694-359707] 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 + + * /, 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 + +2012-03-15 18:22 +0000 [r359620] 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 + +2012-03-15 00:53 +0000 [r359454-359559] 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 + + * /, 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 + + * /, 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 + + * 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 + + * 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 + +2012-03-14 22:28 +0000 [r359453] 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 + +2012-03-14 17:42 +0000 [r359358] 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 + +2012-03-14 17:24 +0000 [r359355] 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 + +2012-03-14 10:54 +0000 [r359051-359260] 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 + + * /, 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 + + * 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 + + * /, utils/astman.c: Fix incorrect sizeof() in astman. ........ + Merged revisions 359116 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * /, 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 + + * /, 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 + + * /, 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 + + * /, main/md5.c: Fix incorrect sizeof() on a pointer in MD5Final(). + ........ Merged revisions 359059 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * 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 + + * res/res_ais.c, /, res/ais/clm.c, res/ais/evt.c, res/ais/ais.h: + Dump cache of published events when a node joins the cluster. + Also use a more reliable method for stopping the poll() thread. + ........ Merged revisions 359053 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * 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 + +2012-03-13 20:36 +0000 [r358944-358989] 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 + + * 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 + +2012-03-13 16:58 +0000 [r358811-358860] Tilghman Lesher + + * /, UPGRADE.txt, CHANGES: Requested changes documenting the fixed + AEL functionality. ........ Merged revisions 358859 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * 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 + +2012-03-08 16:50 +0000 [r358644] 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 + +2012-03-07 18:28 +0000 [r358531] 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 + +2012-03-07 16:13 +0000 [r358485] 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 + +2012-03-07 15:17 +0000 [r358436-358441] 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 + + * 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 + +2012-03-06 17:46 +0000 [r358261-358378] 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 + + * /, 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 + + * 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 + +2012-03-05 18:58 +0000 [r358215] 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 + +2012-03-05 16:42 +0000 [r358163] 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 + +2012-03-05 15:59 +0000 [r358116] 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) Patch-by: + Jeremiah Gowdy ........ Merged revisions 358115 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2012-03-02 23:28 +0000 [r357987-358033] 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 + + * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c, + channels/misdn/isdn_lib.c: 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 + + * /, 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 + +2012-03-02 21:03 +0000 [r357941] 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 + +2012-03-02 18:37 +0000 [r357895] 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 + +2012-03-02 15:59 +0000 [r357812] 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 + +2012-03-02 15:50 +0000 [r357810] 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 + +2012-03-02 01:05 +0000 [r357762] 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 + +2012-03-01 14:18 +0000 [r357667] 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 + +2012-02-29 20:39 +0000 [r357576-357620] Walter Doekes + + * include/asterisk/stringfields.h, main/utils.c: 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. + + * 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) 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 + +2012-02-28 22:29 +0000 [r357458-357497] 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 + + * /, 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 + +2012-02-28 21:21 +0000 [r357421] 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 + +2012-02-28 20:58 +0000 [r357408] 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 + +2012-02-28 20:42 +0000 [r357357-357405] Jonathan Rose + + * UPGRADE.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 + + * /, UPGRADE.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 + + * /, 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 + +2012-02-28 19:35 +0000 [r357353] 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 + +2012-02-28 18:51 +0000 [r357318] Mark Michelson + + * channels/sip/security_events.c: Add a security event for the case + where fake authentication challenge is sent. + +2012-02-28 18:11 +0000 [r357271] 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 + +2012-02-28 14:46 +0000 [r357213] 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 + +2012-03-29 Asterisk Development Team + + * Asterisk 10.3.0 Released. + +2012-03-26 Asterisk Development Team + + * Asterisk 10.3.0-rc3 Released. + + * AST-2012-003 + + * AST-2012-002 + + * /main/manager.c, include/asterisk/manager.h: Fix AMI deadlock + regression by allowing AMI action callback to be reentrant + + Fix AMI module reload deadlock from ASTERISK-18479 when it tired to + fix the race between calling an AMI action callback and + unregistering that action. Refixes ASTERISK-13874 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 path 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/ + +2012-03-06 Asterisk Development Team + + * Asterisk 10.3.0-rc2 Released. + + * 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. + +2012-01-30 22:16 +0000 [r353369-353321] Alec L Davis + + * channels/sip/include/dialog.h, /, channels/chan_sip.c, + channels/sip/include/sip.h: 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/ ........ + + * /, channels/chan_sip.c: 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 ........ + +2012-01-30 23:28 +0000 [r353397] 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 + +2012-01-31 17:21 +0000 [r353463] 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 + +2012-02-01 00:00 +0000 [r353503] 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 + +2012-02-01 15:05 +0000 [r353551] 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 + +2012-02-01 15:51 +0000 [r353599] 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 + +2012-02-01 21:16 +0000 [r353771-353721] Jonathan Rose + + * /, 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 + + * /, 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 + +2012-02-02 18:48 +0000 [r353820] Mark Michelson + + * configs/http.conf.sample, main/manager.c, /, main/http.c, + configs/manager.conf.sample, include/asterisk/manager.h: 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 + +2012-02-02 20:11 +0000 [r353868] 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 + +2012-02-02 22:27 +0000 [r353916] 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 + +2012-02-03 16:22 +0000 [r354000-353962] 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 + + * /, 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 + +2012-02-06 17:31 +0000 [r354217-354119] 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 + + * /, 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 + +2012-02-07 15:19 +0000 [r354270] 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 + +2012-02-07 21:17 +0000 [r354349] 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 + +2012-02-09 02:25 +0000 [r354493] 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 + +2012-02-09 02:54 +0000 [r354496] Richard Mudgett + + * 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 + +2012-02-09 16:35 +0000 [r354543] Matthew Jordan + + * /, 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 + +2012-02-09 17:04 +0000 [r354546] 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 + +2012-02-09 17:08 +0000 [r354548] 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 + +2012-02-09 19:54 +0000 [r354703-354656] Kinsey Moore + + * /, 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 + + * /, 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 + +2012-02-09 22:03 +0000 [r354750] 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 + +2012-02-10 18:05 +0000 [r354836] 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 + +2012-02-10 22:00 +0000 [r354890] 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 + +2012-02-13 16:41 +0000 [r354938] Joshua Colp + + * apps/app_confbridge.c: Don't try to play sound files that do not + exist. (closes issue ASTERISK-19188) Reported by: slesru + +2012-02-13 17:24 +0000 [r354959] 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 + +2012-02-13 19:51 +0000 [r355010] 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 + +2012-02-13 22:03 +0000 [r355057] 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 + +2012-02-14 09:49 +0000 [r355137] 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 + +2012-02-14 13:33 +0000 [r355183] 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 + +2012-02-14 15:53 +0000 [r355229] 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 + +2012-02-14 16:27 +0000 [r355271] 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 + +2012-02-14 18:14 +0000 [r355375-355320] Richard Mudgett + + * /, 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 + + * /, 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 + +2012-02-15 17:25 +0000 [r355530-355449] Sean Bright + + * /, 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 + + * /, channels/chan_iax2.c: Only use maxtrunkcall and + maxnontrunkcall in chan_iax2 if IAX_OLD_FIND is specified. These + variables are only accessed from the IAX_OLD_FIND path, so there + is no reason to keep them updated otherwise. ........ Merged + revisions 355458 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * /, 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 + +2012-02-16 18:32 +0000 [r355620-355575] Richard Mudgett + + * /, 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 + + * /, 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 + +2012-02-16 20:01 +0000 [r355623] 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 + +2012-02-17 19:06 +0000 [r355733] 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 + +2012-02-17 19:34 +0000 [r355794-355747] 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 + + * 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 + +2012-02-18 07:58 +0000 [r355851] 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 + +2012-02-18 17:02 +0000 [r355896-355895] Paul Belanger + + * /: Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) + ........ Merged revisions 355839 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * /: Revert commit + +2012-02-19 17:50 +0000 [r355998-355902] Sean Bright + + * /, 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 + + * /, 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 + + * 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 + + * 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 + + * /, 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 + +2012-02-21 04:30 +0000 [r356074] Kinsey Moore + + * main/ccss.c: Add missing newline to ccss state change + notification Move along, nothing to see here... + +2012-02-21 11:17 +0000 [r356108] 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 + +2012-02-22 14:53 +0000 [r356215] Matthew Jordan + + * /, channels/chan_sip.c: 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 21:18 +0000 [r356297] Terry Wilson + + * main/loader.c, res/res_calendar.c, /, + include/asterisk/calendar.h: 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 + +2012-02-23 03:23 +0000 [r356431-356428] 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 + + * addons/chan_ooh323.c, /: Fix -Werror=unused-but-set-variable + compiler error (gcc 4.6.2) ........ Merged revisions 356430 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2012-02-23 15:40 +0000 [r356476] 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 + +2012-02-23 19:52 +0000 [r356522] 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 + +2012-02-24 15:07 +0000 [r356651-356605] Matthew Jordan + + * res/res_srtp.c, channels/sip/sdp_crypto.c, + include/asterisk/res_srtp.h, main/rtp_engine.c, /, + include/asterisk/rtp_engine.h: 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 + + * 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 + +2012-02-24 18:27 +0000 [r356690] Richard Mudgett + + * /, 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 + +2012-02-25 17:22 +0000 [r356798] 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 + +2012-02-27 15:30 +0000 [r356961] 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 + +2012-02-27 16:05 +0000 [r356964] 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 + +2012-02-27 23:36 +0000 [r357095] Richard Mudgett + + * 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 + +2012-03-05 Asterisk Development Team + + * Asterisk 10.2.0 Released. + +2012-03-01 Asterisk Development Team + + * Asterisk 10.2.0-rc4 Released. + + * 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. + +2012-02-28 Asterisk Development Team + + * Asterisk 10.2.0-rc3 Released. + + * main/channel.c: Fix callerid of Originated calls. + + The callerid of originated calls (independent of mechanism) was not + being passed to the outbound channel. This patch fixes that. Thanks + to Matt Riddell for tracking this down. + (closes issue ASTERISK-19385) + Reported by: ornix + + * 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) + + * 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 + + +2012-02-10 Asterisk Development Team + + * Asterisk 10.2.0-rc2 Released. + + * channels/chan_sip.c: Fix SIP INFO DTMF handling for non-numeric + codes. In ASTERISK-18924, SIP INFO DTMF handling was 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 against '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 + + * apps/app_parkandannounce.c: Fix crash in ParkAndAnnounce from + uninitialized caller_id storage (closes issue ASTERISK-19311) + Reported by: tootai + Tested by: rmudgett + + * 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/ + + * 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 + + * channels/chan_sip.c,include/asterisk/dnsmgr.h,main/dnsmgr.c: 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. + +2012-02-01 Asterisk Development Team + + * Asterisk 10.2.0-rc1 Released. + + * Test results: + http://bamboo.asterisk.org/browse/TESTING-ASTERISK1020RCS-2 + +2012-01-30 12:48 +0000 [r353261] Kevin P. Fleming + + * /, 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 + +2012-01-29 02:44 +0000 [r353176] 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 + +2012-01-28 14:51 +0000 [r353127] 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 + +2012-01-28 04:27 +0000 [r353078] 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 + +2012-01-27 21:37 +0000 [r352992-353039] Richard Mudgett + + * apps/app_queue.c: Audit of ao2_iterator_init() usage for v10. + Missed one. + + * tests/test_format_api.c: Audit of ao2_iterator_init() usage for + v10. Fix double format_cap iterator cleanup. + +2012-01-27 19:19 +0000 [r352965] 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 + +2012-01-27 18:36 +0000 [r352956] Richard Mudgett + + * res/res_srtp.c, 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: 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 + +2012-01-27 00:08 +0000 [r352863] Alec L Davis + + * /, channels/chan_sip.c, channels/sip/include/sip.h: 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 20:22 +0000 [r352817] 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 + +2012-01-26 19:07 +0000 [r352756] 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 + +2012-01-26 06:33 +0000 [r352705] Alec L Davis + + * /, channels/chan_sip.c: 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 22:23 +0000 [r352651] 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 + +2012-01-25 21:18 +0000 [r352616] Kevin P. Fleming + + * /, main/test.c: Avoid unnecessary rebuilds of main/test.c. + main/test.c includes "asterisk/version.h", when it should include + "asterisk/ast_version.h" instead (and it should use the + ast_get_version() and ast_get_version_num() functions). This + commit modifies it to extract the Asterisk version information + using the proper APIs, and as a result means that main/test.c no + longer needs to be rebuilt when a Subversion checkout is updated + or modified. ........ Merged revisions 352612 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2012-01-25 17:30 +0000 [r352556] 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 + +2012-01-25 17:16 +0000 [r352520] 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/ + +2012-01-25 16:54 +0000 [r352516] 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. + +2012-01-25 16:49 +0000 [r352515] 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 + +2012-01-25 16:41 +0000 [r352512] 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 + +2012-01-24 22:22 +0000 [r352430] 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 + +2012-01-24 20:35 +0000 [r352373] 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 + +2012-01-24 17:02 +0000 [r352292] 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 + +2012-01-23 20:30 +0000 [r352228-352231] Mark Michelson + + * /, main/features.c: Fix grammar of comment. ........ Merged + revisions 352230 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * /, 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 + +2012-01-23 19:18 +0000 [r352149] 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 + +2012-01-23 17:34 +0000 [r352091] 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 + +2012-01-21 00:21 +0000 [r352035] 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 + +2012-01-21 00:08 +0000 [r352015-352017] 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 + + * /, 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 + +2012-01-20 19:35 +0000 [r351816-351861] 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 + + * /: Recorded merge of revisions 351858 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Allow + ilbc code to build under dev mode GCC 4.6.3 found some set/unused + variables in the ILBC code. + + * 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. + + * 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. + +2012-01-20 15:59 +0000 [r351762] 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 + +2012-01-20 15:54 +0000 [r351761] 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 + +2012-01-20 13:01 +0000 [r351708] 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 + +2012-01-19 23:25 +0000 [r351646] 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 + +2012-01-19 22:43 +0000 [r351612] 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 + +2012-01-19 21:47 +0000 [r351560] 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 + +2012-01-19 21:12 +0000 [r351505] 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 + +2012-01-18 21:05 +0000 [r351451] Matthew Jordan + + * 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/hpInput.c + (added), codecs/ilbc/gainquant.c (added), codecs/ilbc/iCBSearch.h + (added), codecs/ilbc/hpOutput.c (added), codecs/ilbc/rfc3951.txt + (added), codecs/ilbc/filter.h (added), codecs/ilbc/hpInput.h + (added), codecs/ilbc/LPCencode.c (added), codecs/ilbc/gainquant.h + (added), codecs/codec_ilbc.c, codecs/ilbc/hpOutput.h (added), + codecs/ilbc/StateSearchW.c (added), codecs/ilbc/PATENTS (added), + contrib/scripts/get_ilbc_source.sh, codecs/ilbc/LPCencode.h + (added), codecs/ilbc/LICENSE (added), codecs/ilbc/StateSearchW.h + (added), codecs/ilbc/iCBConstruct.c (added), + 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/enhancer.h (added), codecs/ilbc/FrameClassify.c + (added), codecs/ilbc/iLBC_define.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): 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 + +2012-01-18 15:57 +0000 [r351408] 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 + +2012-01-17 17:22 +0000 [r351308] 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 + +2012-01-17 17:08 +0000 [r351289] 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 + +2012-01-17 16:54 +0000 [r351286] 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 + +2012-01-17 16:07 +0000 [r351234] 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 + +2012-01-17 01:43 +0000 [r351183] Russell Bryant + + * /, channels/chan_sip.c: 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 21:17 +0000 [r351081-351131] 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 + + * /, 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 + +2012-01-16 19:12 +0000 [r351028] 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 + +2012-01-16 17:11 +0000 [r350978] 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. + +2012-01-16 17:06 +0000 [r350976] 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 + +2012-01-16 14:27 +0000 [r350938] 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. + +2012-01-15 20:12 +0000 [r350886-350889] 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 + + * /, 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 + +2012-01-14 16:41 +0000 [r350790-350838] 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 + + * contrib/scripts/install_prereq, /, channels/chan_h323.c, + addons/chan_mobile.c, res/res_pktccops.c: 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 + +2012-01-13 22:10 +0000 [r350737] 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 + +2012-01-13 21:51 +0000 [r350734] 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 + +2012-01-13 21:41 +0000 [r350731] 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 + +2012-01-13 20:31 +0000 [r350680] 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 + +2012-01-13 17:29 +0000 [r350585] Richard Mudgett + + * 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 + +2012-01-13 16:59 +0000 [r350550-350553] 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 + + * 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/ + +2012-01-12 16:04 +0000 [r350502] 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 + +2012-01-11 22:51 +0000 [r350312-350453] 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 + + * 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 + + * /, 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 + +2012-01-09 21:55 +0000 [r350221] 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 + +2012-01-09 19:34 +0000 [r350180] 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 + +2012-01-09 18:57 +0000 [r350076-350129] 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 + + * /, 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 + +2012-01-09 15:39 +0000 [r350024] 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 + +2012-01-06 23:25 +0000 [r349977] 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 + +2012-01-06 21:25 +0000 [r349928] 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) + +2012-01-06 16:48 +0000 [r349873] 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 + +2012-01-05 23:56 +0000 [r349822] 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. + +2012-01-05 23:46 +0000 [r349820] 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 + +2012-01-05 22:10 +0000 [r349732] 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 + +2012-01-05 21:55 +0000 [r349673-349729] 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 + + * /, 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 + +2012-01-04 22:19 +0000 [r349608-349619] 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) + + * 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 + +2012-01-04 20:50 +0000 [r349559] 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 + +2012-01-04 20:23 +0000 [r349505-349532] 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 + + * contrib/scripts/autosupport, /, contrib/scripts/autosupport.8: + 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 + +2012-01-04 19:44 +0000 [r349451-349502] 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 + + * 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 + +2012-01-27 Asterisk Development Team + + * Asterisk 10.1.0 Released. + + * Test results: + http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-4 + +2012-01-24 Asterisk Development Team + + * Asterisk 10.1.0-rc2 Released. + + * Test results: + http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-3 + + * 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. (closes issue ASTERISK-18994) Patch-by: Timo Teras + + * channels/chan_sip.c: AST-2012-001: 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 + + * 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 leak for the RTP instance of the transferer. + (closes issue ASERISK-19192) Reported by: Tyuta Vitali + + * res/res_rtp_asterisk: Add pjmedia probation concepts to + res_rtp_asterisk's learning mode. In order to better handle RTP + sources with strictrtp enabled (which is the default setting 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. + + * 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 issue ASTERISK-19173) Reported + by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by + Mark Michelson (license 5049) + + * apps/app_queue.c: Realtime queues failed to load queue + information without queue member table. Revision 342223 + broke this when it changed the return value for + realtime_multientry to return NULL when no results are + returned. (closes issue ASTERISK-19170) Reported by: Rene + Mendoza Tested by: Rene Mendoza + +2011-12-30 Asterisk Development Team + + * Asterisk 10.1.0-rc1 Released. + + * Test results: + http://bamboo.asterisk.org/browse/TESTING-ASTERISK1010RCS-1 + +2011-12-29 15:14 +0000 [r349340] 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 + +2011-12-28 21:33 +0000 [r349290] 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 + +2011-12-28 19:00 +0000 [r349248-349250] Kevin P. Fleming + + * utils: Tell Subversion to gnore the 'astdb2bdb' binary file if it + exists. + + * main/dsp.c, res/res_fax.c, include/asterisk/dsp.h, + include/asterisk/res_fax.h, res/res_fax_spandsp.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). + +2011-12-27 20:53 +0000 [r349195] 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 + +2011-12-27 17:17 +0000 [r349145] 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 + +2011-12-23 17:32 +0000 [r349045] Sean Bright + + * /, apps/app_chanspy.c: 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 15:25 +0000 [r348993] 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 + +2011-12-23 02:30 +0000 [r348952] 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 + +2011-12-22 22:37 +0000 [r348846-348889] 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 + + * 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) + +2011-12-22 20:17 +0000 [r348845] 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 + +2011-12-21 20:13 +0000 [r348736-348793] Richard Mudgett + + * codecs/speex: Make codecs/speex ignore *.i files also. + + * apps/confbridge: Make apps/confbridge ignore *.i files also. + + * /, 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 + +2011-12-19 21:37 +0000 [r348648] 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 + +2011-12-19 19:48 +0000 [r348605] 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) + +2011-12-18 18:28 +0000 [r348517] 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 + +2011-12-16 23:56 +0000 [r348311-348465] 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 + + * 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 + + * 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 + + * 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 + +2011-12-14 22:34 +0000 [r348265] 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. + +2011-12-14 22:05 +0000 [r348213] 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 + +2011-12-14 21:58 +0000 [r348211] Matthew Jordan + + * 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. + +2011-12-14 20:35 +0000 [r348155-348158] 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 + + * /, 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 + +2011-12-13 23:06 +0000 [r348102] 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 + +2011-12-13 15:20 +0000 [r348056] 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 + +2011-12-12 19:24 +0000 [r347996] 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 + +2011-12-12 18:13 +0000 [r347953-347955] Richard Mudgett + + * configs/extensions.conf.sample, 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: Reverting -r347953 for ASTERISK-14122 + + * configs/extensions.conf.sample, 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: 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/ + +2011-12-09 01:29 +0000 [r347812] 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 + +2011-12-08 21:31 +0000 [r347727] 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 + +2011-12-08 20:43 +0000 [r347656] 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/ + +2011-12-08 17:53 +0000 [r347600] 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 + +2011-12-08 16:20 +0000 [r347532] 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 + +2011-12-07 21:39 +0000 [r347439] 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 + +2011-12-07 20:27 +0000 [r347383] 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 + +2011-12-07 20:12 +0000 [r347344] 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/ + +2011-12-06 23:56 +0000 [r347293] 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 + +2011-12-06 21:53 +0000 [r347240] 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 + +2011-12-06 19:42 +0000 [r347124-347167] 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 + + * 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 + + * apps/app_voicemail.c, /: Move setting of voicemail zonetag and + locale up a bit. 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 + +2011-12-06 17:24 +0000 [r347068] 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 + +2011-12-05 17:42 +0000 [r347007] 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 + +2011-12-05 15:02 +0000 [r346955] 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 + +2011-12-05 14:46 +0000 [r346952] 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 + +2011-12-04 10:03 +0000 [r346900] 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 + +2011-12-02 23:27 +0000 [r346856] 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/ + +2011-12-02 16:42 +0000 [r346763] Alexandr Anikin + + * addons/chan_ooh323.c, /, channels/chan_h323.c: 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 21:14 +0000 [r346701] 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 + +2011-12-01 20:37 +0000 [r346565-346698] 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 + + * 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 + +2011-11-30 19:37 +0000 [r346473] 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 + +2011-11-29 00:00 +0000 [r346349] David Vossel + + * include/asterisk/message.h, main/message.c: 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 14:32 +0000 [r346293] 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 + +2011-11-23 22:58 +0000 [r346240] 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 + +2011-11-23 20:45 +0000 [r346145-346198] 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 + + * /, 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 + +2011-11-23 17:14 +0000 [r346087] 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 + +2011-11-23 16:19 +0000 [r346040] 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/ + +2011-11-23 16:10 +0000 [r346031] 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 + +2011-11-23 16:06 +0000 [r346029] Paul Belanger + + * res/res_format_attr_celt.c, res/res_format_attr_silk.c: Added + support level for new modules + +2011-11-22 23:00 +0000 [r345977] 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 + +2011-11-22 20:31 +0000 [r345924] 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 + +2011-11-22 16:40 +0000 [r345882] 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 + +2011-11-21 21:07 +0000 [r345830] 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 + +2011-11-19 15:10 +0000 [r345640-345683] 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 + + * 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 + +2011-11-17 17:29 +0000 [r345558] 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 + +2011-11-16 14:49 +0000 [r345488] 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 + +2011-11-15 20:10 +0000 [r345220-345432] 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 + + * /, 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 + + * /, 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 + + * /, 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 + + * 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 + +2011-11-14 19:10 +0000 [r345164] 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 + +2011-11-14 19:01 +0000 [r345161] 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 + +2011-11-14 16:12 +0000 [r345117] 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 + +2011-11-14 15:10 +0000 [r345064] 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 + +2011-11-12 16:17 +0000 [r344966] 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 + +2011-11-12 00:34 +0000 [r344900] 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 + +2011-11-11 21:58 +0000 [r344845] Walter Doekes + + * include/asterisk/stringfields.h, include/asterisk/utils.h, /, + main/utils.c: 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 + +2011-11-11 21:50 +0000 [r344842] 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 + +2011-11-11 21:36 +0000 [r344836-344839] 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 + + * 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 + +2011-11-11 20:14 +0000 [r344770] 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 + +2011-11-11 18:36 +0000 [r344662-344716] 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 + + * 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 + +2011-11-11 15:33 +0000 [r344609] 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 + +2011-11-10 23:21 +0000 [r344537-344557] Richard Mudgett + + * apps/app_macro.c: Fix app_macro.c MODULEINFO section termination. + (closes issue ASTERISK-18848) Reported by: Tony Mountifield + + * /, 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 + + * /, 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 + +2011-11-10 21:54 +0000 [r344493] David Vossel + + * main/bridging.c: 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 21:14 +0000 [r344386-344440] 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 + + * /, 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 + +2011-11-10 16:25 +0000 [r344334] 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 + +2011-11-09 20:53 +0000 [r344271] 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 + +2011-11-09 20:07 +0000 [r344175-344216] Terry Wilson + + * channels/sip/include/reqresp_parser.h, /, channels/chan_sip.c, + channels/sip/reqresp_parser.c, channels/sip/include/sip.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 + + * /, 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 + +2011-11-09 19:02 +0000 [r344159-344160] Alexandr Anikin + + * /: delete svn:mergeinfo + + * addons/ooh323c/src/ooq931.c, 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: + 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 + +2011-11-09 17:14 +0000 [r344103] 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 + +2011-11-09 15:26 +0000 [r344049] 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 + +2011-11-08 21:59 +0000 [r344004] 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/ + +2011-12-15 Asterisk Development Team + + * Asterisk 10.0.0 Released. + +2011-12-08 Asterisk Development Team + + * Asterisk 10.0.0-rc3 Released. + + * 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/ + + * 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) + + * 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) + Review: https://reviewboard.asterisk.org/r/1610/ + + * 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) + + * Default to nat=yes; warn when nat in general and peer differ + + AST-2011-013. 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/ + + * 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/ + +2011-11-15 Asterisk Development Team + + * Asterisk 10.0.0-rc2 Released. + + * Test results: http://bamboo.asterisk.org/browse/TESTING-ASTERISK10RCS-3 + + * Ensure that a null vmexten does not cause a segfault + + 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 + + * 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 + + * Fix app_macro.c MODULEINFO section termination. + + (closes issue ASTERISK-18848) + Reported by: Tony Mountifield + +2011-11-08 Asterisk Development Team + + * Asterisk 10.0.0-rc1 Released. + + * Test results: http://bamboo.asterisk.org/browse/AST10-LUCID-317 + +2011-11-08 19:27 +0000 [r343944] wdoekes : + + * /, 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 + +2011-11-08 18:29 +0000 [r343900] David Vossel + + * channels/chan_sip.c: 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 18:01 +0000 [r343852] 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 + +2011-11-08 13:26 +0000 [r343789-343792] Leif Madsen + + * /: Recorded merge of revisions 343791 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ 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. + + * 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. + +2011-11-07 22:37 +0000 [r343743] 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) + +2011-11-07 21:42 +0000 [r343691] 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 + +2011-11-07 21:27 +0000 [r343677] 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 + +2011-11-07 20:31 +0000 [r343635] 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 + +2011-11-07 19:55 +0000 [r343580] wdoekes : + + * 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 + +2011-11-07 19:51 +0000 [r343578] 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 + +2011-11-07 18:39 +0000 [r343533] Matthew Nicholson + + * main/format.c: list all of the codecs associated with a + particular format id for CLI command "core show codec" AST-699 + +2011-11-04 15:11 +0000 [r343445] Alexandr Anikin + + * addons/ooh323c/src/ooq931.c, 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: + 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 + +2011-11-03 20:31 +0000 [r343393] wdoekes : + + * /, 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 + +2011-11-03 19:57 +0000 [r343337] 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 + +2011-11-03 15:39 +0000 [r343221-343277] 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 + + * 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 + +2011-11-02 22:24 +0000 [r343158-343192] wdoekes : + + * /, 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 + + * include/asterisk/stringfields.h, include/asterisk/utils.h, /, + main/utils.c: 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 + +2011-11-02 19:33 +0000 [r343048-343103] 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 + + * /, 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 + +2011-11-02 13:45 +0000 [r342991] 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 + +2011-11-01 20:58 +0000 [r342870-342929] wdoekes : + + * /, 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 + + * contrib/realtime/mysql/sipfriends.sql (removed), + 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: 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 + +2011-10-31 17:46 +0000 [r342824] 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(). + +2011-10-31 16:04 +0000 [r342770] 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 + +2011-10-30 02:21 +0000 [r342715] Terry Wilson + + * res/res_calendar.c: Don't crash on empty notify channel + +2011-10-29 04:26 +0000 [r342662] Richard Mudgett + + * /, 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 + +2011-10-27 20:10 +0000 [r342605] Matthew Nicholson + + * main/dsp.c: tweak the v21 detector to detect an additional + pattern of hits and misses + +2011-10-27 19:41 +0000 [r342546-342603] 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 + + * /, 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 + +2011-10-25 22:05 +0000 [r342485-342488] 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 + + * /, 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 + +2011-10-25 21:10 +0000 [r342381-342436] 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 + + * /, 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 + + * /, 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 + +2011-10-25 19:08 +0000 [r342277-342329] 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 + + * pbx/pbx_spool.c, /: 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 01:25 +0000 [r342224] 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 + +2011-10-24 22:32 +0000 [r342183] Richard Mudgett + + * include/asterisk/astobj2.h: Fix ao2obj.h comment typos and add + missing link/unlink nolock debug defines. + +2011-10-24 19:51 +0000 [r342062] 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 + +2011-10-24 07:31 +0000 [r341920-342017] Gregory Nietsky + + * apps/app_queue.c: queues container needs locking when using the + OBJ_NOLOCK flag + + * 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. + + * /, apps/app_queue.c: Revert Janitor patch 341920 For now + + * /, apps/app_queue.c: Whitespace Fixups / Add Braces This + janitorial patch is related to work on RB1538 ........ Merged + revisions 341906 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2011-10-21 16:42 +0000 [r341807-341810] 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 + + * /, 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 + +2011-10-20 21:58 +0000 [r341718] Richard Mudgett + + * include/asterisk/features.h, /, main/features.c, res/res_agi.c: + 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 + +2011-10-20 21:27 +0000 [r341665-341707] Paul Belanger + + * /, funcs/func_callerid.c: Fixed typo from previous commit + ........ Merged revisions 341704 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * /, 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 + +2011-10-20 18:20 +0000 [r341580-341599] Gregory Nietsky + + * configs/queues.conf.sample: 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: 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 15:14 +0000 [r341530] 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 + +2011-10-19 21:23 +0000 [r341486] Matthew Nicholson + + * apps/app_queue.c: 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 19:01 +0000 [r341436] 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 + +2011-10-19 07:42 +0000 [r341380] 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 + +2011-10-19 07:23 +0000 [r341377] 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 + +2011-10-18 23:42 +0000 [r341315] 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 + +2011-10-18 23:33 +0000 [r341313] Alexandr Anikin + + * addons/chan_ooh323.c, /: 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) ........ + +2011-10-18 21:11 +0000 [r341255] Richard Mudgett + + * channels/chan_mgcp.c, include/asterisk/features.h, + channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_sip.c, main/features.c, channels/chan_iax2.c, + channels/sip/include/sip.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 + +2011-10-17 17:36 +0000 [r341190] 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 + +2011-10-17 16:53 +0000 [r341148] 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 + +2011-10-17 16:38 +0000 [r341122-341146] Paul Belanger + + * tests/test_format_api.c: Set 'core' support level for + test_format_api.c + + * 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 + +2011-10-17 16:18 +0000 [r341094] Jason Parker + + * CHANGES: Add information about limitations of new codec support + in channel drivers. (issue ASTERISK-18680) + +2011-10-17 15:39 +0000 [r341089] 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 + +2011-10-14 21:36 +0000 [r341023] 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 + +2011-10-14 20:50 +0000 [r340971] Kinsey Moore + + * res/res_rtp_asterisk.c, /, channels/chan_sip.c: 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 18:23 +0000 [r340931] Jonathan Rose + + * utils/utils.xml, funcs/func_jitterbuffer.c: Some additional + module documentation changes for 10 for the menuselect change. + (issue ASTERISK-18268) + +2011-10-14 16:39 +0000 [r340879] 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 + +2011-10-14 16:18 +0000 [r340868] Jonathan Rose + + * funcs/func_realtime.c, build_tools/cflags.xml, utils/utils.xml, + /, res/res_fax.c, apps/app_celgenuserevent.c, + codecs/codec_dahdi.c, apps/app_system.c, res/res_curl.c: Fixes + some support level info so that it can be read by menuselect. + (issue ASTERISK-18268) Review: + https://reviewboard.asterisk.org/r/1525/ ........ Merged + revisions 340863 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + +2011-10-13 22:54 +0000 [r340810] 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 + +2011-10-13 08:46 +0000 [r340770] Gregory Nietsky + + * channels/chan_sip.c: Only send MWI Notify on register if the + registration is successful. lastmsgssent was removed from + chan_sip and the old behavior of sending a mwi notify on register + [except when subscribemwi is set] was restored but this must only + happen when registration succeeds. leaking information for + unsuccessful registrations is not secure. + +2011-10-13 06:59 +0000 [r340718] Stefan Schmidt + + * channels/chan_sip.c: 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 06:56 +0000 [r340578-340716] 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 + + * /, 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 + + * /, channels/chan_sip.c: 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 20:33 +0000 [r340577] Stefan Schmidt + + * /, channels/chan_sip.c: 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 20:08 +0000 [r340471-340523] 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 + + * /, 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 + +2011-10-12 16:28 +0000 [r340419] 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 + +2011-10-11 21:05 +0000 [r340281-340366] 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 + + * 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 + + * 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 + +2011-10-11 18:41 +0000 [r340280] 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 + +2011-10-10 22:55 +0000 [r340219-340222] Terry Wilson + + * main/db.c: 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/astdb2bdb.c (added): Add a missing file for the astdb2bdb + conversion utility + + * utils/Makefile, utils/utils.xml, UPGRADE.txt: 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/ + +2011-10-10 20:30 +0000 [r340165] Matthew Jordan + + * /, channels/chan_sip.c: 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 14:15 +0000 [r340109] Matthew Nicholson + + * main/loader.c, main/xmldoc.c, 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: + 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/ ........ + +2011-10-09 22:18 +0000 [r339992-340031] Damien Wedhorn + + * channels/chan_skinny.c: 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: 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 01:18 +0000 [r339831-339942] igorg : + + * channels/chan_unistim.c, /: 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 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 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 19:36 +0000 [r339777] Richard Mudgett + + * /, apps/app_url.c: 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 + ........ + +2011-10-06 23:08 +0000 [r339722] Damien Wedhorn + + * channels/chan_skinny.c: Reject v17 skinny devices in Asterisk10 + Small fix for Asterisk10 to reject skinny devices with skinny + firmware version17 and above. Review: + https://reviewboard.asterisk.org/r/1497/ + +2011-10-06 22:58 +0000 [r339720] Richard Mudgett + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + autoconf/ast_ext_lib.m4: 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 20:47 +0000 [r339681] 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 (in trunk) and myself. + +2011-10-06 17:53 +0000 [r339626] Richard Mudgett + + * main/udptl.c, /, channels/chan_sip.c: 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 [r339586] Leif Madsen + + * /, build_tools/prep_tarball: 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 17:01 +0000 [r339508-339512] Richard Mudgett + + * apps/app_dial.c, /: 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 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 16:32 +0000 [r339507] Matthew Nicholson + + * /, res/res_fax.c: 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 06:28 +0000 [r339463] Gregory Nietsky + + * res/res_fax.c: 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 22:56 +0000 [r339407] Richard Mudgett + + * Makefile, /: 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 19:44 +0000 [r339298-339353] Jonathan Rose + + * /, main/say.c: 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 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 11:49 +0000 [r339245] Alexandr Anikin + + * addons/ooh323c/src/memheap.c, /: 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 ........ + +2011-10-03 20:13 +0000 [r339145-339148] Leif Madsen + + * /, channels/chan_sip.c: 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 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 18:52 +0000 [r339089] Alexandr Anikin + + * addons/ooh323c/src/memheap.c, /: 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 18:44 +0000 [r339088] Terry Wilson + + * /, channels/chan_sip.c, main/file.c: 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 15:54 +0000 [r339011-339045] Matthew Nicholson + + * res/res_fax.c: 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: Don't clear the AST_FAX_TECH_MULTI_DOC flag right + after we set it. + + * res/res_fax.c: properly remove the AST_FAX_TECH_GATEWAY flag + (instead of setting all of the other flags) + +2011-10-03 14:38 +0000 [r338904-338997] Gregory Nietsky + + * CHANGES: Documentation noting the extension of CHANNEL() for + chan_ooh323 + + * addons/chan_ooh323.c, funcs/func_channel.c: Remove the channel + function OOH323() and place its options into CHANNEL() channel + drivers should not have there own dialplan functions. + + * res/res_fax.c: 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: 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. + +2011-09-30 22:06 +0000 [r338801] Richard Mudgett + + * channels/chan_dahdi.c, /: 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 18:55 +0000 [r338719] Jonathan Rose + + * /, configs/queues.conf.sample: 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 16:35 +0000 [r338664] 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 + +2011-09-29 21:14 +0000 [r338556] Paul Belanger + + * tests/test_amihooks.c, tests/test_security_events.c, /, + tests/test_locale.c, tests/test_logger.c, + tests/test_dlinklists.c, tests/test_linkedlists.c: 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 20:54 +0000 [r338552] Jason Parker + + * /, tests/test_db.c, tests/test_netsock2.c: 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 18:32 +0000 [r338493] Leif Madsen + + * /, channels/chan_sip.c: Merged revisions 338492 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r338492 | lmadsen | 2011-09-29 13:31:33 -0500 (Thu, 29 Sep 2011) + | 6 lines Update documentation for SIP_HEADER. The SIP_HEADER + function only works on the the initial SIP INVITE. The + documentation was updated in trunk, but not in 1.8 or 10, so I'm + making them match. (Closes issue ASTERISK-18640) ........ + +2011-09-29 12:16 +0000 [r338417] Gregory Nietsky + + * /, channels/chan_sip.c, channels/sip/include/sip.h: 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 + ........ + +2011-09-28 22:36 +0000 [r338253-338323] Richard Mudgett + + * /, channels/sig_pri.c: 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 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 20:54 +0000 [r338228] 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 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 20:26 +0000 [r338225] Richard Mudgett + + * channels/chan_dahdi.c, /: 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 ........ + +2011-09-27 20:13 +0000 [r338085] Paul Belanger + + * /, apps/app_macro.c: 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 ........ + +2011-09-26 19:35 +0000 [r337974] Richard Mudgett + + * cdr/cdr_manager.c, cdr/cdr_custom.c, apps/app_voicemail.c, + 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: 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 ........ + +2011-09-27 Asterisk Development Team + + * Asterisk 10.0.0-beta2 Released. + + * Based on revision that passed automated testing + (http://bamboo.asterisk.org/browse/AST10-LUCID-178) + +2011-09-26 19:35 +0000 [r337974] Richard Mudgett + + * cdr/cdr_manager.c, cdr/cdr_custom.c, apps/app_voicemail.c, + 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: 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 19:18 +0000 [r337840-337902] Gregory Nietsky + + * /, contrib/init.d/rc.archlinux.asterisk: 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 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 00:45 +0000 [r337775] Russell Bryant + + * configs/res_pktccops.conf.sample, /: 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 21:37 +0000 [r337721] Richard Mudgett + + * /, channels/sig_pri.c: 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 18:43 +0000 [r337640] Paul Belanger + + * CREDITS, apps/app_meetme.c, CHANGES: Revert previous commit New + feature should be added into trunk, unfortunately it is too late + for the Asterisk 10 branch. + +2011-09-22 15:47 +0000 [r337595-337597] Jonathan Rose + + * channels/sip/security_events.c (added), + channels/sip/include/security_events.h (added): 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/ + + * configs/logger.conf.sample, channels/chan_sip.c, + include/asterisk/event_defs.h, main/security_events.c, + main/event.c, CHANGES, channels/sip/include/sip.h, + include/asterisk/security_events_defs.h: 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/ + +2011-09-22 11:44 +0000 [r337431-337542] Gregory Nietsky + + * res/res_srtp.c, /: 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 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) + ........ + + * apps/app_originate.c, CHANGES: Revert commit r337261 This commit + is for trunk not version 10 ----- 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. ----- + + * main/channel.c, /: 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 21:25 +0000 [r337342-337380] Tilghman Lesher + + * apps/app_voicemail.c, /: More silly spacing changes ..... Merged + revisions 337353 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * apps/app_voicemail.c, /: ........ Dumb little spacing fix. + ........ Merged revisions 337344 from + http://svn.asterisk.org/svn/asterisk/branches/1.8 + + * 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 + +2011-09-21 11:15 +0000 [r337261-337263] Gregory Nietsky + + * configs/sip.conf.sample: Whitespace fixup from SRTP patch + + * apps/app_originate.c, CHANGES: 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 09:32 +0000 [r337178-337219] Olle Johansson + + * configs/extensions.conf.sample, main/pbx.c, CHANGES: 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: 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 22:49 +0000 [r337120] Matthew Jordan + + * apps/app_voicemail.c, 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: 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 22:47 +0000 [r337119] Richard Mudgett + + * funcs/func_strings.c: 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 + +2011-09-20 22:19 +0000 [r337116] 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 [r337062] Kinsey Moore + + * tests/test_pbx.c, main/pbx.c, /: 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 19:12 +0000 [r336978-337008] Richard Mudgett + + * /, channels/sig_ss7.c: 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 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 16:51 +0000 [r336936] 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: + 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 01:03 +0000 [r336878] Russell Bryant + + * res/res_rtp_asterisk.c, /: 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 22:13 +0000 [r336792] Terry Wilson + + * /, channels/chan_sip.c: 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 21:41 +0000 [r336734-336789] Tilghman Lesher + + * funcs/func_strings.c: Ensure substring will not be found in the + previous match. + + * include/asterisk/optional_api.h, Makefile, /, configure, + include/asterisk/autoconfig.h.in, main/Makefile, + codecs/gsm/Makefile, configure.ac, Makefile.rules: 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 20:16 +0000 [r336717] 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 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 18:51 +0000 [r336659] Richard Mudgett + + * apps/app_dial.c, /, UPGRADE-1.8.txt: 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 ........ + +2011-09-19 15:42 +0000 [r336573] 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 15:32 +0000 [r336570] Richard Mudgett + + * /, channels/sig_pri.c: 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 13:48 +0000 [r336502-336504] Olle Johansson + + * Makefile: Revert accidental change + + * Makefile, /, channels/chan_sip.c: 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 13:31 +0000 [r336500] Gregory Nietsky + + * /, channels/chan_h323.c: 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 12:15 +0000 [r336381-336441] Olle Johansson + + * main/manager.c, /: 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 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 22:11 +0000 [r336313-336316] Terry Wilson + + * /, funcs/func_frame_trace.c: 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 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 21:09 +0000 [r336307] Jonathan Rose + + * main/channel.c, main/rtp_engine.c, /, channels/chan_sip.c, + include/asterisk/frame.h: 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 19:10 +0000 [r336235] Sean Bright + + * /, UPGRADE-1.8.txt: 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 10:12 +0000 [r336094-336167] Gregory Nietsky + + * channels/chan_misdn.c, /: 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 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 [r336091] David Vossel + + * main/format_cap.c: Removes some no-op code found in format_cap.c. + +2011-09-15 12:46 +0000 [r336042] Olle Johansson + + * CREDITS, apps/app_meetme.c, CHANGES: 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 08:29 +0000 [r335991] Gregory Nietsky + + * /, channels/chan_agent.c: 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 18:31 +0000 [r335852-335912] Richard Mudgett + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + 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 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 13:28 +0000 [r335791] Matthew Nicholson + + * main/manager.c, /: 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 22:10 +0000 [r335721] Richard Mudgett + + * /, apps/app_directed_pickup.c: 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 21:37 +0000 [r335717] 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 + +2011-09-13 18:55 +0000 [r335656] Tilghman Lesher + + * /, configure, configure.ac: 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 18:47 +0000 [r335653] Matthew Nicholson + + * main/pbx.c, main/manager.c, /: 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 07:24 +0000 [r335510] Russell Bryant + + * include/asterisk/event.h, /, res/ais/evt.c, main/event.c: 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 15:55 +0000 [r335434] Matthew Nicholson + + * main/channel.c, /: 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 14:22 +0000 [r335346] Kinsey Moore + + * apps/app_dial.c, /: 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 13:47 +0000 [r335323] Olle Johansson + + * /, channels/chan_sip.c: 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 [r335321] Kinsey Moore + + * /, channels/chan_iax2.c: 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 11:11 +0000 [r335260] Stefan Schmidt + + * /, channels/chan_sip.c: 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/ ........ + +2011-09-09 16:27 +0000 [r335078] Matthew Jordan + + * channels/chan_mgcp.c, 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: 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 07:23 +0000 [r335014] Gregory Nietsky + + * funcs/func_dialplan.c, apps/app_readexten.c, CHANGES: 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 22:28 +0000 [r334954] Richard Mudgett + + * /, main/logger.c: 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. ........ + +2011-09-07 19:37 +0000 [r334844] Paul Belanger + + * /, channels/chan_iax2.c: 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 19:33 +0000 [r334841] Richard Mudgett + + * /, main/features.c: 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 15:10 +0000 [r334682-334747] Stefan Schmidt + + * /, main/features.c: 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: another clean up + + * 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 08:14 +0000 [r334617-334621] Alec L Davis + + * /, CHANGES, apps/app_queue.c: 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/pbx.c, /: 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/ ........ + +2011-09-06 15:47 +0000 [r334514] Paul Belanger + + * configs/iax.conf.sample, CHANGES, channels/chan_iax2.c: 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 13:58 +0000 [r334455] Gregory Nietsky + + * apps/app_voicemail.c, /: 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 21:08 +0000 [r334297-334357] Richard Mudgett + + * /, res/res_musiconhold.c: 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 + 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 17:39 +0000 [r334235] Tilghman Lesher + + * main/pbx.c, /: 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. ........ + +2011-09-01 17:36 +0000 [r334233] Matthew Nicholson + + * CHANGES: fixed a typo + +2011-09-01 17:30 +0000 [r334230] Tilghman Lesher + + * res/res_config_odbc.c, /: 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 18:53 +0000 [r334064-334157] Matthew Nicholson + + * /, channels/chan_sip.c: 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 ........ + + * res/res_fax.c: only alter the gateway_timeout when attching the + gateway to a channel ASTERISK-18219 + +2011-08-31 16:00 +0000 [r334010-334013] Richard Mudgett + + * channels/chan_dahdi.c, /: 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 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 15:19 +0000 [r334007] Kinsey Moore + + * /, channels/chan_sip.c: 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 21:53 +0000 [r333961-333962] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c: security fix. really drop call if + signalling addr is not same as socket addr + + * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooCalls.c, + addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, /, + addons/ooh323c/src/ooCalls.h, addons/ooh323c/src/oochannels.c: + 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 ........ + +2011-08-30 14:01 +0000 [r333895] Matthew Nicholson + + * res/res_fax.c: 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 21:41 +0000 [r333837] Terry Wilson + + * /, channels/chan_sip.c: 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 21:12 +0000 [r333786] Richard Mudgett + + * /, include/asterisk/channel.h, addons/chan_mobile.c: 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 18:22 +0000 [r333716] Matthew Nicholson + + * res/res_fax_spandsp.c: 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 17:28 +0000 [r333681] Terry Wilson + + * main/channel.c, CHANGES: 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 17:12 +0000 [r333631] Matthew Jordan + + * apps/app_voicemail.c, /: 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 15:56 +0000 [r333570] Jonathan Rose + + * /, res/res_jabber.c: 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 09:55 +0000 [r333508] 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 16:28 +0000 [r333410] Jonathan Rose + + * /, res/res_jabber.c: 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 15:58 +0000 [r333370] Matthew Jordan + + * apps/app_voicemail.c, /: 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 19:01 +0000 [r333268] Jason Parker + + * Makefile, /: 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. ........ + +2011-08-25 19:00 +0000 [r333266] Jonathan Rose + + * /, res/res_jabber.c: 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 15:29 +0000 [r333203] Jason Parker + + * Makefile, build_tools/mkpkgconfig, /, configure, configure.ac, + makeopts.in, sounds/Makefile: 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/ ........ + +2011-08-24 16:51 +0000 [r333115] Matthew Nicholson + + * res/res_fax.c: Changed the "timeout" option to "gwtimeout". + ASTERISK-18219 + +2011-08-23 18:15 +0000 [r332878-333011] Richard Mudgett + + * /, apps/app_queue.c: 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 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 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 19:43 +0000 [r332877] Paul Belanger + + * /, channels/chan_gtalk.c: 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 19:41 +0000 [r332875] Richard Mudgett + + * /: Fix merge property. + +2011-08-22 18:40 +0000 [r332832] Matthew Jordan + + * apps/app_voicemail.c, include/asterisk/test.h, main/manager.c, /, + main/file.c, main/test.c, main/app.c, + configs/manager.conf.sample, include/asterisk/manager.h: 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 18:32 +0000 [r332761-332830] Richard Mudgett + + * res/res_config_pgsql.c, res/res_config_odbc.c, /: 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 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 16:29 +0000 [r332756] Matthew Nicholson + + * res/res_fax.c, include/asterisk/res_fax.h: add a way to disable + and/or modify the gateway timeout ASTERISK-18219 + +2011-08-21 14:33 +0000 [r332700] Paul Belanger + + * /, channels/chan_gtalk.c: 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 19:59 +0000 [r332654] Kinsey Moore + + * apps/app_confbridge.c: 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/ + +2011-08-18 21:34 +0000 [r332560] Terry Wilson + + * main/netsock2.c, /: 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 19:29 +0000 [r332504] Kinsey Moore + + * channels/chan_dahdi.c, /: 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 14:48 +0000 [r332369-332447] Tilghman Lesher + + * build_tools/cflags.xml, build_tools/cflags-devmode.xml, /: 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 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 18:09 +0000 [r332321] Terry Wilson + + * /, res/res_timing_timerfd.c: 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 16:01 +0000 [r332265] 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 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 20:11 +0000 [r332177] Paul Belanger + + * tests/test_amihooks.c, 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_pbx.c, + tests/test_strings.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: 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 17:45 +0000 [r332119] Jonathan Rose + + * /, channels/chan_sip.c: 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 17:17 +0000 [r332101] Richard Mudgett + + * /, main/features.c, CHANGES, configs/features.conf.sample, + main/asterisk.c: 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 15:20 +0000 [r332022-332042] Matthew Nicholson + + * channels/sip/include/sip.h: fix a code comment AST-580 + + * UPGRADE.txt, CHANGES: Moved notes about 'storesipcause' to + UPGRADE.txt from CHANGES AST-580 + + * /, channels/chan_sip.c, channels/sip/include/sip.h: 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 ........ + + * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: 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 17:35 +0000 [r331956] Richard Mudgett + + * channels/chan_dahdi.c, /: 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 15:22 +0000 [r331894] Paul Belanger + + * main/rtp_engine.c, /: 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 15:14 +0000 [r331868] David Vossel + + * /, channels/chan_sip.c: 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 ........ + +2011-08-12 19:03 +0000 [r331775] Matthew Nicholson + + * /, apps/app_queue.c: 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 18:59 +0000 [r331715-331772] Richard Mudgett + + * channels/chan_dahdi.c, /: 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 ........ + + * channels/chan_dahdi.c, /: 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 16:31 +0000 [r331659] Terry Wilson + + * /, tests/test_netsock2.c: 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 16:21 +0000 [r331654] Kinsey Moore + + * /, main/logger.c: 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 ........ + +2011-08-12 16:18 +0000 [r331644] Jonathan Rose + + * apps/app_dial.c, /, apps/app_meetme.c: 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 ........ + +2011-08-11 21:54 +0000 [r331579] Jason Parker + + * apps/app_dial.c, /, apps/app_meetme.c: 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 21:42 +0000 [r331576] Richard Mudgett + + * /, funcs/func_shell.c: 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 22:23 +0000 [r331518] Kinsey Moore + + * /, channels/chan_sip.c: 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 20:41 +0000 [r331418-331462] Richard Mudgett + + * /, main/logger.c: 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: Make sure feature_request_and_dial() initializes + outstate if passed in. + + * main/features.c, CHANGES: 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 13:48 +0000 [r331316] Kinsey Moore + + * main/manager.c, /: 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 23:12 +0000 [r331265] Richard Mudgett + + * apps/app_parkandannounce.c, main/pbx.c, /, channels/chan_sip.c, + main/features.c, channels/chan_iax2.c: 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 16:36 +0000 [r331147-331200] Alexandr Anikin + + * addons/ooh323c/src/ooq931.c, addons/ooh323c/src/ooGkClient.c: + 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 + + * addons/ooh323c/src/ooGkClient.c, addons/chan_ooh323.c, /, + addons/ooh323c/src/ooLogChan.c: 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 ........ + +2011-08-09 15:59 +0000 [r331138-331143] Jason Parker + + * /, doc/asterisk.8: 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.sgml, /, doc/asterisk.8, + configs/asterisk.conf.sample, configs/voicemail.conf.sample: + 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.sgml, /, doc/asterisk.8, + configs/asterisk.conf.sample, configs/voicemail.conf.sample: + Revert merge of r306999, due to merge conflict. + +2011-08-08 22:59 +0000 [r331041-331097] Terry Wilson + + * UPGRADE.txt, CHANGES, include/asterisk/manager.h: 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: 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 issue + ASTERISK-17455) + +2011-08-08 20:53 +0000 [r331039] Kinsey Moore + + * /, res/res_musiconhold.c: 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 ........ + +2011-08-05 15:53 +0000 [r330940] David Vossel + + * codecs/codec_resample.c: The slin resampler is no longer + dependent on an external library, but the dependency was not + removed correctly. + +2011-08-05 07:38 +0000 [r330899] Alexandr Anikin + + * addons/ooh323c/src/ooGkClient.c, /, + addons/ooh323c/src/ooCmdChannel.c: 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 20:51 +0000 [r330844] Terry Wilson + + * /, configure, configure.ac: 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 15:15 +0000 [r330706-330763] Kinsey Moore + + * /, main/Makefile: 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 + 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 20:52 +0000 [r330649] Kevin P. Fleming + + * /, res/res_jabber.c: 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 16:17 +0000 [r330576-330586] David Vossel + + * /, channels/chan_iax2.c: 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 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 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 15:23 +0000 [r330434] Kinsey Moore + + * /, main/say.c: 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 ........ + +2011-07-30 23:57 +0000 [r330369] Richard Mudgett + + * main/channel.c, /: 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 15:34 +0000 [r330312] Gregory Nietsky + + * main/channel.c, /: 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 17:19 +0000 [r330204-330217] Sean Bright + + * /, formats/format_wav.c: 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 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 05:25 +0000 [r330162] Paul Belanger + + * apps/app_confbridge.c: Fix typo pointed out on #asterisk Thanks + notten + +2011-07-28 21:44 +0000 [r330108] Terry Wilson + + * main/term.c, /: 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 17:10 +0000 [r330051] Richard Mudgett + + * /, channels/sig_pri.c: 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 15:45 +0000 [r329995] Jason Parker + + * /, channels/chan_sip.c: 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 15:28 +0000 [r329992] Matthew Nicholson + + * /, res/res_fax.c: 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 13:03 +0000 [r329771-329952] Sean Bright + + * configs/confbridge.conf.sample: 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: Correct the spelling of 'conference.' + + * /, channels/chan_sip.c: 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. ........ + + * /, Makefile.moddir_rules: 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 issue ASTERISK-18141) + Reported by: Richard Miller Patches: sort-order.diff uploaded by + seanbright (License #5060) Tested by: leifmadsen ........ + +2011-07-27 18:11 +0000 [r329710] Jonathan Rose + + * /, configs/indications.conf.sample: 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 15:25 +0000 [r329670] Sean Bright + + * main/loader.c: Sort the module list so that 'module show' is + alphabetical. + +2011-07-27 04:25 +0000 [r329614] Tilghman Lesher + + * /, cdr/cdr_odbc.c: 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 14:19 +0000 [r329528-329538] Jonathan Rose + + * apps/app_voicemail.c, /: 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. ........ + + * apps/app_voicemail.c, include/asterisk/app.h, /, + configs/voicemail.conf.sample, main/app.c: 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 19:55 +0000 [r329472] Paul Belanger + + * /, main/enum.c: 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 14:06 +0000 [r329430-329431] Gregory Nietsky + + * main/dsp.c, include/asterisk/dsp.h: 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/1272/ + +2011-07-22 21:14 +0000 [r329331-329334] Richard Mudgett + + * main/pbx.c, /: Make use less redundant loop construct for + iterating over hints. + + * main/pbx.c, /: 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/ + ........ + +2011-07-22 Leif Madsen + + * Asterisk 10.0.0-beta1 Released. + +2011-07-21 20:22 +0000 [r329257] Russell Bryant + + * channels/chan_dahdi.c, main/features.c, + include/asterisk/netsock2.h, CHANGES, channels/sig_pri.c, + include/asterisk/rtp_engine.h: s/1.10/10.0/ + +2011-07-21 18:05 +0000 [r329200-329204] Richard Mudgett + + * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: 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 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 ........ + +2011-07-21 17:27 +0000 [r329188] Jason Parker + + * UPGRADE.txt: Fix version number in UPGRADE.txt. + +2011-07-21 16:52 +0000 [r329145] Richard Mudgett + + * /, main/features.c: 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 16:04 +0000 [r329104] Russell Bryant + + * / (added): Change Asterisk 2.0 to 2.0 in binary + +2011-07-20 21:31 +0000 [r329056] Paul Belanger + + * /, UPGRADE-1.8.txt: Merged revisions 329055 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/2.0 + ................ r329055 | pabelanger | 2011-07-20 17:27:50 -0400 + (Wed, 20 Jul 2011) | 9 lines Merged revisions 329027 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r329027 | pabelanger | 2011-07-20 17:20:36 -0400 (Wed, + 20 Jul 2011) | 2 lines Asterisk now requires libpri 1.4.11+ for + PRI support. ........ ................ + +2011-07-20 20:19 +0000 [r328996] Terry Wilson + + * /, tests/test_netsock2.c: Merged revisions 328992 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/2.0 + ................ r328992 | twilson | 2011-07-20 15:18:25 -0500 + (Wed, 20 Jul 2011) | 12 lines Merged revisions 328987 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328987 | twilson | 2011-07-20 15:16:58 -0500 (Wed, 20 Jul 2011) + | 5 lines We can't guarantee an eth0 is present FreeBSD test + fails on this case presumably because there is no eth0 on the + test machine. Better to just remove this test for now. ........ + ................ + +2011-07-20 19:03 +0000 [r328937] Kinsey Moore + + * /, channels/chan_sip.c: Merged revisions 328936 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/2.0 + ................ r328936 | kmoore | 2011-07-20 14:01:37 -0500 + (Wed, 20 Jul 2011) | 15 lines Merged revisions 328935 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328935 | kmoore | 2011-07-20 14:00:23 -0500 (Wed, 20 Jul 2011) | + 8 lines Inband DTMF regression The functionality of inband DTMF + in chan_sip relied upon ast_rtp_instance_dtmf_mode_get/set not + working properly to avoid calling ast_rtp_instance_dtmf_begin/end + on RTP streams with inband DTMF. According to documentation, + ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF, + never inband. This fixes the regression introduced in revision + 328823. ........ ................ + +2011-07-19 21:32 +0000 [r328880-328881] Kevin P. Fleming + + * Makefile, /, Makefile.moddir_rules, sounds/Makefile: Merged + revisions 328879 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/2.0 + ................ r328879 | kpfleming | 2011-07-19 16:31:16 -0500 + (Tue, 19 Jul 2011) | 23 lines Merged revisions 328878 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328878 | kpfleming | 2011-07-19 16:29:07 -0500 (Tue, 19 Jul + 2011) | 17 lines Revert partial attempt at handling pathnames + with spaces. Revision 299794 attempted to improve the build + system to be able to handle pathnames (primarily DESTDIR) with + spaces in them, since this is common on some platforms (including + Mac OSX). Unfortunately, the changes were incomplete and did not + actually provide the desired behavior, and as a side effect the + functionality that ensured that stale headers in the Asterisk + 'include' directory were removed got broken. In addition, the + check for stale (and possibly incompatible) modules in the + Asterisk 'modules' directory also got broken, and would never + report any stale modules. Users upgrading to this version or + later versions would then see unexpected module load errors. + Since there are few users who actually want to install Asterisk + into paths that contain spaces, and a proper fix for the build + system would take many hours, the best solution for now is to + just revert the partial solution. ........ ................ + + * /: Edit the merge properties to match their names. + +2011-07-19 21:21 +0000 [r328877] Russell Bryant + + * /: Fix properties after twilson's change so merges still work + +2011-07-19 18:07 +0000 [r328772-328825] Kinsey Moore + + * res/res_rtp_asterisk.c, main/rtp_engine.c, /, + channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged + revisions 328824 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328824 | kmoore | 2011-07-19 13:05:21 -0500 + (Tue, 19 Jul 2011) | 18 lines Merged revisions 328823 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | + 11 lines RTP bridge away with inband DTMF and feature detection + When deciding whether Asterisk was allowed to bridge the call + away from the core, chan_sip did not take into account the usage + of features on dialed channels that require monitoring of DTMF on + channels utilizing inband DTMF. This would cause Asterisk to + allow the call to be locally or remotely bridged, preventing + access to the data required to detect activations of such + features. (closes 17237) Review: + https://reviewboard.asterisk.org/r/1302/ ........ + ................ + + * /, apps/app_meetme.c: Merged revisions 328771 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328771 | kmoore | 2011-07-19 10:46:54 -0500 + (Tue, 19 Jul 2011) | 18 lines Merged revisions 328770 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328770 | kmoore | 2011-07-19 10:43:32 -0500 (Tue, 19 Jul 2011) | + 11 lines MeetMe requests a PIN twice in some circumstances If a + call to MeetMe includes both the dynamic(D) and always request + PIN(P) options, MeetMe will ask for the PIN two times: once for + creating the conference and once for entering the conference. + This behavior was introduced in rev 311616 when adding the + CONFFLAG_ALWAYSPROMPT option to the logic branch controlling PIN + entry for joining a conference. (closes AST-601) Review: + https://reviewboard.asterisk.org/r/1305/ ........ + ................ + +2011-07-19 02:00 +0000 [r328718] Terry Wilson + + * /, include/asterisk/linkedlists.h, tests/test_linkedlists.c + (added): Merged revisions 328717 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328717 | twilson | 2011-07-18 20:55:32 -0500 + (Mon, 18 Jul 2011) | 14 lines Merged revisions 328716 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328716 | twilson | 2011-07-18 20:35:53 -0500 (Mon, 18 Jul 2011) + | 7 lines Make AST_LIST_REMOVE safer AST_LIST_REMOVE shouldn't + modify the element passed in if it isn't found. This commit also + adds linked list unit tests. Review: + https://reviewboard.asterisk.org/r/1321/ ........ + ................ + +2011-07-18 20:51 +0000 [r328610-328665] Mark Murawki + + * apps/app_dial.c, /: Merged revisions 328664 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328664 | markm | 2011-07-18 16:50:13 -0400 + (Mon, 18 Jul 2011) | 15 lines Merged revisions 328663 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328663 | markm | 2011-07-18 16:47:04 -0400 (Mon, 18 Jul 2011) | + 9 lines app_dial may double free a channel datastore When + starting a call with originate, and having the callee channel run + Bridge() on pickup, we will double free the dialed_interface_info + datastore, causing a crash. Make sure to check if the datastore + still exists before trying to free it. (closes issue + ASTERISK-17917) Reported by: Mark Murawski Tested by: Mark + Murawski ........ ................ + + * /, channels/chan_sip.c: Merged revisions 328611 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328611 | markm | 2011-07-18 08:56:49 -0400 + (Mon, 18 Jul 2011) | 15 lines Merged revisions 328608 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328608 | markm | 2011-07-18 08:35:57 -0400 (Mon, 18 Jul 2011) | + 9 lines If the sip private structure is null, sip_setoption() + will defref the null pointer and crash. Ideally, sip_setoption + shouldn't be called if there is a lack of a sip private + structure. But this will fix a crash. (closes issue + ASTERISK-17909) Reported by: Mark Murawski Tested by: Mark + Murawski ........ ................ + + * /, main/asterisk.c: Merged revisions 328609 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328609 | markm | 2011-07-18 08:37:53 -0400 + (Mon, 18 Jul 2011) | 15 lines Merged revisions 328593 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) | + 8 lines Fixed invalid read and null pointer deref on asterisk + shutdown. In some cases when starting asterisk with -c and + hitting control-c to shutdown, there will be an invalid read and + null pointer deref causing a crash. (closes issue ASTERISK-17927) + Reported by: Mark Murawski Tested by: Mark Murawski, Kinsey + Moore, Tilghman Lesher ........ ................ + +2011-07-18 07:12 +0000 [r328542] Tilghman Lesher + + * /, funcs/func_odbc.c: Merged revisions 328541 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328541 | tilghman | 2011-07-18 02:11:26 -0500 + (Mon, 18 Jul 2011) | 9 lines Merged revisions 328540 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r328540 | tilghman | 2011-07-18 02:10:15 -0500 (Mon, 18 + Jul 2011) | 2 lines Typo ........ ................ + +2011-07-15 21:41 +0000 [r328502] Alexandr Anikin + + * addons/ooh323c/src/ooGkClient.c, /: Merged revisions + 328428-328429 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328428 | may | 2011-07-15 23:31:09 +0400 (Fri, + 15 Jul 2011) | 13 lines Merged revisions 328427 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328427 | may | 2011-07-15 23:22:24 +0400 (Fri, 15 Jul 2011) | 7 + lines small gk processing fixes: - decrease for 1 second + registration ttl for very low expirations (some providers expire + few earlier than TTL) - delete rrq and registration expire timers + on URQ received as we make new registration. ........ + ................ r328429 | may | 2011-07-15 23:35:50 +0400 (Fri, + 15 Jul 2011) | 2 lines delete unproperly changed svn props + ................ + +2011-07-15 21:19 +0000 [r328449-328459] Leif Madsen + + * /, apps/app_macro.c: Merged revisions 328451 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ + r328451 | lmadsen | 2011-07-15 16:17:25 -0500 (Fri, 15 Jul 2011) + | 1 line Build app_macro by default because things depend on it. + ........ + + * /, UPGRADE-1.10.txt, UPGRADE.txt, CHANGES: Merged revisions + 328448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ + r328448 | lmadsen | 2011-07-15 16:57:15 -0400 (Fri, 15 Jul 2011) + | 2 lines Update UPGRADE.txt and CHANGES files. Update + documentation files stating that deprecated modules are no longer + built by default. ........ + +2011-07-15 08:19 +0000 [r328381] Damien Wedhorn + + * channels/chan_skinny.c: Add SLA to skinny. Adds sublines to + skinny lines. Each subline can be attached to an SLA + station/trunk combo. Includes the following functionality: Callid + is persistent for both in/out calls on all skinny devices. Can + join, hold, resume. All sublines appear under a single line + button. See: + https://wiki.asterisk.org/wiki/display/~wedhorn/Skinny+SLA for + doc. (closes issue ASTERISK-17947) Review: + https://reviewboard.asterisk.org/r/1239/ + +2011-07-15 00:23 +0000 [r328318-328344] Richard Mudgett + + * main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c, + include/asterisk/extconf.h, include/asterisk/pbx.h, + apps/app_queue.c: Merged revisions 328329 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ + r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) + | 2 lines Make hint watcher callback take const strings for + context and exten parameters. ........ + + * /, channels/chan_sip.c: Merged revisions 328317 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328317 | rmudgett | 2011-07-14 18:28:49 -0500 + (Thu, 14 Jul 2011) | 13 lines Merged revisions 328302 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328302 | rmudgett | 2011-07-14 18:12:06 -0500 (Thu, 14 Jul 2011) + | 6 lines Missing SIP pvt and channel unlock in + sip_set_rtp_peer(). Regression introduced by -r326144. Add + missing SIP pvt and channel unlock in sip_set_rtp_peer(). + ........ ................ + +2011-07-14 20:28 +0000 [r328259] Leif Madsen + + * funcs/func_speex.c, apps/app_playtones.c, + bridges/bridge_softmix.c, apps/app_alarmreceiver.c, + res/res_calendar_caldav.c, apps/app_ices.c, apps/app_exec.c, + channels/chan_iax2.c, res/res_pktccops.c, channels/chan_skinny.c, + pbx/pbx_ael.c, formats/format_h263.c, cdr/cdr_odbc.c, + cdr/cdr_manager.c, utils/refcounter.c, funcs/func_timeout.c, + formats/format_wav.c, apps/app_softhangup.c, codecs/codec_g726.c, + bridges/bridge_simple.c, funcs/func_cut.c, apps/app_talkdetect.c, + apps/app_db.c, funcs/func_callcompletion.c, funcs/func_channel.c, + funcs/func_iconv.c, pbx/pbx_config.c, res/res_odbc.c, + apps/app_voicemail.c, formats/format_sln.c, + apps/app_authenticate.c, apps/app_readexten.c, + res/res_phoneprov.c, apps/app_userevent.c, codecs/codec_gsm.c, + tests/test_func_file.c, apps/app_setcallerid.c, + res/res_config_odbc.c, funcs/func_audiohookinherit.c, + apps/app_osplookup.c, funcs/func_odbc.c, cel/cel_custom.c, + tests/test_utils.c, apps/app_mp3.c, res/res_timing_timerfd.c, + codecs/codec_resample.c, formats/format_h264.c, + apps/app_directory.c, formats/format_siren14.c, + tests/test_amihooks.c, res/res_config_pgsql.c, + funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c, + res/res_crypto.c, apps/app_originate.c, channels/chan_jingle.c, + apps/app_forkcdr.c, funcs/func_blacklist.c, apps/app_sms.c, + formats/format_g723.c, utils/extconf.c, tests/test_poll.c, + apps/app_stack.c, apps/app_verbose.c, utils/check_expr.c, + funcs/func_module.c, codecs/codec_adpcm.c, tests/test_event.c, + cdr/cdr_adaptive_odbc.c, apps/app_image.c, + formats/format_wav_gsm.c, utils/stereorize.c, pbx/pbx_loopback.c, + tests/test_time.c, funcs/func_shell.c, apps/app_skel.c, + channels/chan_alsa.c, apps/app_externalivr.c, + apps/app_milliwatt.c, formats/format_gsm.c, res/res_speech.c, + apps/app_dial.c, apps/app_page.c, apps/app_fax.c, utils/astman.c, + apps/app_disa.c, res/res_monitor.c, apps/app_waitforring.c, + addons/cdr_mysql.c, res/res_fax_spandsp.c, + res/res_timing_kqueue.c, apps/app_chanspy.c, apps/app_cdr.c, + channels/chan_unistim.c, funcs/func_base64.c, + channels/chan_multicast_rtp.c, funcs/func_md5.c, + apps/app_meetme.c, tests/test_gosub.c, funcs/func_sysinfo.c, + funcs/func_vmcount.c, res/res_musiconhold.c, cdr/cdr_radius.c, + apps/app_followme.c, res/res_config_sqlite.c, + apps/app_controlplayback.c, cdr/cdr_csv.c, formats/format_ilbc.c, + channels/chan_phone.c, funcs/func_enum.c, main/manager.c, + funcs/func_groupcount.c, tests/test_stringfields.c, + tests/test_locale.c, tests/test_devicestate.c, + funcs/func_frame_trace.c, funcs/func_aes.c, cdr/cdr_sqlite.c, + tests/test_astobj2.c, apps/app_ivrdemo.c, res/res_clioriginate.c, + apps/app_jack.c, apps/app_nbscat.c, res/res_calendar_icalendar.c, + codecs/codec_a_mu.c, tests/test_ast_format_str_reduce.c, + tests/test_dlinklists.c, res/res_convert.c, apps/app_waituntil.c, + pbx/pbx_lua.c, utils/astcanary.c, apps/app_queue.c, + channels/chan_oss.c, cdr/cdr_tds.c, channels/chan_usbradio.c, + apps/app_flash.c, apps/app_senddtmf.c, funcs/func_callerid.c, + addons/app_saycountpl.c, cel/cel_pgsql.c, apps/app_dahdibarge.c, + channels/chan_local.c, funcs/func_dialgroup.c, + tests/test_logger.c, apps/app_record.c, funcs/func_env.c, + funcs/func_strings.c, res/res_timing_dahdi.c, + apps/app_chanisavail.c, bridges/bridge_multiplexed.c, + res/res_rtp_multicast.c, cel/cel_odbc.c, channels/chan_dahdi.c, + pbx/pbx_spool.c, funcs/func_pitchshift.c, formats/format_pcm.c, + apps/app_dumpchan.c, main/http.c, res/res_clialiases.c, + res/res_calendar_exchange.c, res/res_ais.c, funcs/func_sprintf.c, + codecs/codec_g722.c, tests/test_expr.c, cel/cel_tds.c, + tests/test_app.c, utils/smsq.c, apps/app_morsecode.c, + formats/format_ogg_vorbis.c, tests/test_sched.c, + res/res_calendar_ews.c, apps/app_speech_utils.c, + tests/test_acl.c, apps/app_sendtext.c, funcs/func_cdr.c, + utils/hashtest2.c, utils/ael_main.c, apps/app_mixmonitor.c, + formats/format_g726.c, utils/streamplayer.c, res/res_calendar.c, + cel/cel_radius.c, channels/chan_vpb.cc, tests/test_heap.c, + addons/format_mp3.c, res/res_snmp.c, apps/app_dictate.c, + channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c, + res/res_jabber.c, funcs/func_uri.c, cel/cel_manager.c, + apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c, + apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, + codecs/codec_lpc10.c, apps/app_read.c, cdr/cdr_syslog.c, + codecs/codec_alaw.c, res/res_adsi.c, agi/eagi-test.c, + utils/conf2ael.c, tests/test_pbx.c, apps/app_channelredirect.c, + formats/format_vox.c, res/res_stun_monitor.c, tests/test_aoc.c, + pbx/pbx_dundi.c, funcs/func_devstate.c, + addons/res_config_mysql.c, funcs/func_rand.c, + apps/app_readfile.c, addons/chan_ooh323.c, + cdr/cdr_sqlite3_custom.c, /, apps/app_sayunixtime.c, + apps/app_test.c, res/res_http_post.c, res/res_smdi.c, + main/features.c, funcs/func_srv.c, apps/app_amd.c, + pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c, + formats/format_g719.c, channels/chan_bridge.c, + apps/app_privacy.c, apps/app_echo.c, codecs/codec_speex.c, + apps/app_saycounted.c, apps/app_dahdiras.c, + channels/chan_agent.c, funcs/func_math.c, res/res_ael_share.c, + apps/app_transfer.c, res/res_mutestream.c, apps/app_playback.c, + res/res_config_curl.c, channels/chan_misdn.c, funcs/func_curl.c, + tests/test_skel.c, apps/app_macro.c, apps/app_zapateller.c, + codecs/codec_ilbc.c, addons/app_mysql.c, + tests/test_substitution.c, utils/muted.c, utils/hashtest.c, + funcs/func_sha1.c, formats/format_siren7.c, + tests/test_security_events.c, funcs/func_config.c, + bridges/bridge_builtin_features.c, funcs/func_volume.c, + res/res_agi.c, apps/app_confbridge.c, addons/chan_mobile.c, + apps/app_parkandannounce.c, res/res_security_log.c, + cdr/cdr_custom.c, apps/app_while.c, res/res_rtp_asterisk.c, + funcs/func_dialplan.c, funcs/func_db.c, apps/app_festival.c, + res/res_limit.c, res/res_fax.c, apps/app_waitforsilence.c, + channels/chan_console.c, apps/app_getcpeid.c, + funcs/func_global.c, res/res_srtp.c, funcs/func_extstate.c, + tests/test_strings.c, res/res_timing_pthread.c, + apps/app_directed_pickup.c, channels/chan_h323.c, + cel/cel_sqlite3_custom.c, codecs/codec_ulaw.c, + channels/chan_nbs.c, formats/format_g729.c: Merged revisions + 328247 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 + (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) + | 6 lines Introduce tags in MODULEINFO. This + change introduces MODULEINFO into many modules in Asterisk in + order to show the community support level for those modules. This + is used by changes committed to menuselect by Russell Bryant + recently (r917 in menuselect). More information about the support + level types and what they mean is available on the wiki at + https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States + ........ ................ + +2011-07-14 19:56 +0000 [r328208] Jonathan Rose + + * /, res/res_monitor.c: Merged revisions 328207 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 + ................ r328207 | jrose | 2011-07-14 14:45:18 -0500 + (Thu, 14 Jul 2011) | 13 lines Merged revisions 328205 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328205 | jrose | 2011-07-14 14:21:02 -0500 (Thu, 14 Jul 2011) | + 6 lines Monitor application arguments requirements fixed. Monitor + was requiring options in spite of no individual option on Monitor + being required. Review: https://reviewboard.asterisk.org/r/1320/ + ........ ................ + +2011-07-14 17:47 +0000 [r328163] Matthew Nicholson + + * /, main/dsp.c: Merged revisions 328162 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ + r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul + 2011) | 3 lines tune the v21 preamble detector to properly detect + the desired sequence of hits and misses ........ + +2011-07-13 22:10 +0000 [r328121] David Vossel + + * /, apps/app_mixmonitor.c: Merged revisions 328120 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.10 + ........ r328120 | dvossel | 2011-07-13 17:09:34 -0500 (Wed, 13 + Jul 2011) | 15 lines Preserve sample rate quality of wideband + mixmonitor recordings. MixMonitor has the ability to record in + any file format Asterisk supports, but the quality of wideband + audio is not preserved. This is because regardless of the sample + rate the call is being recorded in, the audio is always + downsampled to 8khz and then upsampled to whatever wideband + format it is being written as. This patch resolves this by + requesting the audio from the audiohook in the signed linear + format closest to the sample rate of the format we are writing. + This fix is only possible for Asterisk 1.10 because audio hooks + in 1.8 are not capable of wideband audio. Review: + https://reviewboard.asterisk.org/r/1314/ ........ + +2011-07-13 21:06 +0000 [r328079] Leif Madsen + + * BUGS, UPGRADE-1.10.txt (added), UPGRADE.txt: Add UPGRADE-1.10.txt + file from UPGRADE.txt. + +2011-07-13 20:40 +0000 [r328075-328076] Russell Bryant + + * /: set 1.10 merge properties + + * /: remove 1.8 merge properties + +2011-07-13 18:47 +0000 [r328016] Richard Mudgett + + * /, configs/features.conf.sample: Merged revisions 328014 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r328014 | rmudgett | 2011-07-13 13:46:38 -0500 (Wed, 13 Jul 2011) + | 1 line Add ATXFER_NULL_TECH note in features.conf.sample. + ........ + +2011-07-12 23:02 +0000 [r327953] Kevin P. Fleming + + * main/manager.c, /: Merged revisions 327950 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul + 2011) | 14 lines Correct double-free situation in manager output + processing. The process_output() function calls ast_str_append() + and xml_translate() on its 'out' parameter, which is a pointer to + an ast_str buffer. If either of these functions need to + reallocate the ast_str so it will have more space, they will free + the existing buffer and allocate a new one, returning the address + of the new one. However, because process_output only receives a + pointer to the ast_str, not a pointer to its caller's variable + holding the pointer, if the original ast_str is freed, the caller + will not know, and will continue to use it (and later attempt to + free it). (reported by jkroon on #asterisk-dev) ........ + +2011-07-12 20:08 +0000 [r327891] Matthew Nicholson + + * /, apps/app_directory.c: Merged revisions 327890 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r327890 | mnicholson | 2011-07-12 15:07:20 -0500 (Tue, + 12 Jul 2011) | 2 lines search in the current context for 'a' and + 'o' instead of 'default' ........ + +2011-07-12 19:39 +0000 [r327889] Jason Parker + + * Makefile, /: Merged revisions 327888 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327888 | qwell | 2011-07-12 14:38:44 -0500 (Tue, 12 Jul 2011) | + 1 line Fix uninstall target, so that modules dir gets cleared + again. ........ + +2011-07-12 19:18 +0000 [r327856] Matthew Jordan + + * /, apps/app_voicemail.c: Merged revisions 327852 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r327852 | mjordan | 2011-07-12 14:10:34 -0500 (Tue, 12 + Jul 2011) | 12 lines Added additional checks for mailbox / + password beginning with '*' character A bug existed such that if + a user entered a password with '*', and the extension 'a' did not + exist, an invalid mailbox would be created and the user + authenticated. The code was changed to prevent this from + occurring, and to prevent users from having mailboxes or + passwords defined that begin with the '*' character. (closes + issue ASTERISK-17443) Reported by: Kevin Scott Adams Tested by: + Matt Jordan Review: https://reviewboard.asterisk.org/r/1316/ + ........ + +2011-07-12 15:38 +0000 [r327794] Tilghman Lesher + + * tests/test_substitution.c, /: Merged revisions 327793 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327793 | tilghman | 2011-07-12 10:35:46 -0500 (Tue, 12 Jul 2011) + | 14 lines Use 'printf' (POSIX issue 4) instead of 'echo -n', for + portability. The problem with using 'echo -n' is that it is not + portable. While BSD systems required that the '-n' option be + removed and interpreted, System V required that all strings + should be echoed with no interpretation of options. This + fundamental difference of behavior means that it is never + possible to use the '-n' flag to echo in tests which are meant to + be portable. In this case, on Mac OS X 10.6, the /bin/sh shell + builtin 'echo' uses the System V semantics of the command, and + thus the SHELL test failed on that platform. + http://pubs.opengroup.org/onlinepubs/009695399/utilities/echo.html#tag_04_41_16 + ........ + +2011-07-12 15:23 +0000 [r327769] Matthew Nicholson + + * res/res_fax.c, include/asterisk/dsp.h, main/dsp.c: do v21 + detection instead of CED detection for the fax gateway + +2011-07-12 14:55 +0000 [r327749] David Vossel + + * main/bridging.c: Send video update frame to new video source in + follow_talker correctly. + +2011-07-12 14:40 +0000 [r327748] Kinsey Moore + + * apps/app_confbridge.c: Segfault on shutdown when confbridge is + active When undergoing a shutdown and channels are kicked out of + a bridge, a segfault occurs because ConfBridge tries to play + sounds on the bridge after the underlying channels have been + blown away due to the shutdown. (closes ASTERISK-18040) Review: + https://reviewboard.asterisk.org/r/1283/ + +2011-07-11 20:06 +0000 [r327684] Matthew Nicholson + + * tests/test_substitution.c: use printf instead of echo -n in + substitution test + +2011-07-11 19:49 +0000 [r327683] Terry Wilson + + * /, include/asterisk/jingle.h, channels/chan_gtalk.c: Merged + revisions 327682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327682 | twilson | 2011-07-11 12:41:59 -0700 (Mon, 11 Jul 2011) + | 9 lines Update chan_gtalk to work with changed GMail-based + calls The messages sent by the GMail client have changed, but + include the old-style messages as well. This patch checks for + this case and uses the old-style offer. (closes issue + ASTERISK-18084) Review: https://reviewboard.asterisk.org/r/1312/ + ........ + +2011-07-11 18:44 +0000 [r327640] David Vossel + + * include/asterisk/bridging.h, bridges/bridge_softmix.c, + main/bridging.c: Updates follow_talker video_mode in confbridge + application. follow_talker mode originally echoed the same video + stream to all participants. As the primary talker switched + around, the video stream would result in the talker seeing + themselves. Now the primary talker sees the last person who was + talking rather than themselves. + +2011-07-11 17:23 +0000 [r327469-327598] Matthew Nicholson + + * res/res_fax.c: renamed fax_gateway_send_ced() to + fax_gateway_request_t38() + + * res/res_fax.c: actually do something with the ced timeout, also + added more debug output + + * res/res_fax.c: write silence on the channel during t.38 + negotiation + + * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327512 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327512 | mnicholson | 2011-07-11 08:53:59 -0500 (Mon, 11 Jul + 2011) | 2 lines reset our buffer each iteration when doing + variable substitution ........ + + * res/res_fax.c: Delay sending an CED tone generated T.38 reinvite + to give the CED tone generating party time to send its own T.38 + reinvite. Also don't forward frames through the gateway if we are + negotiating T.38. + + * res/res_fax.c: fixed wording in a comment + +2011-07-11 10:57 +0000 [r327413] Tzafrir Cohen + + * /, main/Makefile: Merged revisions 327411 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327411 | tzafrir | 2011-07-11 13:46:34 +0300 (ב', 11 יול 2011) | + 5 lines fix building the Debian armhf (HardFloat) port Fixes + http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385 + (Missing pthreads) ........ + +2011-07-10 01:37 +0000 [r327359] Alexandr Anikin + + * addons/chan_ooh323.c, configs/chan_ooh323.conf.sample: Full T.38 + handshaking and fax detection Add full t.38 handshaking for + OOH323 that are required for newest T.38 gateway codes. Add fax + detection (cng tone, t38) and dialplan redirection to fax ext on + fax event detected. Add OOH323() function to set/get t38support + and faxdetect parameters. (closes issue ASTERISK-17754) Reported + by: irroot Patches: ooh323_faxdetect.patch uploaded by irroot + (license 52) issue19183-final.patch uploaded by may213 (license + 454) Tested by: may213, irroot Review: + https://reviewboard.asterisk.org/r/1174/ + +2011-07-08 22:25 +0000 [r327246] Jason Parker + + * main/stdtime, utils, codecs, utils/db1-ast/recno, apps, cel, + apps/confbridge, cdr, formats, codecs/gsm/src, + utils/db1-ast/hash, funcs, bridges, codecs/lpc10, + utils/db1-ast/db, codecs/g722, utils/db1-ast/mpool, main, + codecs/speex, channels/sip, pbx, res, res/ael, channels, + utils/db1-ast/btree: Add .o files to svn:ignore property, since + it's only ignored if locally configured to do so. + +2011-07-08 21:43 +0000 [r327212] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 327211 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327211 | rmudgett | 2011-07-08 16:41:58 -0500 (Fri, 08 Jul 2011) + | 9 lines INVITE 403 Forbidden response always retransmits the + maximum times. Asterisk sends a 403 Forbidden response if + authentication fails for an INVITE as required. However, it + ignores the ACK and keeps retransmitting the response. * Made not + delete the to-tag in the dialog so the expected ACK can be + matched with the dialog and stop the retransmissions. ........ + +2011-07-08 20:33 +0000 [r327116-327168] David Vossel + + * UPGRADE.txt, CHANGES: Adds entry in UPDATES.txt for removal of + formats/format_sln16.c. Fixes typo in CHANGES as well. + + * CHANGES: Updates CHANGES log to reflect new slinear read/write + file interpreters. + + * formats/format_sln.c, formats/format_sln16.c (removed): Support + for writing and reading raw slin files 8khz-192khz. + + * formats/format_attr_silk.c (removed), formats/format_attr_celt.c + (removed), res/res_format_attr_silk.c (added), + res/res_format_attr_celt.c (added): Moves celt and silk format + attribute files into res folder. It was inconsistent to have the + silk and celt format attribute modules in the format file + interpreter folder. + +2011-07-08 19:54 +0000 [r327107] Matthew Nicholson + + * main/pbx.c, tests/test_substitution.c, /: Merged revisions 327106 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul + 2011) | 11 lines Reset our ast_str before passing it on to + dialplan function backends. It is possible for a dialplan backend + to not modify the given buffer or ast_str and still return + success. This causes any previous value stored in the buffer to + be used as if the new function call provided it. Some functions + also append to the given buffer assuming it is empty. The + test_substitution unit test has also been modified to detect this + problem. (closes issue ASTERISK-17878) ........ + +2011-07-08 16:00 +0000 [r327045-327047] Russell Bryant + + * /, tests/test_netsock2.c: Merged revisions 327046 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r327046 | russell | 2011-07-08 11:00:05 -0500 (Fri, 08 + Jul 2011) | 2 lines Fix an error and add more log message info to + help see why this fails on FreeBSD. ........ + + * channels/chan_dahdi.c, /: Merged revisions 327044 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r327044 | russell | 2011-07-08 10:28:44 -0500 (Fri, 08 + Jul 2011) | 2 lines Resolve some set-but-unused-variable + warnings. ........ + +2011-07-08 01:26 +0000 [r327000] Richard Mudgett + + * main/pbx.c, /: Merged revisions 326985 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011) + | 12 lines Some code cleanup in pbx.c * Mostly comment and format + changes. * ast_context_remove_extension_callerid() and + ast_add_extension_nolock() will write lock the found specific + context. * ast_context_find() will now tolerate a NULL name. * + Eliminated some inlined versions of find_context() and + find_context_locked(). ........ + +2011-07-07 22:39 +0000 [r326943] Jason Parker + + * include/asterisk/celt.h: I think reviewboard broke this. The + whole file was doubled. + +2011-07-07 22:17 +0000 [r326855-326904] David Vossel + + * formats/format_attr_celt.c (added): Adds the format_attr_celt + file which was also missing from the CELT pass through patch. + + * include/asterisk/celt.h (added): Adds missing celt.h file from + celt pass-through support patch. + + * CHANGES: Fixes spelling errors in CHANGES as well as adding a few + entries for CELT and confbridge. + + * main/channel.c, main/format.c, res/res_rtp_asterisk.c, + main/frame.c, main/rtp_engine.c, channels/chan_sip.c, + include/asterisk/format.h, configs/codecs.conf.sample: Adds + pass-through support for codec CELT. This patch adds pass-through + support for CELT. CELT formats are defined in codecs.conf and can + be configured to any sample rate a CELT endpoint supports. This + patch also addresses a crash in channel.c resulting from a frame + list being freed incorrectly. This crash was discovered while + testing a CELT translator which had to split encoded audio into + multiple frames. The codec translator is not a part of this + patch, but may be contributed in the future. Review: + https://reviewboard.asterisk.org/r/1294/ + +2011-07-07 19:20 +0000 [r326842] Tilghman Lesher + + * /, res/res_http_post.c: Merged revisions 326830 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326830 | tilghman | 2011-07-07 14:17:19 -0500 (Thu, 07 Jul 2011) + | 1 line libgen.h is also needed on Darwin for basename(3) + ........ + +2011-07-07 17:24 +0000 [r326782] David Vossel + + * configs/confbridge.conf.sample, + apps/confbridge/include/confbridge.h, apps/app_confbridge.c, + apps/confbridge/conf_config_parser.c: Updates confbridge.conf + video documentation and adds dtmf action for releasing video src. + +2011-07-07 16:50 +0000 [r326750] Terry Wilson + + * utils/astdb2sqlite3.c, main/db.c: Use older functions out of + deference to older distros + +2011-07-07 16:18 +0000 [r326694] Jonathan Rose + + * res/res_config_odbc.c, /: Merged revisions 326689 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r326689 | jrose | 2011-07-07 11:04:51 -0500 (Thu, 07 Jul + 2011) | 10 lines res_odbc patch by tilghman to fix integers with + null values Addresses some improper sql statements in res_odbc + that would cause an update to fail on realtime peers due to + trying to set as "(NULL)" rather than an actual NULL. (closes + issue #1922STERISK-17791) Reported by: marcelloceschia Patches: + 20110505__issue19223.diff.txt uploaded by tilghman (license 14) + ........ + +2011-07-07 15:28 +0000 [r326682-326684] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 326683 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326683 | mnicholson | 2011-07-07 10:28:25 -0500 (Thu, 07 Jul + 2011) | 3 lines use sips: or sip: depending on the transport in + use when building reply digest URIs ........ + + * /, channels/chan_sip.c: Merged revisions 326681 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326681 | mnicholson | 2011-07-07 10:25:49 -0500 (Thu, 07 Jul + 2011) | 3 lines make the uri parameter used in reply digests more + standards compliant in certain cases by prepending "sip:" or + "sips:" to it ........ + +2011-07-07 09:49 +0000 [r326636] Tzafrir Cohen + + * contrib/scripts/live_ast: 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/ + +2011-07-06 20:58 +0000 [r326589] Terry Wilson + + * utils/db1-ast/btree/bt_open.c, utils/db1-ast/hash/hash_log2.c, + utils/db1-ast/hash/hsearch.c, utils/db1-ast/btree/bt_page.c, + utils/db1-ast/hash/page.h, utils/db1-ast/mpool, configure, + utils/db1-ast/btree/extern.h, utils/db1-ast/include/db.h, + main/db.c, utils/db1-ast/btree/bt_seq.c, + utils/db1-ast/recno/recno.h, main/Makefile, + utils/db1-ast/btree/bt_utils.c, utils/db1-ast/recno/rec_seq.c, + configure.ac, utils/db1-ast/btree/bt_close.c, CHANGES, + utils/db1-ast/hash/search.h, utils/db1-ast/hash/README, + utils/db1-ast/recno/rec_open.c, utils/db1-ast/hash/hash_bigkey.c, + utils/db1-ast/recno/rec_delete.c, Makefile, + utils/db1-ast/include, utils/db1-ast/hash/hash_buf.c, + utils/db1-ast/db, utils/db1-ast/libdb.map, + utils/db1-ast/include/ndbm.h, utils/db1-ast/include/compat.h, + utils/db1-ast/mpool/mpool.c, utils/db1-ast/btree/bt_debug.c, + main/asterisk.c, utils/db1-ast (added), + utils/db1-ast/btree/bt_split.c, utils, utils/db1-ast/recno, + utils/db1-ast/btree/bt_delete.c, + utils/db1-ast/include/circ-queue.h, tests/test_db.c, + utils/db1-ast/Makefile, utils/db1-ast/hash/extern.h, + utils/db1-ast/recno/rec_search.c, utils/db1-ast/btree/bt_get.c, + utils/db1-ast/hash/hash.c, utils/db1-ast/btree/btree.h, + utils/db1-ast/db/db.c, utils/db1-ast/hash/hash.h, + utils/db1-ast/include/mpool.h, utils/db1-ast/recno/rec_get.c, + utils/db1-ast/hash/hash_func.c, utils/utils.xml, + utils/astdb2sqlite3.c (added), utils/db1-ast/btree/bt_overflow.c, + UPGRADE.txt, utils/db1-ast/btree/bt_conv.c, + utils/db1-ast/btree/bt_search.c, utils/db1-ast/btree/bt_put.c, + utils/db1-ast/recno/rec_utils.c, utils/Makefile, + utils/db1-ast/hash/hash_page.c, utils/db1-ast/hash, + utils/db1-ast/mpool/README, utils/db1-ast/hash/ndbm.c, + main/db1-ast (removed), utils/db1-ast/recno/rec_close.c, + utils/db1-ast/recno/rec_put.c, utils/db1-ast/recno/extern.h, + utils/db1-ast/btree: Replace Berkeley DB with SQLite 3 There were + some bugs in the very ancient version of Berkeley DB that + Asterisk used. Instead of spending the time tracking down the + bugs in the Berkeley code we move to the much better documented + SQLite 3. Conversion of the old astdb happens at runtime by + running the included astdb2sqlite3 utility. The ast_db API with + SQLite 3 backend should behave identically to the old Berkeley + backend, but in the future we could offer a much more robust + interface. We do not include the SQLite 3 library in the source + tree, but instead rely upon the distribution-provided libraries. + SQLite is so ubiquitous that this should not place undue burden + on administrators. + +2011-07-06 17:39 +0000 [r326485-326544] David Vossel + + * channels/chan_sip.c: Fixes newlines from being stripped from out + of dialog sip MESSAGES. + + * /, res/res_timing_timerfd.c: Merged revisions 326484 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r326484 | dvossel | 2011-07-06 10:26:49 -0500 (Wed, 06 + Jul 2011) | 10 lines Reverts fix for timerfd locking issue. jrose + discovered a performance issue with this fix that prevents his + analog phones from working when using timerfd as a timing source. + Until it is understood what is causing this performance problem, + this patch is being reverted. ........ + +2011-07-05 22:11 +0000 [r326412] Tilghman Lesher + + * channels/chan_jingle.c, channels/chan_dahdi.c, + funcs/func_speex.c, /, channels/chan_sip.c, codecs/codec_speex.c, + funcs/func_aes.c, pbx/pbx_dundi.c, channels/chan_gtalk.c, + apps/app_queue.c, channels/chan_iax2.c, res/res_jabber.c, + apps/app_stack.c, channels/chan_mgcp.c, apps/app_voicemail.c: + Merged revisions 326411 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326411 | tilghman | 2011-07-05 17:08:29 -0500 (Tue, 05 Jul 2011) + | 14 lines Add the attribute "type" to each "" for + menuselect. This matters only when autoconf fails to detect that + weak linking is supported. External optional dependencies will + become optional in both cases, as they are removed at compile + time when not detected. However, runtime-optional modules are + made mandatory when weak linking is not found. This change + affects only the external optional dependencies; previously, they + were incorrectly required when weak linking support was not + detected. Patches: 20110702__issue18062__asterisk_trunk.diff.txt + by tilghman (License #5003) Tested by: iasgoscouk ........ + +2011-07-05 20:25 +0000 [r326368] Kinsey Moore + + * contrib/scripts/file.convert.sh (added): Prompt conversion script + Several variables in the script control which files are converted + and the source and destination formats. Patch-by: Trey Blancher + (closes AST-560) + +2011-07-05 17:35 +0000 [r326321] Richard Mudgett + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged + revisions 326291 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326291 | rmudgett | 2011-07-05 12:22:59 -0500 (Tue, 05 Jul 2011) + | 23 lines Used auth= parameter freed during "sip reload" causes + crash. If you use the auth= parameter and do a "sip reload" while + there is an ongoing call. The peer->auth data points to free'd + memory. The patch does several things: 1) Puts the authentication + list into an ao2 object for reference counting to fix the + reported crash during a SIP reload. 2) Converts the + authentication list from open coding to AST list macros. 3) Adds + display of the global authentication list in "sip show settings". + (closes issue ASTERISK-17939) Reported by: wdoekes Patches: + jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by + rmudgett Review: https://reviewboard.asterisk.org/r/1303/ JIRA + SWP-3526 ........ + +2011-07-05 16:46 +0000 [r326267] Mark Murawki + + * main/manager.c, CHANGES: New feature: AMI Action FilterAdd This + adds a new action, FilterAdd to the manager interface that allows + control over event filters for the current session (closes issue + ASTERISK-16795) Reported by: kobaz Tested by: kobaz,loloski + +2011-07-05 13:38 +0000 [r326210] Matthew Jordan + + * /, main/file.c: Merged revisions 326209 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326209 | mjordan | 2011-07-05 08:23:57 -0500 (Tue, 05 Jul 2011) + | 7 lines Updated filestream destructor to block until move is + complete when cache is used When a cache directory is used, the + process is forked and a mv command is executed to move the + temporary file to the permanent location. This caused issues with + voicemail, where a race condition occurred when the parent + expected the file to be in the permanent location prior to the mv + command completing. The parent process is now blocked until the + mv command completes. (closes issue ASTERISK-17724) Reported by: + Adiren P. Tested by: mjordan ........ + +2011-07-01 21:11 +0000 [r326145] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 326144 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r326144 | rmudgett | 2011-07-01 16:07:22 -0500 (Fri, 01 Jul 2011) + | 16 lines Better way to get chan and pvt lock for issue + ASTERISK-17431. Redoes -r308945 for issue ASTERISK-17431 deadlock + fix for sip_set_udptl_peer() and sip_set_rtp_peer(). * Lock the + channels in the defined order and avoid the need for a deadlock + avoidance loop. * Lock the channel before getting the pointer to + the private structure to be sure that the pointer will not change + due to a masquerade or channel hangup. * To preserve sanity, + check that chan and p->owner are the same. (Pointer rearangements + should not happen without the protection of locks because bad + things tend to happen otherwise.) ........ + +2011-07-01 16:36 +0000 [r326056-326101] Gregory Nietsky + + * CHANGES: Change CHANGES move the commits to the right place + r296249 r318141 Application changes + + * CHANGES: Change CHANGES move the commits to the right place in + the file missed in review + +2011-07-01 12:45 +0000 [r326006] Matthew Nicholson + + * res/res_fax.c, res/res_fax_spandsp.c: updated irroots info for + the authors section + +2011-06-30 21:05 +0000 [r325937] David Vossel + + * channels/chan_bridge.c: Fixes warning message caused by + confbridge playback chan not being answered. + +2011-06-30 20:47 +0000 [r325936] Richard Mudgett + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 325935 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325935 | rmudgett | 2011-06-30 15:39:45 -0500 (Thu, 30 Jun 2011) + | 11 lines Misc minor changes in chan_sip. * Add load failure + exit if primary SIP container(s) could not get created in + chan_sip.c:load_module(). * Removed a redundant static prototype. + * Some typos. * Some whitespace. ........ + +2011-06-30 20:33 +0000 [r325931] David Vossel + + * configs/confbridge.conf.sample, + apps/confbridge/include/confbridge.h, + include/asterisk/bridging.h, include/asterisk/dsp.h, + bridges/bridge_softmix.c, apps/app_confbridge.c, CHANGES, + main/bridging.c, main/dsp.c, apps/app_voicemail.c, + apps/confbridge/conf_config_parser.c: Video support for + ConfBridge. Review: https://reviewboard.asterisk.org/r/1288/ + +2011-06-30 20:24 +0000 [r325900] Matthew Jordan + + * /, apps/app_voicemail.c: Merged revisions 325877 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325877 | mjordan | 2011-06-30 15:09:48 -0500 (Thu, 30 + Jun 2011) | 9 lines Patched voicemail user option for emailbody / + emailsubject Incorporated changes per ASTERISK-16795; updated + unit tests to check for vmu->emailbody / vmu->emailsubject + (closes issue ASTERISK-16795) Reported by: mdeneen Tested by: + mjordan ........ + +2011-06-30 19:31 +0000 [r325864] Jonathan Rose + + * /, res/res_musiconhold.c: Merged revisions 325821 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325821 | jrose | 2011-06-30 14:17:32 -0500 (Thu, 30 Jun + 2011) | 10 lines Fixes an issue with Music on Hold classes losing + files in playlist when realtime is used. The bug occurs rather + intermittently and I relied on the reporters to test the patch. + After a sanity check and some testing, I'm giving it an OK. + (closes issue ASTERISK-17875) Reported by: David Cunningham + Patches: res_musiconhold.c.mohrt17875_v1 uploaded by Igor + Goncharovsky (license #5009) ........ + +2011-06-30 18:22 +0000 [r325815-325816] Matthew Nicholson + + * res/res_fax.c, include/asterisk/res_fax.h, CHANGES, + res/res_fax_spandsp.c: Fax gateway functionality (i.e. + translating between a T.30 terminal and a T.38 terminal). Can be + enabled on a channel by setting FAXOPT(gateway)=yes in the + dialplan. Big thanks to irroot for porting this code to use the + framehooks api. + + * main/frame.c: copy all flags on asterisk frames instead of just + the timing flag + +2011-06-29 21:50 +0000 [r325741] Kinsey Moore + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged + revisions 325740 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325740 | kmoore | 2011-06-29 16:49:21 -0500 (Wed, 29 Jun 2011) | + 7 lines chan_sip: cleanup from the introduction of ast_str Remove + the length field from sip_req and sip_pkt in chan_sip since they + are redundant (ast_str holds its own length) and refactor the + necessary functions. Review: + https://reviewboard.asterisk.org/r/1281/ ........ + +2011-06-29 19:02 +0000 [r325674] David Vossel + + * /, res/res_timing_timerfd.c: Merged revisions 325673 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325673 | dvossel | 2011-06-29 13:59:33 -0500 (Wed, 29 + Jun 2011) | 6 lines Fixes timerfd locking issue. (closes + ASTERISK-17867, ASTERISK-17415) Patches: fix uploaded by kobaz + Review: https://reviewboard.asterisk.org/r/1255/ ........ + +2011-06-29 18:18 +0000 [r325611-325616] Richard Mudgett + + * /, apps/app_queue.c: Merged revisions 325614 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325614 | rmudgett | 2011-06-29 13:16:45 -0500 (Wed, 29 Jun 2011) + | 5 lines Fixed some error exit cleanup in app_queue.c. * Fixed + error exit cleanup in app_queue.c copy_rules() and + reload_queue_rules(). ........ + + * /, apps/app_queue.c: Merged revisions 325610 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325610 | rmudgett | 2011-06-29 13:05:15 -0500 (Wed, 29 Jun 2011) + | 18 lines Response to QueueRule manager command does not contain + ActionID if it was specified. * Add ActionID support as + documented for the QueueRule AMI action. * Remove documentation + for ActionID with the Queues AMI action. The output does not + follow normal AMI response output and there is no place to put an + ActionID header. (closes issue AST-602) Reported by: Vlad + Povorozniuc Patches: jira_ast_602_v1.8.patch (license #5621) + patch uploaded by rmudgett Tested by: Vlad Povorozniuc, rmudgett + Review: https://reviewboard.asterisk.org/r/1295/ JIRA SWP-3575 + ........ + +2011-06-29 16:19 +0000 [r325538-325547] Matthew Nicholson + + * main/channel.c, /: Merged revisions 325545 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325545 | mnicholson | 2011-06-29 11:18:39 -0500 (Wed, 29 Jun + 2011) | 2 lines make framehooks prevent native bridging (for real + this time) ........ + + * apps/app_dial.c, main/rtp_engine.c, /: Merged revisions 325537 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun + 2011) | 2 lines don't do native/remote bridging if a framehook is + active on the channel ........ + +2011-06-29 06:39 +0000 [r325483] Gregory Nietsky + + * configs/queues.conf.sample, UPGRADE.txt, CHANGES, + apps/app_queue.c: Commit "distrotech" app_queue changes to Trunk + * Added general option negative_penalty_invalid default off. when + set members are seen as invalid/logged out when there penalty is + negative. for realtime members when set remove from queue will + set penalty to -1. * Added queue option autopausedelay when + autopause is enabled it will be delayed for this number of + seconds since last successful call if there was no prior call the + agent will be autopaused immediately. * Added member option + ignorebusy this when set and ringinuse is not will allow per + member control of multiple calls as ringinuse does for the Queue. + - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for + realtime members - QUEUE_MEMBER is now R/W supporting setting + paused, ignorebusy and penalty. (closes issue ASTERISK-17421) + (closes issue ASTERISK-17391) Reported by: irroot Tested by: + irroot, jrose Review: https://reviewboard.asterisk.org/r/1119/ + +2011-06-28 21:51 +0000 [r325417] Kevin P. Fleming + + * /, channels/chan_sip.c: Merged revisions 325416 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325416 | kpfleming | 2011-06-28 16:50:43 -0500 (Tue, 28 Jun + 2011) | 3 lines Fix random misspelling noticed on asterisk-users. + ........ + +2011-06-28 20:32 +0000 [r325345] David Vossel + + * /, channels/chan_sip.c: Merged revisions 325339 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r325339 | dvossel | 2011-06-28 15:31:00 -0500 (Tue, 28 Jun 2011) + | 4 lines Fixes locking inversion caused by holding sip pvt lock + during async_goto. (closes ASTERISK-17352) ........ + +2011-06-28 17:38 +0000 [r325213] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 325212 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325212 | rmudgett | 2011-06-28 12:30:16 -0500 (Tue, 28 + Jun 2011) | 7 lines Use the device name and not the channel name + to initialize the device state. Correct ASTERISK-11323 + implementation as I don't see how it ever worked as claimed when + it used the channel name and not the device name. (issue + ASTERISK-11323) ........ + +2011-06-28 16:04 +0000 [r325153] Jonathan Rose + + * /, res/res_musiconhold.c: Merged revisions 325152 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325152 | jrose | 2011-06-28 10:46:29 -0500 (Tue, 28 Jun + 2011) | 5 lines Fixes moh reload breaking custom mode moh classes + when the config file is untouched (closes issue ASTERISK-17730) + Reported by: sdolloff ........ + +2011-06-28 15:34 +0000 [r325151] David Vossel + + * channels/chan_sip.c: Fixes issue with video and text not being + reinvited correctly with directmedia If a SDP does not modify the + session, we ignore it. However, we were defaulting no text and + video support to true before checking to see if the sdp modified + anything or not. This would result in process_sdp ignoring an sdp + but removing video and text from the call during direct media + reinvites. + +2011-06-28 15:12 +0000 [r325092] Leif Madsen + + * /, build_tools/prep_tarball: Merged revisions 325091 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r325091 | lmadsen | 2011-06-28 10:12:00 -0500 (Tue, 28 + Jun 2011) | 1 line Remove line from prep_tarball that kills + mkrelease. ........ + +2011-06-28 00:07 +0000 [r325046] Terry Wilson + + * channels/chan_sip.c: Don't forget to build the Via when sending + MESSAGE + +2011-06-27 16:32 +0000 [r324961] Tilghman Lesher + + * /, main/asterisk.c: Merged revisions 324955 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324955 | tilghman | 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011) + | 5 lines Save and restore errno from within signal handlers. + This is recommended by the POSIX standard, as well as by the + sigaction(2) manpage for various platforms that we support (e.g. + Mac OS X). ........ + +2011-06-27 15:38 +0000 [r324915] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 324914 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324914 | rmudgett | 2011-06-27 10:37:19 -0500 (Mon, 27 Jun 2011) + | 21 lines When subscribing MWI to an unsolicited mailbox the + first notification is incorrect. A remote peer subscribed to MWI + with the unsolicited option and a local phone subscribed to the + remote mailbox. The notify message-summary events are sent + correctly except for the first one when subscribing, which will + always be 0. This means the phone MWI indicator will be wrong + until the mailbox read/unread count changes and the event is + fired. Looks like this is a regression from ASTERISK-16149. * Fix + the logic to check the cache and if allowed then fallback to + manually counting mailbox messages. (closes issue ASTERISK-17997) + Reported by: rsw686 Patches: jira_asterisk_17997_v1.8.patch + (license #5621) uploaded by rmudgett Tested by: rsw686 JIRA + SWP-3551 ........ + +2011-06-24 20:50 +0000 [r324850] Richard Mudgett + + * /, pbx/pbx_config.c: Merged revisions 324849 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324849 | rmudgett | 2011-06-24 15:46:01 -0500 (Fri, 24 Jun 2011) + | 15 lines Syntax errors in dialplan do not display the file + name. When issuing the CLI command "dialplan reload" syntax + errors and warnings are displayed on the console. The offending + line number is displayed on the console, but the file name is not + displayed. Errors caught in main/config.c do display the file + name. (closes issue ASTERISK-17985) Reported by: ulogic Patches: + pbx_config.patch uploaded by ulogic (License #5685) modified + format Tested by: rmudgett JIRA SWP-3554 ........ + +2011-06-24 16:50 +0000 [r324769] Jonathan Rose + + * include/asterisk/logger.h, /: Merged revisions 324768 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324768 | jrose | 2011-06-24 11:48:06 -0500 (Fri, 24 Jun 2011) | + 11 lines DTMF wasn't being logged on connected consoles when + enabled in logger.conf Previously in order for DTMF to be logged + in a connected console session, the user would have to do logger + set channel DTMF on. This corrects that so that it is on by + default. This issue was caused by an off by one error incurred by + a logger level count of 6 in logger.h where it should have been + 7. (closes issue: ASTERISK-17974) Reported by: Luke H ........ + +2011-06-23 18:56 +0000 [r324708-324709] Kinsey Moore + + * apps/app_confbridge.c: ConfBridge: redundant code cleanup There + is no reason to clean up features twice. Review: + https://reviewboard.asterisk.org/r/1279/ + + * /, channels/chan_sip.c: Merged revisions 324678 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r324678 | kmoore | 2011-06-23 13:29:17 -0500 + (Thu, 23 Jun 2011) | 11 lines Merged revisions 324643 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r324643 | kmoore | 2011-06-23 13:21:12 -0500 (Thu, 23 Jun 2011) | + 4 lines Addresses AST-2011-008, memory corruption and remote + crash in SIP driver. AST-2011-008 ........ ................ + +2011-06-23 18:31 +0000 [r324664-324689] David Vossel + + * /, channels/sip/reqresp_parser.c: Merged revisions 324685 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324685 | dvossel | 2011-06-23 13:31:00 -0500 (Thu, 23 Jun 2011) + | 8 lines Fixes sip crash when calling remove_uri_parameters with + NULL AST-2011-009 (closes issue ASTERISK-18017) Reported by: + jaredmauch ........ + + * /, main/features.c, channels/chan_iax2.c, + include/asterisk/frame.h: Merged revisions 324652 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r324652 | dvossel | 2011-06-23 13:23:21 -0500 + (Thu, 23 Jun 2011) | 20 lines Merged revisions 324634 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500 + (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) + | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver + Thanks to twilson for identifying the issue and providing the + patches. AST-2011-010 ........ ................ ................ + +2011-06-23 03:16 +0000 [r324558] Terry Wilson + + * /, tests/test_netsock2.c: Merged revisions 324557 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r324557 | twilson | 2011-06-22 22:10:38 -0500 (Wed, 22 + Jun 2011) | 5 lines Remove tests for parsing address with invalid + port getaddrinfo on OS X returns with EAI_NONAME error when + passed a port greater than 65535. Linux throws no error, so + remove the tests for now. ........ + +2011-06-22 19:17 +0000 [r324495] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 324491 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324491 | rmudgett | 2011-06-22 14:16:29 -0500 (Wed, 22 Jun 2011) + | 1 line Use correct variable for text SRTP media. ........ + +2011-06-22 19:12 +0000 [r324487] Terry Wilson + + * main/netsock2.c, /, channels/chan_sip.c, + include/asterisk/netsock2.h, tests/test_netsock2.c (added): + Merged revisions 324484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) + | 20 lines Stop sending IPv6 link-local scope-ids in SIP messages + The idea behind the patch listed below was used, but in a more + targeted manner. There are now address stringification functions + for addresses that are meant to be sent to a remote party. + Link-local scope-ids only make sense on the machine from which + they originate and so are stripped in the new functions. There is + also a host sanitization function added to chan_sip which is used + for when peer and dialog tohost fields or sip_registry hostnames + are used to craft a SIP message. Also added are some basic unit + tests for netsock2 address parsing. (closes issue ASTERISK-17711) + Reported by: ch_djalel Patches: + asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel + (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ + ........ + +2011-06-22 18:45 +0000 [r324480-324482] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 324481 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 Also fixed a + reference leak in an error path in sip_msg_send(). ........ + r324481 | rmudgett | 2011-06-22 13:41:20 -0500 (Wed, 22 Jun 2011) + | 19 lines Timout or error on INFO or MESSAGE transaction causes + call to be lost. When exchanging INFO messages within a call, 4xx + error causes the call to be disconnected although RFC 2976 + explicitly states that such transactions do not modify the state + of the dialog. When exchanging MESSAGE messages within a call, + 4xx error causes the call to be disconnected. To provide least + surprise, we should not disconnect the call since a MESSAGE is + like INFO in this case. (Implied by RFC 3428 Section 2) (closes + issue ASTERISK-17901) Reported by: neutrino88 Review: + https://reviewboard.asterisk.org/r/1257/ Review: + https://reviewboard.asterisk.org/r/1258/ JIRA SWP-3486 ........ + + * /, channels/chan_sip.c: Merged revisions 324479 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324479 | rmudgett | 2011-06-22 13:26:55 -0500 (Wed, 22 Jun 2011) + | 1 line Comments and whitespace in chan_sip.c ........ + +2011-06-21 21:55 +0000 [r324365-324422] David Vossel + + * apps/app_confbridge.c: Fixes issue with channel write format + being incorrectly restored when MOH is used in confbridge. + + * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 324364 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) + | 10 lines Fixes locking inversion issue in ast_async_goto() + During this function we can not hold the "chan" lock while doing + the masquerade, the explicit goto on the tmp chan, or the channel + alloc. Instead we need to get the channel lock, store off + information about the channel that we need, and then let the + channel lock go for the remainder of the function. Review: + https://reviewboard.asterisk.org/r/1275/ ........ + +2011-06-21 16:06 +0000 [r324304] Kinsey Moore + + * apps/app_confbridge.c: ConfBridge does not handle hangup properly + When playing back a prompt to a channel, confbridge neglects to + check for hangup events causing lockup condititions for hangups + that occur before actually joining the conference. This change + ensures that the user is removed from the conference in the event + of a premature hangup. Review: + https://reviewboard.asterisk.org/r/1277/ + +2011-06-21 15:49 +0000 [r324302] David Vossel + + * channels/chan_sip.c: Fixes issue with finding correct extension + when message context is used. + +2011-06-20 18:13 +0000 [r324242] Leif Madsen + + * /, configs/queuerules.conf.sample: Merged revisions 324241 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324241 | lmadsen | 2011-06-20 13:12:32 -0500 (Mon, 20 Jun 2011) + | 2 lines Remove extra 'the'. Reported by Vlad Povorozniuc + ........ + +2011-06-20 17:34 +0000 [r324238] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 324237 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324237 | twilson | 2011-06-20 12:33:07 -0500 (Mon, 20 Jun 2011) + | 12 lines Ignore media offers with a port of 0 Section 5.1 of + RFC3264 states: A port number of zero in the offer indicates that + the stream is offered but MUST NOT be used. (closes issue + ASTERISK-17845) Reported by: jacco Patches: issue19281_2.patch + uploaded by jacco (license 1277) Tested by: jacco, twilson + ........ + +2011-06-17 18:52 +0000 [r324177-324179] Leif Madsen + + * main/manager.c, /: Merged revisions 324178 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324178 | lmadsen | 2011-06-17 14:51:16 -0400 (Fri, 17 Jun 2011) + | 2 lines Add Username and Secret fields to manager Login action. + Pointed out by Vlad Povorozniuc ........ + + * /, apps/app_meetme.c: Merged revisions 324176 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324176 | lmadsen | 2011-06-17 14:38:40 -0400 (Fri, 17 Jun 2011) + | 2 lines Fix typo in documentation. Pointed out by Vlad + Povorozniuc ........ + +2011-06-17 18:23 +0000 [r324175] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 324174 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r324174 | rmudgett | 2011-06-17 13:23:19 -0500 (Fri, 17 + Jun 2011) | 5 lines Add header string to libpri debug output. Add + header string to libpri debug output so the libpri output can be + found/extracted easier from huge debug trace files. ........ + +2011-06-17 15:32 +0000 [r324131] Leif Madsen + + * main/pbx.c, /: Merged revisions 324115 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r324115 | lmadsen | 2011-06-17 11:14:54 -0400 (Fri, 17 Jun 2011) + | 3 lines Fix grammar in documentation for Goto() and GotoIf() + (closes issue ASTERISK-18023) Reported by: Tim Osman ........ + +2011-06-16 22:49 +0000 [r324050] Terry Wilson + + * main/channel.c, channels/chan_local.c, /, channels/chan_sip.c, + include/asterisk/channel.h: Merged revisions 324048 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 + Jun 2011) | 8 lines Lock the channel before calling the setoption + callback The channel needs to be locked before calling these + callback functions. Also, sip_setoption needs to lock the pvt and + a check p->rtp is non-null before using it. Review: + https://reviewboard.asterisk.org/r/1220/ ........ + +2011-06-16 18:13 +0000 [r323991] Richard Mudgett + + * /, tests/test_event.c: Merged revisions 323990 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323990 | rmudgett | 2011-06-16 13:12:32 -0500 (Thu, 16 Jun 2011) + | 5 lines The test_event unit test is occasionally failing. Wait + for the special posted event to process before adding a new + subscription. ........ + +2011-06-16 15:59 +0000 [r323673-323933] Terry Wilson + + * Makefile, /: Merged revisions 323932 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323932 | twilson | 2011-06-16 10:58:22 -0500 (Thu, 16 Jun 2011) + | 4 lines Don't assume ASTDBDIR exists It most likely doesn't on + FreeBSD ........ + + * /, tests/test_db.c: Merged revisions 323866 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323866 | twilson | 2011-06-15 15:03:58 -0500 (Wed, 15 Jun 2011) + | 2 lines Remove now-useless cast of ARRAY_LEN ........ + + * include/asterisk/utils.h, /: Merged revisions 323863 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r323863 | twilson | 2011-06-15 14:58:18 -0500 (Wed, 15 + Jun 2011) | 2 lines Make ARRAY_LEN() return the same type on x86 + and x86_64 systems ........ + + * /, tests/test_db.c: Merged revisions 323859 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323859 | twilson | 2011-06-15 14:45:20 -0500 (Wed, 15 Jun 2011) + | 2 lines Fix more ARRAY_LEN format string issues ........ + + * /, main/features.c: Merged revisions 323754 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r323754 | twilson | 2011-06-15 13:21:52 -0500 + (Wed, 15 Jun 2011) | 23 lines Merged revisions 323733 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r323733 | twilson | 2011-06-15 13:13:00 -0500 + (Wed, 15 Jun 2011) | 16 lines Merged revisions 323732 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) + | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a + recent DTMF change. This patch makes sure that dynamic features + are also checked when deciding whether or not to pass DTMF + through or store it for interpreting. (closes issue + ASTERISK-17914) Reported by: vrban ........ ................ + ................ + + * /, tests/test_db.c: Merged revisions 323672 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323672 | twilson | 2011-06-15 10:09:51 -0700 (Wed, 15 Jun 2011) + | 5 lines Cast ARRAY_LEN to size_t for ast_logging 32-bit and + 64-bit machines return different types for ARRAY_LEN(), so cast + it before using in a format string. ........ + +2011-06-15 16:49 +0000 [r323671] Richard Mudgett + + * /, tests/test_event.c, main/event.c: Merged revisions + 323669-323670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323669 | rmudgett | 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011) + | 21 lines [regression] Voicemail MWI is no longer sent. When + leaving a voicemail, the MWI message is never sent. The same + thing happens when checking a voicemail and marking it as read. + If you restart Asterisk, everything comes up at that state + correctly, but changes to the messages in voicemail causes the + light to not be set appropriately. Very easy to reproduce. * Made + ast_event_check_subscriber() return TRUE if there are ANY + subscribers to an event type when there are no restricting ie + values passed. This allows an event being queued to be queued. + (closes issue ASTERISK-18002) Reported by: lmadsen Tested by: + lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded + by rmudgett (License #5621) (closes issue ASTERISK-18019) + ........ r323670 | rmudgett | 2011-06-15 11:43:31 -0500 (Wed, 15 + Jun 2011) | 7 lines Add a test to the event unit tests to catch + ASTERISK-18002. The new tests check to see if there are ANY + subscribers to the event type when ast_event_check_subscriber() + is not passed any specific ie values. (issue ASTERISK-18002) + ........ + +2011-06-15 16:19 +0000 [r323621] Jonathan Rose + + * res/res_config_pgsql.c, /: Merged revisions 323610 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r323610 | jrose | 2011-06-15 11:09:24 -0500 (Wed, 15 Jun + 2011) | 7 lines Adds PQclear calls on result to various parts of + res_conf_pgsql (closes issue ASTERISK-17812) Reported by: + byronclark Patches: pgsql_pqclear.patch uploaded by byronclark + (license 1200) ........ + +2011-06-15 15:33 +0000 [r323609] Sean Bright + + * main/manager.c, /: Merged revisions 323608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 + (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 + (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun + 2011) | 25 lines Resolve a segfault/bus error when we try to map + memory that falls on a page boundary. The fix for ASTERISK-15359 + was incorrect in that it added 1 to the length of the mmap'd + region. The problem with this is that reading/writing to that + extra byte outside of the bounds of the underlying fd causes a + bus error. The real issue is that we are working with both a FILE + * and the raw fd underneath it and not synchronizing between + them. The code that was removed in ASTERISK-15359 was correct, + but we weren't flushing the FILE * before mapping the fd. Looking + at the manager code in 1.4 reveals that the FILE * in 'struct + mansession' is never used except to create a temporary file that + we immediately fdopen. This means we just need to write a 0 byte + to the fd and everything will just work. The other branches + require a call to fflush() which, while not a guaranteed fix, + should reduce the likelihood of a crash. This all makes sense in + my head. (closes issue ASTERISK-16460) Reported by: + Ravelomanantsoa Hoby (hoby) Patches: + issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license + #5060) ........ ................ ................ + +2011-06-15 13:45 +0000 [r323517] Kinsey Moore + + * apps/app_confbridge.c, CHANGES: CONFBRIDGE_INFO function to get + conference data Added the CONFBRIDGE_INFO dialplan function to + get information about a conference bridge including locked status + and number of parties, admins, and marked users. Review: + https://reviewboard.asterisk.org/r/1271/ + +2011-06-15 00:51 +0000 [r323397-323457] Richard Mudgett + + * /, main/event.c: Merged revisions 323456 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323456 | rmudgett | 2011-06-14 19:50:20 -0500 (Tue, 14 Jun 2011) + | 1 line Add missing break in ast_event_get_cached(). ........ + + * main/netsock2.c, main/dnsmgr.c, /: Merged revisions 323392,323394 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323392 | rmudgett | 2011-06-14 12:21:24 -0500 (Tue, 14 Jun 2011) + | 6 lines Add more strict hostname checking to + ast_dnsmgr_lookup(). Change suggested in review. Review: + https://reviewboard.asterisk.org/r/1240/ ........ r323394 | + rmudgett | 2011-06-14 12:21:39 -0500 (Tue, 14 Jun 2011) | 2 lines + Made ast_sockaddr_split_hostport() port warning msgs more + meaningful. ........ + +2011-06-14 17:03 +0000 [r323374] Terry Wilson + + * res/res_rtp_asterisk.c, main/rtp_engine.c, /, + channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged + revisions 323370 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) + | 10 lines Add rtpkeepalives back to 1.8 The RTP-engine + conversion left out support for handling rtpkeepalives. This + patch adds them back. (closes issue ASTERISK-17304) Reported by: + lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........ + +2011-06-14 16:47 +0000 [r323372] Jonathan Rose + + * /, channels/chan_sip.c: Merged revisions 323371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323371 | jrose | 2011-06-14 11:38:43 -0500 (Tue, 14 Jun 2011) | + 12 lines Changes contact use in build_peer to use the FORCE_RPORT + flag instead of RPORT_PRESENT It turned out that this was causing + NAT=Yes to always use rport when present which was against 1.6.2 + behavior and the check itself was redundant since the only way + this segment of code could be reached was if RPORT_PRESENT was + already evaluated as true earlier. (closes issue ASTERISK-17789) + Reported by: byronclark Patches: use_sip_nat_force_rport.patch + uploaded by byronclark (license 1200) ........ + +2011-06-14 14:37 +0000 [r323325] David Vossel + + * channels/chan_sip.c: Store sip peer name as var data on a + outofcall msg. + +2011-06-13 20:44 +0000 [r323272] Kinsey Moore + + * apps/confbridge/conf_config_parser.c: Config inheritance doesn't + work with ConfBridge() menu definitions Current behavior in + ConfBridge menu definitions is that first definition takes + precedence, even in templated situations. This change allows + inheritance and overriding to work as expected so that the last + definition takes precedence. (closes ASTERISK-17986) Review: + https://reviewboard.asterisk.org/r/1267/ + +2011-06-13 19:54 +0000 [r323214] Leif Madsen + + * main/channel.c, /: Merged revisions 323213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323213 | lmadsen | 2011-06-13 15:51:52 -0400 (Mon, 13 Jun 2011) + | 6 lines Avoid dividing by zero with L() option to Dial() + Reported by: nicolasom Patches: issue-17995.patch - nicolasom + (License #5994) ........ + +2011-06-13 19:43 +0000 [r323212] David Vossel + + * channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/sip/include/sip.h: Addition of + "outofcall_message_context" sip.conf option. Review: + https://reviewboard.asterisk.org/r/1265/ + +2011-06-13 19:03 +0000 [r323155] Leif Madsen + + * /, res/res_agi.c: Merged revisions 323154 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323154 | lmadsen | 2011-06-13 15:00:41 -0400 (Mon, 13 Jun 2011) + | 6 lines Tweak documentation for AGI Hangup command. (closes + issue ASTERISK-17999) Reported by: Ben Klang Patches: + hangup-doc.diff - uploaded by Ben Klang (License #5876) ........ + +2011-06-13 14:38 +0000 [r323106-323107] Kinsey Moore + + * apps/confbridge/include/confbridge.h, apps/app_confbridge.c: MOH + for only user not working with ConfBridge This adds the + playing_moh flag to the conference_bridge_user struct that + signifies when MOH should be playing so code doesn't have to + guess whether MOH is playing. This change also adds the necessary + checking to ensure that MOH continues playing for a single user + in a conference after the join sound is played when configured to + do so. (closes ASTERISK-17988) Review: + https://reviewboard.asterisk.org/r/1263/ + + * apps/app_confbridge.c: ConfBridge: Use of bridge or user profiles + that don't exist Bridge and user profiles are not checked for + existence before use. The lack of a fully formed bridge profile + can cause a segfault when sounds are accessed. This change + ensures that bridge and user profiles exist prior to usage + attempts. Review: https://reviewboard.asterisk.org/r/1264/ + +2011-06-10 19:22 +0000 [r323041] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 323040 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r323040 | mnicholson | 2011-06-10 14:20:41 -0500 (Fri, 10 Jun + 2011) | 5 lines Unlock the sip channel during fax detection like + chan_dahdi does to prevent a deadlock with ast_autoservice_stop. + (closes issue ASTERISK-17798) tested by mnicholson ........ + +2011-06-10 15:30 +0000 [r322866-322982] Terry Wilson + + * /, main/db.c: Merged revisions 322981 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322981 | twilson | 2011-06-10 08:29:00 -0700 (Fri, 10 Jun 2011) + | 11 lines Avoid a DB1 infinite loop bug Explicity check the last + entry in the DB and make sure that we don't iterate past it. + Since there can be no duplicates, this just makes sure that we + stop after matching the last key. This patch also refactors the + code to get away from some code duplication. A previous patch + added many astdb tests and this patch passed them. Review: + https://reviewboard.asterisk.org/r/1259/ ........ + + * /, tests/test_db.c (added): Merged revisions 322923 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r322923 | twilson | 2011-06-09 19:33:23 -0700 (Thu, 09 + Jun 2011) | 2 lines Add some astdb unit tests ........ + + * /, include/asterisk/astdb.h: Merged revisions 322865 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r322865 | twilson | 2011-06-09 15:29:20 -0700 (Thu, 09 + Jun 2011) | 4 lines Correct ast_db_deltree documentation + ast_db_deltree returns -1 on error, otherwise the number of + deletions ........ + +2011-06-09 17:43 +0000 [r322808] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 322807 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322807 | mnicholson | 2011-06-09 12:37:07 -0500 (Thu, 09 Jun + 2011) | 5 lines don't drop any voice frames when checking for + T.38 during early media (closes issue ASTERISK-17705) Review: + https://reviewboard.asterisk.org/r/1186/ patch by oej reported by + oej ........ + +2011-06-09 16:47 +0000 [r322750] Richard Mudgett + + * /, apps/app_directed_pickup.c, main/features.c, + include/asterisk/features.h: Merged revisions 322749 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 + Jun 2011) | 15 lines Remove potential deadlock in call pickup + race. Deadlock is possible in ast_do_pickup() when holding the + target channel lock and trying to get the chan channel lock. + Also, holding the target lock when calling + ast_channel_masquerade() is not a good idea because that routine + does deadlock avoidance. * Removed the need to hold the target + lock after marking the target with a datastore and getting the + connected line data off of the target channel. * Moved + can_pickup() to ast_can_pickup() in features.c. Now all the call + pickup methods use the same basic call pickup availability check. + Review: https://reviewboard.asterisk.org/r/1234/ ........ + +2011-06-09 11:05 +0000 [r322544] Damien Wedhorn + + * channels/chan_skinny.c: Add autoanswer to skinny. Autoanswer + added to skinny based on incoming chan var SKINNY_AUTOANSWER. + Initial value must be the time to autoanswer in ms, then + optionally :BEEP to play a tone when answered and :MUTE to mute + the mic when answering. eg 3000:MUTE:BEEP will ring for 3 secs, + then answer, mute the mic, and play a beep. just 3000 would + answer afer 3 secs of ringing with no beep and full two way + audio. + +2011-06-08 20:48 +0000 [r322426-322485] Richard Mudgett + + * /, apps/app_queue.c: Merged revisions 322484 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322484 | rmudgett | 2011-06-08 15:46:55 -0500 (Wed, 08 Jun 2011) + | 15 lines Ring all queue with more than 255 agents will cause + crash. 1. Create a ring-all queue with 500 permanent agents. 2. + Call it. 3. Asterisk will crash. The watchers array in + app_queue.c has a hard limit of 255. Bounds checking is not done + on this array. No sane person should put 255 people in a ring-all + queue, but we should not crash anyway. * Added bounds checking to + the watchers array. JIRA AST-464 JIRA SWP-2903 ........ + + * main/dnsmgr.c, /: Merged revisions 322425 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322425 | rmudgett | 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011) + | 16 lines SRV lookup attempted for SIP peers listed as an IP + address. Asterisk attempts to SRV lookup a host name even if the + host name is an IP address. Regression introduced when IPv6 + support was added. * Restored the check in ast_dnsmgr_lookup() to + see if the given host name is an IP address. The IP address could + be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815) + Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett + Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett + (License #5621) Review: https://reviewboard.asterisk.org/r/1240/ + ........ + +2011-06-08 11:38 +0000 [r322381] Damien Wedhorn + + * channels/chan_skinny.c: Remove skinny do_monitor and use + ast_sched_start instead The do_monitor seemed to be there for + task scheduling and network monitoring. However, the network + monitoring has a dedicated thread so the ast_io_wait was + basically just a usleep as it didn't actually seem to be + monitoring anything. Review: + https://reviewboard.asterisk.org/r/1256/ + +2011-06-08 06:45 +0000 [r322323] Gregory Nietsky + + * /, channels/chan_sip.c: Merged revisions 322322 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322322 | irroot | 2011-06-08 08:18:38 +0200 (Wed, 08 Jun 2011) | + 18 lines Make handle_request_publish do dialog expiration and + destruction. This patch fixes handle_request_publish so that it + does dialog expiration and destruction. Without this patch the + incoming PUBLISH requests will get stuck in the dialog list. + Restarting asterisk is the only way to remove them. Personal + observation on one system the server hung up while looping + through the channels rendering asterisk unusable and all sip + phones unregisterd when they try reregister more requests are + added. (closes issue #18898) Reported by: gareth Tested by: + loloski, Chainsaw, wimpy, se, kuj, irroot Jira: + https://issues.asterisk.org/jira/browse/ASTERISK-17915 Review: + https://reviewboard.asterisk.org/r/1253 ........ + +2011-06-07 23:14 +0000 [r322284] Richard Mudgett + + * channels/chan_sip.c, include/asterisk/message.h: Correct some + whitespace and a reference debug message. + +2011-06-07 19:17 +0000 [r322244] Russell Bryant + + * res/res_jabber.c: Actually check the "sendtodialplan" option + setting for xmpp. (closes issue ASTERISK-17978) Reported by: + elguero Patches: stop_messages_going_to_dialplan.patch (license + #5026) + +2011-06-07 18:01 +0000 [r322190] Paul Belanger + + * configs/sip_notify.conf.sample, /: Merged revisions 322189 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322189 | pabelanger | 2011-06-07 13:59:13 -0400 (Tue, 07 Jun + 2011) | 4 lines Use correct syntax for 'sip notify snom-reboot' + (closes issue ASTERISK-17915) ........ + +2011-06-06 19:39 +0000 [r322111-322128] Gregory Nietsky + + * apps/app_queue.c: Remove Unused Var Warning + rt_handle_member_record + + * apps/app_queue.c: Refactor rt_handle_member_record Review: + https://reviewboard.asterisk.org/r/1172 + +2011-06-06 19:15 +0000 [r322070] Jonathan Rose + + * include/asterisk/logger.h, /, main/asterisk.c: Merged revisions + 322069 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r322069 | jrose | 2011-06-06 14:07:56 -0500 (Mon, 06 Jun 2011) | + 8 lines Fixes level toggling for logger set levels since it was + reversed (closes issue ASTERISK-17850) Reported by: Luke H Tested + by: jrose, Luke H Review: + https://reviewboard.asterisk.org/r/1244/ ........ + +2011-06-03 22:15 +0000 [r321814-321927] Richard Mudgett + + * cel/cel_radius.c, /, cdr/cdr_radius.c: Merged revisions 321926 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321926 | rmudgett | 2011-06-03 17:09:36 -0500 (Fri, 03 Jun 2011) + | 18 lines Asterisk crash when unloading cdr_radius/cel_radius. + The rc_openlog() API call is passed a string that is used by + openlog() to format log messages. The openlog() does not copy the + string it just keeps a pointer to it. When the module is + unloaded, the string is gone from memory. Depending upon module + load order and if the other module then has an error, a crash + happens. * Pass rc_openlog() a strdup'd string with the + understanding that there will be a small memory leak if the + cdr_radius/cel_radius modules are unloaded. * Call rc_destroy() + to free the rc handle memory when the module is unloaded. JIRA + AST-483 JIRA SWP-3062 ........ + + * /, main/ccss.c: Merged revisions 321924 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321924 | rmudgett | 2011-06-03 16:49:17 -0500 (Fri, 03 Jun 2011) + | 5 lines Be more explicit for CCSS generic device state event + subscription. Make CCSS generic device state event subscription + specify the AST_EVENT_IE_STATE ie exists to be safe. ........ + + * /, tests/test_event.c, main/event.c: Merged revisions 321871 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321871 | rmudgett | 2011-06-03 15:58:13 -0500 (Fri, 03 Jun 2011) + | 27 lines Event subscription fixes. Must commit the subscription + fixes together with the integration subscription tests. The + subscription fixes cause an erroneously passing test to fail. The + new subscription tests detect errors without the subscription + fixes. * Added missing event_names[] table entry. * Reworked + ast_event_check_subscriber()/match_sub_ie_val_to_event() to + correctly detect if a subscriber exists for the proposed event. * + Made match_ie_val() and match_sub_ie_val_to_event() check the + buffer length for RAW payload types. * Fixed error handling + memory leak in ast_event_sub_activate(), ast_event_unsubscribe(), + and ast_event_queue(). * Made ast_event_new() and + ast_event_check_subscriber() better protect themselves from an + invalid payload type. * Added container lock protection between + removing old cache events and adding the new cached event in + ast_event_queue_and_cache()/event_update_cache(). * Added new + event subscription tests. ........ + + * include/asterisk/event.h, /, channels/chan_sip.c, main/event.c, + channels/chan_iax2.c: Merged revisions 321812-321813 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r321812 | rmudgett | 2011-06-03 14:55:21 -0500 (Fri, 03 + Jun 2011) | 1 line Correct IAX2 and SIP event subscription + description string. ........ r321813 | rmudgett | 2011-06-03 + 14:56:09 -0500 (Fri, 03 Jun 2011) | 1 line Constify subscription + description parameter string. ........ + +2011-06-03 18:25 +0000 [r321752] Russell Bryant + + * tests/test_astobj2.c, main/astobj2.c: Fix some astobj2 iterator + breakage, add another unit test. Review: + https://reviewboard.asterisk.org/r/1254/ + +2011-06-03 13:18 +0000 [r321689] Leif Madsen + + * /, configs/queues.conf.sample: Merged revisions 321685 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321685 | lmadsen | 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011) + | 5 lines Also document the 'queue-minute' option. (closes issue + #19386) Reported by: juanmol ........ + +2011-06-02 22:09 +0000 [r321617] Russell Bryant + + * channels/chan_sip.c: Fix message destination extension. Don't + send all messages to 's'. Get the destination from the request + URI. (Found using automated test cases). + +2011-06-01 23:12 +0000 [r321548] Richard Mudgett + + * main/cdr.c, /: Merged revisions 321547 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321547 | rmudgett | 2011-06-01 18:11:55 -0500 (Wed, 01 Jun 2011) + | 1 line CDR comment tweaks. ........ + +2011-06-01 21:31 +0000 [r321546] Russell Bryant + + * main/channel.c, channels/chan_sip.c, configs/jabber.conf.sample, + include/asterisk/message.h (added), include/asterisk/jabber.h, + include/asterisk/channel.h, configs/sip.conf.sample, + include/asterisk/_private.h, CHANGES, res/res_jabber.c, + main/message.c (added), channels/sip/include/sip.h, + main/asterisk.c: Support routing text messages outside of a call. + Asterisk now has protocol independent support for processing text + messages outside of a call. Messages are routed through the + Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. + There are options in sip.conf and jabber.conf that enable these + features. There is a new application, MessageSend(). There are + two new functions, MESSAGE() and MESSAGE_DATA(). Documentation + will be available on the project wiki, wiki.asterisk.org. Thanks + to Terry Wilson for the assistance with development and to David + Vossel for helping with some additional testing. Review: + https://reviewboard.asterisk.org/r/1042/ + +2011-06-01 20:11 +0000 [r321538] Brett Bryant + + * /, apps/app_voicemail.c: Merged revisions 321537 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r321537 | bbryant | 2011-06-01 16:10:02 -0400 (Wed, 01 + Jun 2011) | 8 lines This patch fixes an issue with using the + wrong voicemail folders with greetings. (closes issue #17871) + Reported by: edhorton Patches: digium_bug_17871_2 uploaded by + fhackenberger (license 592) Tested by: edhorton, fhackenberger + ........ + +2011-06-01 10:45 +0000 [r321529] Alexandr Anikin + + * addons/chan_ooh323.c, /, addons/ooh323c/src/ooh245.c, + addons/ooh323c/src/oochannels.c: Merged revisions 321528 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321528 | may | 2011-06-01 14:40:19 +0400 (Wed, 01 Jun 2011) | 14 + lines Fix double alerting, add forced alerting before answer Fix + double alerting (it wasn't fixed here by issue #18542) Add forced + alerting before connect (if it wasn't before) Try to send all + packets from outgoing queue rather than one only Call goes into + clearing state when disconnect command is received (closes issue + #19361) Reported by: vmikhelson Patches: issue19361-3.patch + uploaded by may213 (license 454) Tested by: vmikhelson ........ + +2011-05-31 20:55 +0000 [r321518] Richard Mudgett + + * include/asterisk/acl.h, /, include/asterisk/dnsmgr.h: Merged + revisions 321517 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321517 | rmudgett | 2011-05-31 15:54:35 -0500 (Tue, 31 May 2011) + | 1 line Update some comments. ........ + +2011-05-31 19:01 +0000 [r321516] David Vossel + + * channels/chan_local.c, /: Merged revisions 321515 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r321515 | dvossel | 2011-05-31 13:52:54 -0500 (Tue, 31 + May 2011) | 12 lines Chan_local locking cleanup. This patch + removes all of the unnecessary deadlock avoidance loops that + occur in chan_local. It also resolves an issue with a deadlock + triggered by local channel optimizations. (issue #18028) Review: + https://reviewboard.asterisk.org/r/1231/ ........ + +2011-05-31 16:06 +0000 [r321512] Leif Madsen + + * /, channels/chan_sip.c: Merged revisions 321511 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321511 | lmadsen | 2011-05-31 12:04:47 -0400 (Tue, 31 May 2011) + | 8 lines Enhance NOTICE message to know who couldn't access the + dialplan. (closes issue #19390) Reported by: lmadsen Patches: + __20110531-sip-notice-tweak.txt uploaded by lmadsen (license 10) + Tested by: russell ........ + +2011-05-28 00:29 +0000 [r321338-321445] Richard Mudgett + + * /, res/res_agi.c: Merged revisions 321436 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321436 | rmudgett | 2011-05-27 19:27:52 -0500 (Fri, 27 May 2011) + | 4 lines Some hagi launch cleanup. Inspired by issue 19256. This + patch would also fix the crash. ........ + + * main/srv.c, /: Merged revisions 321392 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321392 | rmudgett | 2011-05-27 18:45:41 -0500 (Fri, 27 May 2011) + | 12 lines Crash when using hagi and no servers are available. + When none of the servers returned by the SRV querey respond, + asterisk crashes. The problem is that if the loop over all the + SRV entries finishes then the srv_context has already been + cleaned up. * Make ast_srv_cleanup() check to see if the context + is already cleaned up. (closes issue #19256) Reported by: + byronclark ........ + + * /, apps/app_privacy.c, UPGRADE.txt, CHANGES: Merged revisions + 321337 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 Also revert + -r321331 and -r321332. ........ r321337 | rmudgett | 2011-05-27 + 17:06:43 -0500 (Fri, 27 May 2011) | 7 lines The app_privacy args + have undocumented "options" position, interferes with "context" + position. * Add documention for unused "options" position to + match existing code. (closes issue #19273) Reported by: + mdavenport ........ + +2011-05-27 21:40 +0000 [r321334] Leif Madsen + + * /, main/features.c: Merged revisions 321333 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321333 | lmadsen | 2011-05-27 17:40:23 -0400 (Fri, 27 May 2011) + | 7 lines Allow parking lot hints and musicclass to be set. + (closes issue #19378) Reported by: sboily_proformatique Patches: + pf_parkinghint_music_fix uploaded by sboily proformatique + (license 206) Tested by: russell ........ + +2011-05-27 21:37 +0000 [r321331-321332] Richard Mudgett + + * UPGRADE.txt: Add note about PrivacyManager to UPGRADE.txt + + * /, apps/app_privacy.c, CHANGES: Merged revisions 321330 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321330 | rmudgett | 2011-05-27 16:31:25 -0500 (Fri, 27 May 2011) + | 8 lines The app_privacy args have undocumented "options" + position, interferes with "context" position. * Add documention + for unused "options" position to match existing code. The + trunk(v1.10) version will remove the unused options position. + (closes issue #19273) Reported by: mdavenport ........ + +2011-05-27 16:35 +0000 [r321289] Jonathan Rose + + * /, channels/sip/reqresp_parser.c: Merged revisions 321273 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321273 | jrose | 2011-05-27 09:59:34 -0500 (Fri, 27 May 2011) | + 3 lines markm committed a patch I was working on yesterday, this + fixes it to mesh up with suggestions by mnicholson. ........ + +2011-05-27 08:37 +0000 [r321212] Alec L Davis + + * /, main/features.c: Merged revisions 321211 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321211 | alecdavis | 2011-05-27 20:31:15 +1200 (Fri, 27 May + 2011) | 16 lines Fix *8 directed pickup locks system during + pickupsound play out move playout from sip_pickup_thread to + bridge using BRIDGE_PLAY_SOUND method, This stop the clash of 2 + threads trying to write audio to same channel. In addition fixes + choppy audio beep in issue 19177. (issue #18654) (issue #19177) + Reported by: Docent Patches: review1232-1.8.diff.txt alecdavis + (license 585) Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/1232/ ........ + +2011-05-26 21:50 +0000 [r321101-321156] Mark Murawki + + * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged + revisions 321155 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321155 | markm | 2011-05-26 17:48:45 -0400 (Thu, 26 May 2011) | + 10 lines Fixed build problem with dev mode enabled, which was + caused by commit 321100. Reformulated patch to be more generic. + Moved the sip uri parse variable initalization to parse_uri_full + in reqresp_parser.c. This will ensure that any use of parse uri + will have null output variables if the parse fails. (closes issue + #19346) Reported by: kobaz Tested by: kobaz,JonathanRose Review: + [full review board URL with trailing slash] ........ + + * main/netsock2.c, /, channels/chan_sip.c: Merged revisions 321100 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321100 | markm | 2011-05-26 16:09:35 -0400 (Thu, 26 May 2011) | + 11 lines ast_sockaddr_resolve() in netsock2.c may deref a null + pointer Added a null check in netsock2 ast_sockaddr_resolve() as + well as added default initalizers in chan_sip + parse_uri_legacy_check() to make sure that invalid uris will make + null (and not undefined) user,pass,domain,transport variables + (closes issue #19346) Reported by: kobaz Patches: netsock2.patch + uploaded by kobaz (license 834) Tested by: kobaz, Marquis + ........ + +2011-05-26 18:10 +0000 [r321045] Richard Mudgett + + * /, include/asterisk/netsock2.h: Merged revisions 321044 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321044 | rmudgett | 2011-05-26 13:10:17 -0500 (Thu, 26 May 2011) + | 1 line Update ast_sockaddr comment with an important note. + ........ + +2011-05-26 17:35 +0000 [r321043] Terry Wilson + + * main/rtp_engine.c, /: Merged revisions 321042 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r321042 | twilson | 2011-05-26 10:29:54 -0700 (Thu, 26 May 2011) + | 6 lines Initialize stack-allocated ast_sockaddrs before use It + is important to always initialize ast_sockaddrs before use--even + if they are passed to ast_sockaddr_copy as the underlying storage + could be bigger than what ends up being copied--leaving part of + the data unitialized. ........ + +2011-05-26 16:54 +0000 [r321003] Russell Bryant + + * /, channels/chan_alsa.c: Merged revisions 320947 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320947 | russell | 2011-05-26 10:57:13 -0500 (Thu, 26 + May 2011) | 2 lines Remove some variables that were set but + unused. ........ + +2011-05-26 15:55 +0000 [r320946] Terry Wilson + + * main/channel.c, main/utils.c, include/asterisk/stringfields.h: + Use va_copy for stringfields The ast_string_field_build_va + functions were written to take to separate va_lists to work + around FreeBSD 4 not having va_copy defined. In the end, we don't + support anything using gcc < 3 anyway because we use va_copy all + over the place anyway. This patch just simplifies things by + removing the second va_list function arguments in favor of + va_copy. Review: https://reviewboard.asterisk.org/r/1233/ --This + line, and those below, will be ignored-- M + include/asterisk/stringfields.h M main/utils.c M main/channel.c + +2011-05-25 22:28 +0000 [r320820-320884] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 320883 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320883 | rmudgett | 2011-05-25 17:25:18 -0500 (Wed, 25 May 2011) + | 17 lines Native SIP CCSS sends bad CC cancel SUBSCRIBE message. + The SUBSCRIBE message used to cancel a CC request has incorrect + To/From SIP headers. They are reversed and the dialog tags are + the same when they should not be. If pedantic mode was disabled, + then the cancel would have succeeded despite the incorrect + message. * The SIP_OUTGOING flag was not set correctly for the + dialog and I had to move some CC subscribe handling code as a + result. * Initialized the dialog subscribed type to + CALL_COMPLETION earlier. If a CC request SUBSCRIBE message comes + in and the CC instance is not found, the 404 response was + duplicated. JIRA AST-568 JIRA SWP-3493 ........ + + * apps/app_dial.c, main/channel.c, main/manager.c, /, + apps/app_meetme.c, apps/app_fax.c, main/features.c, CHANGES, + apps/app_queue.c, UPGRADE-1.8.txt: Merged revisions 320823 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) + | 18 lines The AMI Newstate event contains different information + between v1.4 and v1.8. The addition of connected line support in + v1.8 changes the behavior of the channel caller ID somewhat. The + channel caller ID value no longer time shares with the connected + line ID on outgoing call legs. The timing of some AMI + events/responses output the connected line ID as caller ID. These + party ID's are now separate. * The ConnectedLineNum and + ConnectedLineName headers were added to many AMI events/responses + if the CallerIDNum/CallerIDName headers were also present. + (closes issue #18252) Reported by: gje Tested by: rmudgett + Review: https://reviewboard.asterisk.org/r/1227/ ........ + + * main/channel.c, /, main/format_cap.c, main/features.c, + include/asterisk/channel.h: Merged revisions 320796 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320796 | rmudgett | 2011-05-25 11:23:11 -0500 (Wed, 25 + May 2011) | 17 lines Give zombies a safe channel driver to use. + Recent crashes from zombie channels suggests that they need a + safe home to goto. When a masquerade happens, the physical part + of the zombie channel is hungup. The hangup normally sets the + channel private pointer to NULL. If someone then blindly does a + callback to the channel driver, a crash is likely because the + private pointer is NULL. The masquerade now sets the channel + technology of zombie channels to the kill channel driver. Related + to the following issues: (issue #19116) (issue #19310) Review: + https://reviewboard.asterisk.org/r/1224/ ........ + +2011-05-25 15:43 +0000 [r320772] Gregory Nietsky + + * funcs/func_channel.c, CHANGES: CHANNEL(pickupgroup) Allow Setting + / Reading the pickupgroup of a channel with func_channel.c + (closes issue #19045) Reported by: irroot Review: + https://reviewboard.asterisk.org/r/1148/ + +2011-05-25 00:52 +0000 [r320717] Terry Wilson + + * /, addons/chan_mobile.c: Merged revisions 320716 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320716 | twilson | 2011-05-24 17:49:10 -0700 (Tue, 24 + May 2011) | 4 lines Cast data as char * before using S_OR This is + required for compiling successfully under dev mode ........ + +2011-05-23 18:00 +0000 [r320651] Richard Mudgett + + * main/manager.c, /, CHANGES: Merged revisions 320650 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320650 | rmudgett | 2011-05-23 12:53:44 -0500 (Mon, 23 + May 2011) | 16 lines Add ConnectedLineNum/Name headers to output + of AMI action Status. * Add ConnectedLineNum and + ConnectedLineName headers to the output of the AMI action Status. + This makes it easier to find out who the channel is connected to + without having to lookup BridgedChannel or when they are + connected to an application (e.g.: VoiceMail) which has no + bridged channel. * Bridged channels with no CallerID had "" + instead of "" output, that might be a bug as "" + was what older versions used. (closes issue #18158) Reported by: + gareth Patches: svn-292308.diff uploaded by gareth (license 208) + ........ + +2011-05-23 16:28 +0000 [r320606] David Vossel + + * main/tcptls.c, /: Merged revisions 320568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r320568 | dvossel | 2011-05-23 11:18:33 -0500 + (Mon, 23 May 2011) | 14 lines Merged revisions 320562 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r320562 | dvossel | 2011-05-23 11:15:18 -0500 (Mon, 23 May 2011) + | 9 lines Adds missing part to the ast_tcptls_server_start fails + second attempt to bind patch. (closes issue #19289) Reported by: + wdoekes Patches: + issue19289_delay_old_address_setting_tcptls_2.patch uploaded by + wdoekes (license 717) ........ ................ + +2011-05-23 16:20 +0000 [r320579] Tilghman Lesher + + * /, configure, configure.ac: Merged revisions 320573 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320573 | tilghman | 2011-05-23 11:19:32 -0500 (Mon, 23 + May 2011) | 7 lines GNU libiconv uses symbol "libiconv_open" + instead of "iconv_open". (closes issue #19344) Reported by: + rohanl Patches: iconv-check.patch uploaded by rohanl (license + 1284) ........ + +2011-05-23 15:48 +0000 [r320561] Kevin P. Fleming + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 320560 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320560 | kpfleming | 2011-05-23 10:47:14 -0500 (Mon, 23 May + 2011) | 4 lines Don't generate spurious "No: command not found" + messages when running the configure script on a system that has + neither gmime-config nor pkg-config. ........ + +2011-05-23 14:40 +0000 [r320505] Jonathan Rose + + * /, channels/chan_sip.c: Merged revisions 320504 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320504 | jrose | 2011-05-23 09:33:20 -0500 (Mon, 23 May 2011) | + 10 lines Fixes segfault occuring in chan_sip.c at + __set_address_from_contact Checks to see if domain contains + anything before sending it off to ast_sockaddr_resolve which is + where the segfault was occuring due to null str. (closes issue + #18857) Reported by: sybasesql Review: + https://reviewboard.asterisk.org/r/1225/ ........ + +2011-05-22 23:36 +0000 [r320446] Tilghman Lesher + + * /, res/res_odbc.c: Merged revisions 320445 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r320445 | tilghman | 2011-05-22 18:34:57 -0500 + (Sun, 22 May 2011) | 15 lines Merged revisions 320444 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r320444 | tilghman | 2011-05-22 18:25:51 -0500 (Sun, 22 May 2011) + | 8 lines Don't crash when the connection fails. (closes issue + #19250) Reported by: seadweller Patches: + 20110514__issue19250.diff.txt uploaded by tilghman (license 14) + Tested by: seadweller, sum ........ ................ + +2011-05-20 21:40 +0000 [r320340] David Vossel + + * main/tcptls.c, /: Merged revisions 320338 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r320338 | dvossel | 2011-05-20 16:39:36 -0500 + (Fri, 20 May 2011) | 14 lines Merged revisions 320271 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r320271 | dvossel | 2011-05-20 16:24:48 -0500 (Fri, 20 May 2011) + | 8 lines Fixes issue with ast_tcptls_server_start failing on + second attempt to bind. (closes issue #19289) Reported by: + wdoekes Patches: + issue19289_delay_old_address_setting_tcptls.patch uploaded by + wdoekes (license 717) ........ ................ + +2011-05-20 20:53 +0000 [r320238] Richard Mudgett + + * /, apps/app_meetme.c: Merged revisions 320237 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r320237 | rmudgett | 2011-05-20 15:49:03 -0500 + (Fri, 20 May 2011) | 27 lines Merged revisions 320236 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r320236 | rmudgett | 2011-05-20 15:44:54 -0500 + (Fri, 20 May 2011) | 20 lines Merged revisions 320235 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r320235 | rmudgett | 2011-05-20 15:38:22 -0500 (Fri, 20 May 2011) + | 13 lines The meetme CLI command completion leaves conferences + mutex locked. When issuing a meetme kick CLI command and an + invalid (non-existent) conference number is specified, pressing + Tab leaves the conferences mutex locked and, therefore, all + conferences deadlock. Add missing unlock. (closes issue #19336) + Reported by: zvision Patches: app_meetme.diff uploaded by zvision + (license 798) ........ ................ ................ + +2011-05-20 18:49 +0000 [r320181] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 320180 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320180 | mnicholson | 2011-05-20 13:48:46 -0500 (Fri, 20 May + 2011) | 16 lines This commit modifies the way polling is done on + TLS sockets. Because of the buffering the TLS layer does, polling + is unreliable. If poll is called while there is data waiting to + be read in the TLS layer but not at the network layer, the + messaging processing engine will not proceed until something else + writes data to the socket, which may not occur. This change + modifies the logic around TLS sockets to only poll after a failed + read on a non-blocking socket. This way we know that there is no + data waiting to be read from the buffering layer. (closes issue + #19182) Reported by: st Patches: ssl-poll-fix3.diff uploaded by + mnicholson (license 96) Tested by: mnicholson ........ + +2011-05-20 18:29 +0000 [r320178] Jonathan Rose + + * /, apps/app_voicemail.c: Merged revisions 320162 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r320162 | jrose | 2011-05-20 13:12:21 -0500 (Fri, 20 May + 2011) | 15 lines Fixes an imapfolder related crash imapfolders + being set in the general section of voicemail would cause the + inbox folder name to change. Since sound file names are made + based on the names of the folders, this would cause the audio + related to that folder name to change and if Asterisk attempted + to play it, the channel would instantly hang up when the audio + file couldn't be found. This patch searches for the name of the + folder first to leave existing behavior in tact and if that + fails, it uses the normal inbox name to get the sound file + instead. (closes issue #16104) Reported by: blkline Review: + https://reviewboard.asterisk.org/r/1215/ ........ + +2011-05-20 17:04 +0000 [r320058-320060] Richard Mudgett + + * /, main/features.c: Merged revisions 320059 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320059 | rmudgett | 2011-05-20 12:03:49 -0500 (Fri, 20 May 2011) + | 1 line Misc comment cleanup in features.c. ........ + + * main/channel.c, /, main/features.c: Merged revisions 320057 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320057 | rmudgett | 2011-05-20 11:43:02 -0500 (Fri, 20 May 2011) + | 19 lines Crash while transferring a call during DTMF feature + timeout. When a call is being attended transferred during the + time between AST_FRAME_DTMF_BEGIN and AST_FRAME_DTMF_END, the + transferred channel becomes a zombie (so tech data is not + available), making ast_dtmf_stream() segfault when it tries to + send the DTMF digit (at least with SIP channels). Patch based on + feature-end-zombie.patch uploaded by Irontec (license 1256) * + Check for zombies when ast_channel_bridge() returns. * Guarantee + that the fo parameter value is initialized in + ast_channel_bridge() before any returns. (closes issue #19116) + Reported by: Irontec Tested by: rmudgett ........ + +2011-05-20 16:27 +0000 [r320040] Jonathan Rose + + * funcs/func_strings.c, CHANGES: Adds STRREPLACE function Adds a + new STRREPLACe function to func_strings.c that allows users to + search and replace against a variable in the dialplan. (closes + issue #18023) Reported by: wdoekes Review: + https://reviewboard.asterisk.org/r/1219/ + +2011-05-20 16:20 +0000 [r319998-320013] Richard Mudgett + + * /, apps/app_directed_pickup.c, main/features.c: Merged revisions + 320007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r320007 | rmudgett | 2011-05-20 11:19:01 -0500 (Fri, 20 May 2011) + | 2 lines Change some variable names to make pickup code easier + to understand. ........ + + * /, apps/app_directed_pickup.c, main/features.c: Merged revisions + 319997 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319997 | rmudgett | 2011-05-20 10:48:25 -0500 (Fri, 20 May 2011) + | 25 lines Crash when using directed pickup applications. The + directed pickup applications can cause a crash if the pickup was + successful because the dialplan keeps executing. This patch does + the following: * Completes the channel masquerade on a successful + pickup before the application returns. The channel is now + guaranteed a zombie and must not continue executing the dialplan. + * Changes the return value of the directed pickup applications to + return zero if the pickup failed and nonzero(-1) if the pickup + succeeded. * Made some code optimizations that no longer require + re-checking the pickup channel to see if it is still available to + pickup. (closes issue #19310) Reported by: remiq Patches: + issue19310_v1.8_v2.patch uploaded by rmudgett (license 664) + Tested by: alecdavis, remiq, rmudgett Review: + https://reviewboard.asterisk.org/r/1221/ ........ + +2011-05-20 13:42 +0000 [r319867-319939] Jonathan Rose + + * /, channels/chan_sip.c, configs/sip.conf.sample, + channels/sip/include/sip.h: Merged revisions 319938 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May + 2011) | 12 lines Adds legacy_useroption_parsing to address + interoperability concerns. With the new option engaged, Asterisk + should interpret user fields with useroptions contained within + the userfield of the uri by stripping them out of the original + message whenever a semicolon is encountered in the userfield + string. (closes issue #18344) Reported by: danimal Tested by: + jrose Review: https://reviewboard.asterisk.org/r/1223/ ........ + + * /, main/features.c: Merged revisions 319866 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319866 | jrose | 2011-05-19 13:32:38 -0500 (Thu, 19 May 2011) | + 11 lines Fix Randomize option on Park() The randomize option was + generally not working like it should have at all on Park(). This + patch restores intended functionality. (closes issue #18862) + Reported by: davidw Tested by: jrose Review: + https://reviewboard.asterisk.org/r/1222/ ........ + +2011-05-19 18:12 +0000 [r319813] Mark Murawki + + * cel/cel_odbc.c, /: Merged revisions 319812 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319812 | markm | 2011-05-19 13:59:01 -0400 (Thu, 19 May 2011) | + 9 lines In cel_odbc, an uninitialized RWLIST is attempted to be + locked. Added INIT and DESTROY for the RWLIST odbc_tables (closes + issue #19331) Reported by: kobaz Patches: odbc_cel.patch uploaded + by kobaz (license 834) ........ + +2011-05-19 16:52 +0000 [r319759] Richard Mudgett + + * /, main/ccss.c: Merged revisions 319758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319758 | rmudgett | 2011-05-19 11:50:48 -0500 (Thu, 19 May 2011) + | 21 lines CCSS generic agent with POTS and ISDN phones fail + caller busy call-back test. If the following is true after a CCSS + activation: * The generic agent is for an analog phone or ISDN + phone. (Caller party) * The called party becomes available. * The + caller party is not available. When the caller party becomes + available, the caller is not alerted to the called party being + available. The generic agent still thinks the caller is busy. * + Fixed the generic agent device state event subscription to look + for all device states that are considered available. * + Encapsulated the device state test for CCSS generic device + available in cc_generic_is_device_available(). Made the generic + agent and monitor use the new function instead of the manually + coded inline equivalent. JIRA AST-559 JIRA SWP-3462 ........ + +2011-05-18 23:18 +0000 [r319530-319661] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 319654 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r319654 | twilson | 2011-05-18 16:15:58 -0700 + (Wed, 18 May 2011) | 22 lines Merged revisions 319653 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r319653 | twilson | 2011-05-18 16:11:57 -0700 + (Wed, 18 May 2011) | 15 lines Merged revisions 319652 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r319652 | twilson | 2011-05-18 16:04:35 -0700 (Wed, 18 May 2011) + | 8 lines Make sure everyone gets an unhold when a transfer + succeeds Some phones, like the Snom phones, send a hold to the + transfer target after before sending the REFER. We need to make + sure that we unhold the parties that are being connected after + the masquerade. If Local channels with the /nm option are used + when dialing the parties, hold music would still be playing on + the transfer target, even after being connected with the + transferee. ........ ................ ................ + + * /, channels/chan_sip.c: Merged revisions 319552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319552 | twilson | 2011-05-18 13:22:36 -0700 (Wed, 18 May 2011) + | 11 lines Unbreak the storing of registrations for restart The + fix for issue 18882 broke retrieving non-realtime peers from the + ast_db on restart/reload. This patch tries to unbreak things + while leaving the intent of the original fix intact. (closes + issue #19318) Reported by: remiq Patches: diff.txt uploaded by + twilson (license 396) Tested by: lmadsen, remiq ........ + + * apps/app_dial.c, /: Merged revisions 319529 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r319529 | twilson | 2011-05-18 13:05:34 -0700 + (Wed, 18 May 2011) | 24 lines Merged revisions 319528 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r319528 | twilson | 2011-05-18 13:02:06 -0700 + (Wed, 18 May 2011) | 17 lines Merged revisions 319527 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r319527 | twilson | 2011-05-18 12:56:08 -0700 (Wed, 18 May 2011) + | 10 lines Fix app_dial ring groups Revert part of r315643. We + need to remove the datastore here as well. The code in bridging + code will catch anything that app_dial might miss. (closes issue + #19311) Reported by: mspuhler Patches: issue_19311_no_answer.diff + uploaded by elguero (license 37) ........ ................ + ................ + +2011-05-17 22:04 +0000 [r319471] Richard Mudgett + + * /, channels/misdn/isdn_lib.c: Merged revisions 319469 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r319469 | rmudgett | 2011-05-17 16:57:56 -0500 + (Tue, 17 May 2011) | 22 lines Merged revision 319468 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r319468 | rmudgett | 2011-05-17 16:49:31 -0500 (Tue, + 17 May 2011) | 15 lines The mISDN HDLC mode is prevented on + dialed channels. The use of mISDN HDLC mode is prevented if the + mISDN dial technology option 'h1' is used when config option + astdtmf=yes. There is a bug in channels/misdn/isdn_lib.c which + prevents the use of HDLC mode. Instead of setting the channel to + HDLC mode it is set to transparent(no dsp, no hdlc), although + hdlc is not "no hdlc". I.e the logging message is correct, but + the if condition is not. Make check the nodsp and hdlc flags. + JIRA ABE-2787 JIRA SWP-3437 .......... ................ + +2011-05-17 21:59 +0000 [r319470] Damien Wedhorn + + * channels/chan_skinny.c: Remove extraneous line variables. The + vars were either explicitly or implicitly not used. + +2011-05-17 20:13 +0000 [r319427] 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: Option needed for Q931_IE_TIME_DATE to be + optional in CONNECT message. The NEC SV8300 rejects the + Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how + much of the current time is put in Q931_IE_TIME_DATE. * Send + date/time ie never. * Send date/time ie date only. * Send + date/time ie date and hour. * Send date/time ie date, hour, and + minute. * Send date/time ie date, hour, minute, and second. * + Send date/time ie default: Libpri will send date and hhmm only + when in NT PTMP mode to support ISDN phones. (closes issue + #19221) Reported by: kenner JIRA SWP-3396 + +2011-05-17 12:54 +0000 [r319366-319368] Leif Madsen + + * /, apps/app_voicemail.c: Merged revisions 319367 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r319367 | lmadsen | 2011-05-17 07:53:50 -0500 (Tue, 17 + May 2011) | 10 lines Don't create [general] voicemail context + when using users.conf Prior to this patch, app_voicemail would + create a [general] context when parsing users.conf. (closes issue + #18891) Reported by: pdugas Patches: + app_voicemail-ignore-general.patch uploaded by pdugas (license + 1222) app_voicemail-ignore-general-style-guidelines.patch + uploaded by seanbright (license 71) Tested by: pdugas ........ + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 319365 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319365 | lmadsen | 2011-05-17 07:39:37 -0500 (Tue, 17 May 2011) + | 6 lines Make Debian init script lsb compliant (closes issue + #18896) Reported by: manwe Patches: debian_init_lsb.patch + uploaded by manwe (license 1223) ........ + +2011-05-16 21:39 +0000 [r319316] Damien Wedhorn + + * channels/chan_skinny.c: Fix up skinny hints. Probably haven't + been working for a couple of years. May still need some more + love, but they are now working, both as a hint device and + monitoring a hint. Changes centre around the long ago change to + remove the requirement for a device name in a skinny line, and + changes to the transmit_* functions. + +2011-05-16 21:08 +0000 [r319262] Jonathan Rose + + * main/dsp.c: Merged revisions 319261 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319261 | jrose | 2011-05-16 16:00:55 -0500 (Mon, 16 May 2011) | + 2 lines Makes busy detection in dsp.c always allow for at least + one frame (20ms) of error so that 200ms tone lengths don't get + ignored by single frame error lengths. ........ + +2011-05-16 20:41 +0000 [r319260] Richard Mudgett + + * /, main/ccss.c: Merged revisions 319259 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319259 | rmudgett | 2011-05-16 15:33:37 -0500 (Mon, 16 May 2011) + | 13 lines Deadlock between generic CCSS agent and native ISDN + CCSS. Deadlock can occur when the generic CCSS agent is deleting + duplicate CC offers and the native ISDN CC driver is processing + an incoming CC message. The cc_core_instances container lock + cannot be held when an agent or monitor callback is invoked + without the possibility of a deadlock. * Make + kill_duplicate_offers() remove the reference in cc_core_instances + outside of the container lock. JIRA AST-566 JIRA SWP-3469 + ........ + +2011-05-16 18:21 +0000 [r319212] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 319204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r319204 | twilson | 2011-05-16 13:17:43 -0500 + (Mon, 16 May 2011) | 11 lines Merged revisions 319202 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r319202 | twilson | 2011-05-16 11:00:21 -0700 (Mon, 16 May 2011) + | 4 lines Unlink a peer from peers_by_ip when expiring a + registration Review: https://reviewboard.asterisk.org/r/1218/ + ........ ................ + +2011-05-16 15:58 +0000 [r319146] David Vossel + + * /, channels/chan_sip.c: Merged revisions 319145 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r319145 | dvossel | 2011-05-16 10:57:26 -0500 + (Mon, 16 May 2011) | 9 lines Merged revisions 319144 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r319144 | dvossel | 2011-05-16 10:56:16 -0500 (Mon, 16 + May 2011) | 2 lines Fixes issue with peer ref-counting during + handle_request_subscribe. (closes issue #19293) Reported by: + irroot ........ ................ + +2011-05-16 15:54 +0000 [r319143] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 319142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319142 | mnicholson | 2011-05-16 10:53:26 -0500 (Mon, 16 May + 2011) | 8 lines Make sure tcptls_session exists before + dereferencing it. (closes issue #19192) Reported by: stknob + Patches: 10-tcptls-unreachable-peer-segfault.patch uploaded by + Chainsaw (license 723) Tested by: vois, Chainsaw ........ + +2011-05-16 14:56 +0000 [r319087] Gregory Nietsky + + * channels/chan_sip.c, res/res_fax.c, CHANGES, + channels/sip/include/sip.h: When a error in T.38 negotiation + happens or its rejected on a channel the state of the channel + reverts to unknown this should be rejected. this is important for + negotiating T.38 gateway see #13405 This patch adds a option + T38_REJECTED that behaves as T38_DISABLED except it reports state + rejected. Trivial Change to res_fax to honnor UNAVAILABLE and + REJECTED states. (closes issue #18889) Reported by: irroot Tested + by: irroot, darkbasic, mnicholson Review: + https://reviewboard.asterisk.org/r/1115 + +2011-05-16 14:38 +0000 [r319086] Paul Belanger + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + res/res_http_post.c: Merged revisions 319085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r319085 | pabelanger | 2011-05-16 10:35:21 -0400 (Mon, 16 May + 2011) | 10 lines Support gmime-2.4 (closes issue #18863) Reported + by: tzafrir Patches: gmime-2.4-18.diff uploaded by tzafrir + (license 46) Tested by: tzafrir Review: + https://reviewboard.asterisk.org/r/1213/ ........ + +2011-05-16 14:29 +0000 [r319084] David Vossel + + * /, formats/format_wav.c: Merged revisions 319083 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r319083 | dvossel | 2011-05-16 09:26:33 -0500 (Mon, 16 + May 2011) | 5 lines Fixes Big Endian build issue. (closes issue + #19298) Reported by: tzafrir ........ + +2011-05-15 23:17 +0000 [r319024] Damien Wedhorn + + * channels/chan_skinny.c: Add activatesub and dialandactivate sub. + When called, activatesub first cleans up the active sub and then + handles the sub passed. dialandactivatesub first sets sub->exten + and then calls activatesub. Revise handle_offhook to utilise the + callid sent to chan_skinny. Some other minor fixes especially + around d->hookstate (which still needs some more work). + +2011-05-13 18:10 +0000 [r318918-318922] Brett Bryant + + * main/channel.c, /: Merged revisions 318921 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318921 | bbryant | 2011-05-13 14:09:34 -0400 (Fri, 13 May 2011) + | 8 lines Fixes a segmentation fault in dynamic hints when a + channel technology isn't loaded for a hint. (closes issue #18495) + Reported by: bertrand Tested by: bertrand ........ + + * /, res/res_srtp.c: Merged revisions 318919 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318919 | bbryant | 2011-05-13 14:04:50 -0400 (Fri, 13 May 2011) + | 10 lines This patch fixes an issue with SRTP which makes + HOLD/UNHOLD impossible when too much time has passed between + sending audio. (closes issue #18206) Reported by: bernhardsi + Patches: res_srtp_unhold.patch uploaded by bernhards (license + 1138) Tested by: bernhards, notthematrix ........ + + * /, channels/chan_sip.c: Merged revisions 318917 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318917 | bbryant | 2011-05-13 13:56:04 -0400 (Fri, 13 May 2011) + | 11 lines This patch allows TCP peers into the ast_db where they + were previously restricted. (closes issue #18882) Reported by: + cmaj Patches: + patch-chan_sip-1.8.3-rc2-allow-tcp-peer-store-db-and-readonly-rt-backend.diff.txt + uploaded by cmaj (license 830) Tested by: cmaj ........ + +2011-05-13 16:30 +0000 [r318869] Richard Mudgett + + * /, main/features.c: Merged revisions 318868 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318868 | rmudgett | 2011-05-13 11:28:26 -0500 (Fri, 13 May 2011) + | 19 lines CDR's are being written immediately on caller hangup. + CDR's are being written immediately on caller hangup. The + dialplan is not able to modify it in the h exten. The h exten in + the initial context is not run before closing CDR's when the + bridge is unlinked if a macro is active and does not have an h + exten. * Make ast_bridge_call() check for an h exten in the + current context and if a macro is active then the initial + context. The first h exten found is then run before closing the + CDR. (closes issue #18212) Reported by: leearcher Patches: + issue18212_v1.8.patch uploaded by rmudgett (license 664) Tested + by: rmudgett Review: https://reviewboard.asterisk.org/r/1206/ + ........ + +2011-05-13 08:33 +0000 [r318833] Damien Wedhorn + + * channels/chan_skinny.c: Move exten used for dialing from device + to subchannel. There were some issues where if a simple switch + was cancelled and a new switch started before the first had timed + out where the d->exten would be used for both subchannels. This + was bad leading to possible invalid extensions if some digits had + been entered in the abandoned simple switch and the second one + was completed before the first timed out, or the second would be + cancelled because d->exten would be set to nothing on the time + out of the first. + +2011-05-13 01:55 +0000 [r318785] Matthew Nicholson + + * /, channels/sip/reqresp_parser.c: Merged revisions 318720 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318720 | mnicholson | 2011-05-12 18:35:51 -0500 (Thu, 12 May + 2011) | 4 lines Handle ipv6 addresses in the sent-by Via: field. + This change fixes a regression in via header parsing and ipv6 + handling. (closes issue #18951) ........ + +2011-05-13 01:50 +0000 [r318784] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 318783 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318783 | rmudgett | 2011-05-12 20:47:05 -0500 (Thu, 12 May 2011) + | 14 lines PRI early media won't ring. And another way to pass + early media. Don't indicate that there is inband information + present, just assume that the B channel is connected. * Restore + clearing the dialing flag Rx squelch unconditionally when a + PROCEEDING message comes in. (closes issue #19268) Reported by: + tbsky Patches: issue19268_v1.8.patch uploaded by rmudgett + (license 664) Tested by: tbsky ........ + +2011-05-12 22:56 +0000 [r318672] Alec L Davis + + * /, channels/chan_sip.c, apps/app_directed_pickup.c, + main/features.c, include/asterisk/features.h: Merged revisions + 318671 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318671 | alecdavis | 2011-05-13 10:52:08 +1200 (Fri, 13 May + 2011) | 30 lines Fix directed group pickup feature code *8 with + pickupsounds enabled Since 1.6.2, the new pickupsound and + pickupfailsound in features.conf cause many issues. 1). + chan_sip:handle_request_invite() shouldn't be playing out the + fail/success audio, as it has 'netlock' locked. 2). dialplan + applications for directed_pickups shouldn't beep. 3). feature + code for directed pickup should beep on success/failure if + configured. Created a sip_pickup() thread to handle the pickup + and playout the audio, spawned from handle_request_invite. Moved + app_directed:pickup_do() to features:ast_do_pickup(). Functions + below, all now use the new ast_do_pickup() app_directed_pickup.c: + pickup_by_channel() pickup_by_exten() pickup_by_mark() + pickup_by_part() features.c: ast_pickup_call() (closes issue + #18654) Reported by: Docent Patches: + ast_do_pickup_1.8_trunk.diff.txt uploaded by alecdavis (license + 585) Tested by: lmadsen, francesco_r, amilcar, isis242, + alecdavis, irroot, rymkus, loloski, rmudgett Review: + https://reviewboard.asterisk.org/r/1185/ ........ + +2011-05-12 20:44 +0000 [r318600-318635] Damien Wedhorn + + * channels/chan_skinny.c: Consolidate setsubstate_* into + setsubstate and use a switch. Consolidate the functions and add + some debugging info. Allows to be able to set a substate without + explicitly knowing what the state is. + + * channels/chan_skinny.c: Add setsubstate_onhook. Add the + setsubstate_onhook to complete the initial substate handling + procedures. Added dumpsub(sub, forcehangup) which is the common + way of calling setsubstate_onhook. Dumpsub attempts to activate + another sub after setting the current one onhook. + +2011-05-11 18:52 +0000 [r318551-318552] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 318550 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318550 | twilson | 2011-05-11 13:47:33 -0500 (Wed, 11 May 2011) + | 2 lines Comment out the REF_DEBUG that slipped in during + debugging ........ + + * /, channels/chan_sip.c: Merged revisions 318549 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r318549 | twilson | 2011-05-11 13:39:48 -0500 + (Wed, 11 May 2011) | 27 lines Merged revisions 318548 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r318548 | twilson | 2011-05-11 12:15:39 -0500 (Wed, 11 May 2011) + | 19 lines Clean up several chan_sip reference leaks Several + situations in the code could lead to peers or sip_pvt references + being leaked. This would cause RTP ports to never be destroyed + (leading to exhaustion of all available RTP ports) and memory + leaks. The original patch for this issue from rgagnon was the + result of an obscene amount of testing and hard work, for which I + am very grateful. I did some cleanup and added a few additional + refcount fixes that I found. (closes issue #17255) Reported by: + kvveltho Patches: tag-1.6.2.17-r309252-sip-dos-mem-leak-fix.diff + uploaded by rgagnon (license 1202) Tested by: rgagnon, twilson, + wdoekes, loloski Review: https://reviewboard.asterisk.org/r/1101/ + Review: https://reviewboard.asterisk.org/r/1207/ Review: + https://reviewboard.asterisk.org/r/1210/ ........ + ................ + +2011-05-10 23:42 +0000 [r318500] Richard Mudgett + + * /, channels/sig_pri.c, channels/sig_ss7.c: Merged revisions + 318499 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) + | 15 lines Unable to pickup DAHDI/PRI call because call state is + reported as DIALING. The channel state is not updated to RINGING + when an ALERTING message is received. Regression caused when + sig_pri.c (also sig_ss7.c) extracted from chan_dahdi.c. * Added + missing channel state update to RINGING when the + AST_CONTROL_RINGING frame is queued for ISDN and SS7. (closes + issue #19257) Reported by: alecdavis Patches: + issue19257_v1.8_v2.patch uploaded by rmudgett (license 664) + Tested by: alecdavis, rmudgett ........ + +2011-05-10 15:16 +0000 [r318437] Russell Bryant + + * /, channels/chan_iax2.c: Merged revisions 318436 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r318436 | russell | 2011-05-10 10:13:16 -0500 (Tue, 10 + May 2011) | 2 lines chan_iax2: change LOG_NOTICE to LOG_DEBUG in + iax2_read(). ........ + +2011-05-10 00:22 +0000 [r318400] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 318337 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r318337 | twilson | 2011-05-09 15:23:15 -0500 + (Mon, 09 May 2011) | 18 lines Merged revisions 318331 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r318331 | twilson | 2011-05-09 15:04:41 -0500 (Mon, 09 May 2011) + | 12 lines Don't offer video to directmedia callee unless caller + offered it as well Make sure that when directmedia is enabled, + that video is not offered to the callee even if it supports it. + p->vrtp will not exist since the caller didn't offer video. + (closes issue #19195) Reported by: one47 Patches: + sip_cant_add_video_rtp uploaded by one47 (license 23) ........ + ................ + +2011-05-09 23:16 +0000 [r318283-318352] Richard Mudgett + + * /, res/Makefile, res/res_features.exports.in (removed): Merged + revisions 318351 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318351 | rmudgett | 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) + | 6 lines Remove references to res_features and its export file. + The contents of res/res_features.c was moved to into + main/features.c awhile ago. There is no longer any need for the + res/Makefile to reference res_features or the res_features linker + exports file to exist. ........ + + * /, main/features.c: Merged revisions 318282 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318282 | rmudgett | 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011) + | 24 lines Hangup extension executed twice. When a user hangs up + a call, in certain circumstances, the hangup extension can end up + being executed twice: 1) If a call is bridged and the 'h' + extension executes the Hangup application, then the 'h' extension + will be executed twice. 2) If a call is bridged within a macro + (Dial or Queue), it has its own 'h' extension, the main context + also has an 'h' extension, and the macro 'h' extension executes + the Hangup application, then both 'h' extensions will be + executed. * Revert originally commited fix for #16106 and just + set AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in + ast_bridge_call(). The bridge code just executed an 'h' extension + so the main PBX loop does not need to execute one as well. (issue + #16106) Reported by: ajohnson (issue #16548) Reported by: hajekd + ........ + +2011-05-09 17:13 +0000 [r318234] David Vossel + + * /, channels/chan_sip.c: Merged revisions 318233 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r318233 | dvossel | 2011-05-09 12:09:55 -0500 + (Mon, 09 May 2011) | 14 lines Merged revisions 318230 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r318230 | dvossel | 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011) + | 7 lines Fixes cases where sip_set_rtp_peer can return too early + during media path reset. (closes issue #19225) Reported by: one47 + Patches: sip_set_rtp_peer.patch uploaded by one47 (license 23) + ........ ................ + +2011-05-09 17:00 +0000 [r318232] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 318231 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r318231 | rmudgett | 2011-05-09 11:57:18 -0500 + (Mon, 09 May 2011) | 41 lines Don't get early media for ISDN on + outgoing calls. It looks to be a long-standing misinterpretation + of the progress indicator ie values: 1 - Call is not end-to-end + ISDN; further call progress information may be available in-band. + 8 - In-band information or an appropriate pattern is now + available. Only value 8 is handled by chan_dahdi/sig_pri. The 1 + value is not handled as early media probably because the meaning + of the second half of it's description was overlooked. * Test to + see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or + PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path. + (closes issue #18868) Reported by: isrl Patches: + issue18868_19246_v1.8.patch uploaded by rmudgett (license 664) + Tested by: satish_lx .......... No inband progress on + PRI_EVENT_RINGING even if inband flag set. My ISDN-PRI provider + sends an ALERTING with "Inband information or appropriate pattern + now available", but Asterisk only generates and passes the RING + to the SIP extension, not the inband message. Unfortunately, the + inband message is not a ringback tone but a prompt that says the + number is not in service. The SIP extension then hears two rings + and the call is hungup which confuses the caller. * Post an + AST_CONTROL_PROGRESS as well as opening the media path if inband + audio is indicated with an ALERTING message. (closes issue + #19246) Reported by: cristiandimache Patches: + issue19246_v1.8.patch uploaded by rmudgett (license 664) Tested + by: cristiandimache ................ + +2011-05-09 14:41 +0000 [r318194] Leif Madsen + + * main/app.c: Increase prepend filename length. (closes issue + #19238) Reported by: byronclark Patches: + increase_prepend_filename_length.patch uploaded by byronclark + (license 1200) + +2011-05-09 14:37 +0000 [r318162-318193] Jonathan Rose + + * main/features.c: Minor change to 318141 to improve parsing + behavior. + + * /, configs/features.conf.sample: Merged revisions 318148 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) | + 4 lines Documenting an observed behavior of features in + features.conf. Since parkinglots use an integer for the + parkinglot extensions, leading zeros specified in the + configuration file are ignored. ........ + +2011-05-09 14:11 +0000 [r318143] Matthew Nicholson + + * main/channel.c, /: Merged revisions 318142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r318142 | mnicholson | 2011-05-09 09:09:38 -0500 (Mon, 09 May + 2011) | 9 lines Make indicate/control frames WRITE events on + framehooks. Also, if a framehook returns a non-control frame, + don't forward it to the channel. (closes issue #19251) Reported + by: irroot Patches: (modified) framehook_indicate.patch2 uploaded + by irroot (license 52) Tested by: irroot ........ + +2011-05-09 13:56 +0000 [r318141] Jonathan Rose + + * main/features.c, CHANGES: Allows ParkedCall application to + specify a parkinglot. When invoking the app parkedcall, the + argument can now include '@parkinglot' after the extension. + (closes issue #18777) Reported by: cartama Patches: 0018777.diff + uploaded by cartama (license 1157) Review: + https://reviewboard.asterisk.org/r/1209/ + +2011-05-09 07:40 +0000 [r318106] Damien Wedhorn + + * channels/chan_skinny.c: Add setsubstate_callwait. If a call is + made to a line that already has a call and the device is offhook + (ie activeish call), the call is set to CALLWAIT rather than + RINGIN. + +2011-05-07 23:36 +0000 [r318056-318058] Russell Bryant + + * res/res_config_curl.c, /: Merged revisions 318057 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r318057 | russell | 2011-05-07 18:35:37 -0500 (Sat, 07 + May 2011) | 8 lines res_config_curl: fix a crash with static + realtime. (closes issue #18413) Reported by: jmls Patches: + 20101202__issue18413.diff.txt uploaded by tilghman (license 14) + Tested by: jmls ........ + + * /, channels/chan_iax2.c: Merged revisions 318055 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r318055 | russell | 2011-05-07 18:24:18 -0500 (Sat, 07 + May 2011) | 7 lines chan_iax2: Don't overwrite port found with an + SRV lookup. (closes issue #17291) Reported by: jcovert Patches: + chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by jcovert + (license 551) ........ + +2011-05-06 23:07 +0000 [r317996-318019] Damien Wedhorn + + * channels/chan_skinny.c: Only allow voicemail if substate is + OFFHOOK or no channel active (UNSET). (closes issue #17901) + Reported by: salecha + + * channels/chan_skinny.c: Rename sub->parent to sub->line. Improve + readability of code, eg, (sub->parent == d->activeline) becomes + (sub->line == d->activeline). + + * channels/chan_skinny.c: Move the hookstate from line to device. + Long time coming, finally moving the hookstate from line to + device. This may fix some issues where a device has multiple + lines. Previously we had to run through all lines on a device to + see if it was actually onhook or not. + +2011-05-06 21:49 +0000 [r317968-317970] Russell Bryant + + * /, apps/app_meetme.c: Merged revisions 317969 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) + | 10 lines Use the right variable to print the time in a debug + message. The original patch also increased some buffer sizes, but + that was already done in this version. (closes issue #17034) + Reported by: sysreq Patches: asterisk-issue-17034.patch uploaded + by sysreq (license 1009) ........ + + * /, apps/app_meetme.c: Merged revisions 317967 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317967 | russell | 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) + | 2 lines Fix some more "set but unused" compiler warnings. + ........ + +2011-05-06 21:10 +0000 [r317920] David Vossel + + * res/res_rtp_asterisk.c, /: Merged revisions 317918 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r317918 | dvossel | 2011-05-06 16:06:55 -0500 (Fri, 06 + May 2011) | 7 lines Fixes missing colon from To/From headers in + RTCP manager events. (closes issue #18221) Reported by: + clegall_proformatique Patches: 18221_1.patch uploaded by ebroad + (license 878) ........ + +2011-05-06 21:07 +0000 [r317843-317919] Russell Bryant + + * main/pbx.c, /: Merged revisions 317917 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317917 | russell | 2011-05-06 16:06:33 -0500 (Fri, 06 May 2011) + | 7 lines Fix calculation of free RAM to make minmemfree option + work. (closes issue #17124) Reported by: loic Patches: pbx_c.diff + uploaded by loic (license 1020) ........ + + * contrib/scripts/import-cdr-csv-mysql.pl (added): Add a cdr_csv to + MySQL import script to contrib/scripts. (closes issue #17036) + Reported by: precisenetworks Patches: import-cdr-csv-mysql.pl + uploaded by precisenetworks (license 1010) + + * apps/app_userevent.c, CHANGES: Add the Uniqueid header to + Userevent. (closes issue #16962) Reported by: jlpedrosa Patches: + patch.diff uploaded by jlpedrosa (license 1002) + + * /, channels/chan_sip.c: Merged revisions 317867 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317867 | russell | 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) + | 10 lines chan_sip: Destroy variables on a sip_pvt before + copying vars from the sip_peer. Don't duplicate variables on the + sip_pvt. Just reset the variable list each time. (closes issue + #19202) Reported by: wdoekes Patches: + issue19202_destroy_challenged_invite_chanvars.patch uploaded by + wdoekes (license 717) ........ + + * /, channels/chan_sip.c: Merged revisions 317865 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317865 | russell | 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) + | 11 lines chan_sip: fix a deadlock in check_rtp_timeout. Don't + block doing silly deadlock avoidance. Just return and try again + later. The funciton gets called often enough that it's fine. + Also, this change was already made in trunk. (closes issue + #18791) Reported by: irroot Patches: chan_sip.rtptimeout.patch + uploaded by irroot (license 52) ........ + + * addons/app_mysql.c, /: Merged revisions 317837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317837 | russell | 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011) + | 11 lines Fix a crash in the MySQL() application. This code was + not handling channel datastores safely. The channel must be + locked. (closes issue #17964) Reported by: wuwu Patches: + issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license + 71) Tested by: wuwu ........ + +2011-05-06 19:23 +0000 [r317818-317833] Matthew Nicholson + + * CHANGES: Updated CHANGES to note the autoservice changes for + pbx_lua + + * configs/extensions.lua.sample: Updated the sample pbx_lua config + file to reflect autoservice changes. + +2011-05-06 19:15 +0000 [r317807] Russell Bryant + + * /, contrib/realtime/mysql/sipfriends.sql: Merged revisions 317805 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317805 | russell | 2011-05-06 14:14:39 -0500 (Fri, 06 May 2011) + | 7 lines Add a new sipfriends.sql for MySQL that has more fields + in it. (closes issue #16399) Reported by: pabelanger Patches: + sipfriends.sql.v3 uploaded by pabelanger (license 224) ........ + +2011-05-06 19:14 +0000 [r317721-317806] Matthew Nicholson + + * pbx/pbx_lua.c, UPGRADE.txt: Default to starting an autoservice in + pbx_lua. The autoservice is automatically stopped when + applications are executed, so this shouldn't cause any problems. + + * pbx/pbx_lua.c, UPGRADE.txt: Make pbx_lua handle managing the + autoservice better. Make autoservice_start() and + autoservice_stop() return nothing. Also check if the autoservice + flag is set before starting or stopping the autoservice and stop + and start the autoservice when returning control to and getting + control from the pbx engine. + + * UPGRADE.txt: Added note about changes in pbx_lua's behavior when + applications do dialplan jumps + + * CHANGES: Use two spaces after periods for the recent pbx_lua + change descriptions + + * CHANGES: Updated CHANGES for hints support in pbx_lua + + * pbx/pbx_lua.c, CHANGES: Detect Goto in pbx_lua. This code will + actually detect any dialplan jump from any application that calls + ast_explicit_goto(). This change is only being done in trunk as + it may change the way some dialplans execute. + +2011-05-06 16:23 +0000 [r317671] Richard Mudgett + + * /, channels/chan_sip.c: Merged revisions 317670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) + | 22 lines Fix SIP connected line updates. This patch fixes a + couple SIP connected line update problems: 1) The connected line + needs to be updated when the initial INVITE is sent if there is a + peer callerid configured. Previously, the connected line + information did not get reported until the call was connected so + SIP could not report connected line information in ringing or + progress messages. 2) The connected line should not be updated on + initial connect if there is no connected line information. + Previously, all it did was wipe out any default preset + CONNECTEDLINE information set by the dialplan with empty strings. + (closes issue #18367) Reported by: GeorgeKonopacki Patches: + issue18367_v1.8.patch uploaded by rmudgett (license 664) Tested + by: rmudgett Review: https://reviewboard.asterisk.org/r/1199/ + ........ + +2011-05-06 08:21 +0000 [r317596] Terry Wilson + + * /, apps/app_queue.c: Merged revisions 317584 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r317584 | twilson | 2011-05-06 01:18:53 -0700 + (Fri, 06 May 2011) | 20 lines Merged revisions 317575 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r317575 | twilson | 2011-05-06 01:04:17 -0700 + (Fri, 06 May 2011) | 13 lines Merged revisions 317574 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) + | 6 lines Re-fix queue round-robin This part of the change for + r315596 was incorrect. No bridge occurs when doing a roundrobin + dial and no one answers, so this code shouldn't have been + removed. ........ ................ ................ + +2011-05-05 23:47 +0000 [r317426-317531] Russell Bryant + + * Makefile, /: Merged revisions 317530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317530 | russell | 2011-05-05 18:46:54 -0500 (Thu, 05 May 2011) + | 10 lines If the configure script runs, force a rebuild of + menuselect-tree. Some contents in the menuselect tree are + dependent on configure script parameters, namely + --enable-dev-mode. (closes issue #17219) Reported by: Nick_Lewis + Patches: issue_17219.rev1.txt uploaded by russell (license 2) + ........ + + * /, contrib/realtime/mysql/queue_log.sql, + contrib/realtime/mysql/sipfriends.sql: Merged revisions 317486 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317486 | russell | 2011-05-05 18:15:53 -0500 (Thu, 05 May 2011) + | 9 lines Fix some more realtime MySQL schema issues. (closes + issue #18537) Reported by: denzs Patches: sipfriends.sql.svndiff + uploaded by denzs (license 1182) queue_log.sql.svndiff uploaded + by denzs (license 1182) meetme.sql.svndiff uploaded by denzs + (license 1182) ........ + + * /, contrib/realtime/mysql/meetme.sql, + contrib/realtime/mysql/sipfriends.sql: Merged revisions 317484 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317484 | russell | 2011-05-05 18:12:35 -0500 (Thu, 05 May 2011) + | 8 lines Fix some errors in sample MySQL realtime schema files. + (closes issue #18915) Reported by: Dovid Patches: + sipfriends.patch uploaded by Dovid (license 652) meetme.patch + uploaded by Dovid (license 652) ........ + + * CHANGES, res/res_calendar.c: Add "calendar show types" CLI + command. (closes issue #18246) Reported by: junky Patches: + calendar_types.diff uploaded by junky (license 177) + + * cel/cel_pgsql.c, UPGRADE.txt, configs/cel_pgsql.conf.sample, + CHANGES: Add CEL extra field to cel_pgsql. (closes issue #18462) + Reported by: joscas Patches: bug_18462.diff uploaded by snuffy + (license 35) cel_pgsql.conf.sample.issue18462.patch uploaded by + joscas (license 1180) + + * /, cdr/cdr_syslog.c: Merged revisions 317480 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317480 | russell | 2011-05-05 18:00:55 -0500 (Thu, 05 May 2011) + | 8 lines Don't lose cdr_syslog config on a reload. (closes issue + #18679) Reported by: enegaard Patches: + issue18679_seanbright.patch uploaded by seanbright (license 71) + Tested by: enegaard ........ + + * channels/chan_unistim.c, channels/chan_usbradio.c, + channels/chan_dahdi.c, /, channels/chan_sip.c, + channels/chan_skinny.c, channels/chan_h323.c, + channels/chan_alsa.c, channels/chan_console.c, + channels/chan_oss.c, channels/chan_mgcp.c, + channels/misdn_config.c: Merged revisions 317478 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 + May 2011) | 12 lines Fix some consistency issues with + jitterbuffer config. Store the defaults noted in the sample + config files in the jitterbuffer config data structure. This + makes the CLI commands that output these settings show the right + thing. Also only show the settings that are relevant in the + settings CLI commands, based on which jitterbuffer is selected + and whether it's enabled. (closes issue #19083) Reported by: + rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by + rgagnon (license 1202) ........ + + * /, pbx/pbx_lua.c: Merged revisions 317476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317476 | russell | 2011-05-05 17:47:57 -0500 (Thu, 05 May 2011) + | 8 lines Add a datastore fixup to fix a pbx_lua crash. (closes + issue #19055) Reported by: jamhed Patches: + lua_datastore_fixup1.diff uploaded by mnicholson (license 96) + Tested by: mnicholson, jamhed ........ + + * cel/cel_pgsql.c, channels/chan_jingle.c, + channels/sip/sdp_crypto.c, res/res_config_odbc.c, /, + channels/chan_sip.c, res/res_crypto.c, pbx/pbx_lua.c, + channels/iax2-provision.c, pbx/pbx_dundi.c, + channels/chan_console.c, cdr/cdr_radius.c, channels/chan_iax2.c, + res/res_jabber.c, res/res_config_sqlite.c: Merged revisions + 317474 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317474 | russell | 2011-05-05 17:36:33 -0500 (Thu, 05 May 2011) + | 2 lines Fix more "set but unused" warnings. ........ + + * /, main/dsp.c: Merged revisions 317429 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317429 | russell | 2011-05-05 17:11:19 -0500 (Thu, 05 May 2011) + | 5 lines Only display inband DTMF warning if inband DTMF + detection is enabled. (closes issue #18901) Reported by: irroot + ........ + + * /, apps/app_rpt.c: Merged revisions 317427 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317427 | russell | 2011-05-05 16:58:45 -0500 (Thu, 05 May 2011) + | 7 lines Fix potential memory leak, and use of uninitialized + memory. (closes issue #16476) Reported by: junky Patches: + M16476.diff uploaded by junky (license 177) ........ + + * main/manager.c, /: Merged revisions 317425 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317425 | russell | 2011-05-05 16:53:13 -0500 (Thu, 05 May 2011) + | 7 lines Add missing ActioID handling to Events action. (closes + issue #18949) Reported by: edersohe Patches: 0018949.patch + uploaded by edersohe (license 1228) ........ + +2011-05-05 21:20 +0000 [r317395] Sean Bright + + * main/asterisk.c: Add some new editline bindings by default, and + allow for user specified configuration. I excluded the part of + this patch that used the HOME environment variable since the + built-in editline library goes to great lengths to disallow that. + Instead only settings the EDITRC environment variable will use a + user specified file. Also, the default environment variable use + to determine the edit more is AST_EDITMODE instead of AST_EDITOR + (although the latter is still supported). (closes issue #15929) + Reported by: kkm Patches: astcli-editrc-v2.diff uploaded by kkm + (license 888) 015929-astcli-editrc-trunk.240324.diff uploaded by + kkm (license 888) Tested by: seanbright + +2011-05-05 20:46 +0000 [r317382] Damien Wedhorn + + * channels/chan_skinny.c: Move hold stuff to the setsubstate + arrangement. skinny_hold moved to setsubstate_hold and + skinny_unhold integrated into setsubstate_connected. Removed + sub->onhold and replaced with SUBSTATE_HOLD. Also fixed inbound + call answering by queueing an AST_CONTROL_ANSWER on answering a + SUBSTATE_RINGIN sub (was a typo). + +2011-05-05 20:27 +0000 [r317377] Sean Bright + + * /, addons/res_config_mysql.c: Merged revisions 317370 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317370 | seanbright | 2011-05-05 16:25:52 -0400 (Thu, 05 May + 2011) | 10 lines Don't duplicate our data on the stack and just + use the MYSQL_ROW directly. With large result sets we were + blowing out the stack. (closes issue #19090) Reported by: + mickecarlsson Patches: issue19090_trunk_svn.patch uploaded by + seanbright (license 71) Tested by: mickecarlsson ........ + +2011-05-05 19:56 +0000 [r317337] Russell Bryant + + * /, apps/app_queue.c: Merged revisions 317336 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317336 | russell | 2011-05-05 14:55:58 -0500 (Thu, 05 May 2011) + | 7 lines Increase buffer size to be PATH_MAX for a path. (closes + issue #19239) Reported by: byronclark Patches: + queue_announce_length.patch uploaded by byronclark (license 1200) + ........ + +2011-05-05 19:33 +0000 [r317334] Jonathan Rose + + * /, channels/chan_sip.c: Merged revisions 317283 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317283 | jrose | 2011-05-05 14:09:13 -0500 (Thu, 05 May 2011) | + 10 lines Resolves a deadlock that occurs during sip_new This is + based on an uncommitted patch by jpeeler for the issue. Instead + of relocking and then unlocking the channel though, we keep the + lock on the channel until we are finished doing what we need to + the channel. (closes issue #18441) Reported by: Alric ........ + +2011-05-05 18:46 +0000 [r317282] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 317281 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r317281 | russell | 2011-05-05 13:39:44 -0500 + (Thu, 05 May 2011) | 29 lines Merged revisions 317255 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r317255 | russell | 2011-05-05 13:29:53 -0500 + (Thu, 05 May 2011) | 22 lines Merged revisions 317211 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r317211 | russell | 2011-05-05 13:20:29 -0500 (Thu, 05 May 2011) + | 15 lines chan_sip: fix broken realtime peer count, fix memory + leak This patch addresses two bugs in chan_sip: 1) The count of + realtime peers and users was off. The increment checked the value + of the caching option, while the decrement did not. 2) Add a + missing regfree() for a regex. (closes issue #19108) Reported by: + vrban Patches: missing_regfree.patch uploaded by vrban (license + 756) sip_object_counter.patch uploaded by vrban (license 756) + ........ ................ ................ + +2011-05-05 18:09 +0000 [r317198] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 317196 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317196 | mnicholson | 2011-05-05 13:02:52 -0500 (Thu, 05 May + 2011) | 8 lines Set SO_KEEPALIVE on SIP TCP sockets so that they + eventually go away when a peer abruptly disappears. This mostly + occurs after a successful registration. (closes issue #17544) + Reported by: marcelloceschia Patches: (modified) tcptls.patch + uploaded by st (license 907) ........ + +2011-05-05 18:08 +0000 [r317197] David Vossel + + * bridges/bridge_softmix.c, funcs/func_jitterbuffer.c: Fixes + reliability issues with func_jitterbuffer's usage in the new + ConfBridge application. + +2011-05-05 15:06 +0000 [r317059-317105] Leif Madsen + + * /, contrib/scripts/safe_asterisk: Merged revisions 317104 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r317104 | lmadsen | 2011-05-05 11:04:24 -0400 + (Thu, 05 May 2011) | 15 lines Merged revisions 317102 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r317102 | lmadsen | 2011-05-05 10:54:46 -0400 (Thu, 05 May 2011) + | 8 lines Disable console colourization inside safe_asterisk + checks. (closes issue #19213) Reported by: lefoyer Patches: + issue19213_strip_color_in_safe_asterisk-svn.patch uploaded by + wdoekes (license 717) Tested by: wdoekes, lefoyer ........ + ................ + + * Makefile, configs/cel.conf.sample, /: Merged revisions 317058 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r317058 | lmadsen | 2011-05-05 08:27:56 -0400 (Thu, 05 May 2011) + | 7 lines Remove unused directory and clear up some + documentation. (closes issue #19193) Reported by: bchia Patches: + cel-csv.diff uploaded by lathama (license 1028) Tested by: + lathama, Marquis42 ........ + +2011-05-05 09:03 +0000 [r316994-317026] Damien Wedhorn + + * channels/chan_skinny.c: Add setsubstate_congestion and + setsubstate_progress. Move handling of both state handling from + skinny_indicate to it's own sub. Also, modified behaviour to not + hangup the sub and let the dialplan have a chance in doing what + it wants for congestion. Added various states to substate2str and + added these states where applicable for other set_substate_ + procs. + + * channels/chan_skinny.c: Add setsubstate_busy. Move handling of + setting busy state from skinny_indicate to it's own sub. Also, + modified behaviour to not hangup the sub and let the dialplan + have a chance in doing what it wants (eg busy(10); hangup() in + the dialplan now gives a busy indication for 10 secs and then + hangs up. + +2011-05-05 07:09 +0000 [r316962] Stefan Schmidt + + * main/astobj2.c: Adding the Move to Front Hash functionality + Moving a found object to the front of its bucket to reduce the + necessary traversal steps to find an object. This change improves + the search time on large system with many data or in link lists. + (closes issue #19233) Reported by: schmidts Review: + https://reviewboard.asterisk.org/r/1201/ + +2011-05-05 02:34 +0000 [r316920] Sean Bright + + * main/manager.c, /, main/http.c, main/utils.c: Merged revisions + 316917-316919 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316917 | seanbright | 2011-05-04 22:23:28 -0400 (Wed, 04 May + 2011) | 5 lines Make sure that tcptls_session is properly + initialized. (issue #18598) Reported by: ksn ........ r316918 | + seanbright | 2011-05-04 22:25:20 -0400 (Wed, 04 May 2011) | 5 + lines Look at the correct buffer for our digest info instead of + an empty one. (issue #18598) Reported by: ksn ........ r316919 | + seanbright | 2011-05-04 22:30:45 -0400 (Wed, 04 May 2011) | 10 + lines Use the correct HTTP method when generating our digest, + otherwise we always fail. When calculating the 'A2' portion of + our digest for verification, we need the HTTP method that is + currently in use. Unfortunately our mapping function was + incorrect, resulting in invalid hashes being generated and, in + turn, failures in authentication. (closes issue #18598) Reported + by: ksn ........ + +2011-05-04 21:44 +0000 [r316885] Damien Wedhorn + + * channels/chan_skinny.c: Add setsubstate_ringout (equivalent to + AST_STATE ringing). Renamed previous setsubstate_ringout to + setsubstate_dialing for a state when attempting to dial a number, + substate ringout now for when core has indicated that the channel + is actually ringing on the other end. Also added substate2str for + debugging purposes. + +2011-05-04 18:57 +0000 [r316832] Richard Mudgett + + * /, apps/app_meetme.c: Merged revisions 316831 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316831 | rmudgett | 2011-05-04 13:51:40 -0500 (Wed, 04 May 2011) + | 9 lines Wait for leader with Music On Hold allows crosstalk + between participants. Parenthesis in the wrong position. + Regression from issue #14365 when expanding conference flags to + use 64 bits. (closes issue #18418) Reported by: MrHanMan Tested + by: rmudgett ........ + +2011-05-04 16:42 +0000 [r316798] David Vossel + + * channels/chan_sip.c, CHANGES: Reverts rev 316218 as it breaks + parsing the [general] section of sip.conf. The functionality this + patch attempts to achieve should already be possible using + [general](+) in the config file. issue #17957 + +2011-05-04 16:17 +0000 [r316664-316711] Sean Bright + + * /, apps/app_voicemail.c: Merged revisions 316709 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316709 | seanbright | 2011-05-04 12:15:32 -0400 + (Wed, 04 May 2011) | 22 lines Merged revisions 316708 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r316708 | seanbright | 2011-05-04 12:10:59 -0400 + (Wed, 04 May 2011) | 15 lines Merged revisions 316707 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r316707 | seanbright | 2011-05-04 12:08:50 -0400 (Wed, 04 May + 2011) | 8 lines If sox fails when processing a voicemail, don't + delete the original file. (closes issue #18111) Reported by: + sysreq Patches: issue18111_trunk.patch uploaded by seanbright + (license 71) Tested by: seanbright ........ ................ + ................ + + * main/manager.c, /: Merged revisions 316663 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316663 | seanbright | 2011-05-04 10:35:05 -0400 (Wed, 04 May + 2011) | 8 lines Only return a single error via AMI when + requesting a forbidden action. (closes issue #19216) Reported by: + oej Patches: issue19216-1.8-r316204.patch uploaded by seanbright + (license 71) Tested by: seanbright ........ + +2011-05-04 14:26 +0000 [r316618-316657] David Vossel + + * /, apps/app_chanspy.c: Merged revisions 316650 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316650 | dvossel | 2011-05-04 09:25:03 -0500 + (Wed, 04 May 2011) | 15 lines Merged revisions 316644 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r316644 | dvossel | 2011-05-04 09:23:39 -0500 (Wed, 04 May 2011) + | 9 lines Fixes one-way-audio when chanspy activated with the 'o' + option (closes issue #18382) Reported by: jkister Patches: + 0001-Bugfix-18382-one-way-audio-when-chanspy-activated.patch.txt + uploaded by malin (license ) Tested by: firstsip, Greenlightcrm, + malin, wdoekes, boroda, dvossel ........ ................ + + * /, channels/chan_sip.c: Merged revisions 316617 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316617 | dvossel | 2011-05-04 08:44:41 -0500 + (Wed, 04 May 2011) | 19 lines Merged revisions 316616 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r316616 | dvossel | 2011-05-04 08:40:41 -0500 (Wed, 04 May 2011) + | 12 lines Fixes session-timers=refuse not being enforced for + *caller* During handle_request_invite, the session timer mode was + retrieved from a cached variable. This patch forces a peer lookup + of the session timer mode in the case of an incoming invite. + (closes issue #18804) Reported by: wdoekes Patches: + issue18804_session_timer_refuse_caller.patch uploaded by wdoekes + (license 717) issue_18804_v2.diff uploaded by dvossel (license + 671) ........ ................ + +2011-05-04 08:25 +0000 [r316552-316584] Damien Wedhorn + + * channels/chan_skinny.c: Add setsubstate_ringin. Added + setsubstate_ringin. skinny_call now calls sss_ringin rather than + inline. Fixed previous issue so that setsubstate_connected now + use SUBSTATE_RINGIN to determine is an AST_CONTROL_ANSWER should + be queued. + + * channels/chan_skinny.c: Make skinny_answer use + setsubsate_connected. Cosolidated the code so that skinny_answer + now uses the setsubstate procedures rather than doing the + handling inline. + +2011-05-04 07:13 +0000 [r316520] Tzafrir Cohen + + * autoconf/ast_check_pwlib.m4, /, configure: Merged revisions + 316193 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316193 | tzafrir | 2011-05-03 13:57:16 +0300 (ג', 03 מאי 2011) | + 8 lines Re-fix bashism in ./configure: s/let/$(( ))/ A + forward-port in r278985 accidentally re-introduced issue 17485. + Fixing it. Thanks to Jilles Tjoelker for the good report. (closes + issue #17485) ........ + +2011-05-04 07:10 +0000 [r316519] Damien Wedhorn + + * channels/chan_skinny.c: Cleanup skinny callinfo. Cosolidated the + working out of the callinfo to be sent into transmit_callinfo. + Replaced ambiguous sub->outgoing with calldirection which can be + SKINNY_INCOMING or SKINNY_OUTGOING (same value as the skinny + protocol). + +2011-05-04 02:39 +0000 [r316477] Sean Bright + + * /, apps/app_meetme.c: Merged revisions 316476 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316476 | seanbright | 2011-05-03 22:34:01 -0400 + (Tue, 03 May 2011) | 17 lines Merged revisions 316475 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r316475 | seanbright | 2011-05-03 22:23:01 -0400 (Tue, 03 May + 2011) | 10 lines Honor the C option to MeetMe when L is passed. + This fixes a case that r304773 and friends missed. (closes issue + #17317) Reported by: var Patches: meetme-continue-on-l_16218.diff + uploaded by var (license 1227) Tested by: seanbright ........ + ................ + +2011-05-04 00:13 +0000 [r316428-316430] Tilghman Lesher + + * /, addons/cdr_mysql.c, addons/res_config_mysql.c: Merged + revisions 316429 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316429 | tilghman | 2011-05-03 19:12:25 -0500 (Tue, 03 May 2011) + | 7 lines Escape column names in case they contain illegal + characters ('-') or reserved words. (closes issue #19063) + Reported by: festr Patches: patch uploaded by festr (license 443) + ........ + + * channels/chan_sip.c, CHANGES: If multiple [general] contexts + occur from sip.conf (usually due to external includes), merge + them. The original implementation of this did the merging of all + contexts with the same name in the realtime layer, but that + implementation severely breaks drivers which use the same context + name (e.g. iax.conf, type={peer,user}). Therefore, the + implementation needs to do the merging for particular entries + only, based upon what contexts would allow that in the channel + driver itself. This implementation is for chan_sip only, but + others could be added in the future. (closes issue #17957) + Reported by: marcelloceschia Patches: + chan-sip_parsing-general_branch162.patch uploaded by + marcelloceschia (license 1079) Tested by: tilghman + +2011-05-03 22:16 +0000 [r316337] Russell Bryant + + * /, channels/chan_skinny.c, pbx/pbx_dundi.c, channels/chan_mgcp.c: + Merged revisions 316336 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316336 | russell | 2011-05-03 17:13:31 -0500 (Tue, 03 May 2011) + | 8 lines Use htons() instead of ntohs() in some places. (closes + issue #19200) Reported by: wdoekes Patches: + issue19200-trunk.patch uploaded by wdoekes (license 717) + issue19200-1.8.x.patch uploaded by wdoekes (license 717) ........ + +2011-05-03 22:07 +0000 [r316335] David Vossel + + * main/channel.c, /: Merged revisions 316334 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316334 | dvossel | 2011-05-03 17:05:59 -0500 (Tue, 03 May 2011) + | 8 lines Fixes framehook segfault on indicate (closes issue + #19215) Reported by: irroot Patches: framehook_indicate.patch + uploaded by irroot (license 52) ........ + +2011-05-03 21:48 +0000 [r316333] Russell Bryant + + * /, apps/app_minivm.c: Merged revisions 316331 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316331 | russell | 2011-05-03 16:41:11 -0500 (Tue, 03 May 2011) + | 2 lines Resolve another warning. ........ + +2011-05-03 21:45 +0000 [r316332] David Vossel + + * channels/chan_local.c, /: Merged revisions 316330 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316330 | dvossel | 2011-05-03 16:37:59 -0500 + (Tue, 03 May 2011) | 24 lines Merged revisions 316329 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r316329 | dvossel | 2011-05-03 16:29:55 -0500 + (Tue, 03 May 2011) | 17 lines Merged revisions 316328 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r316328 | dvossel | 2011-05-03 16:27:59 -0500 (Tue, 03 May 2011) + | 10 lines Fixes chan_local crashs in local_fixup() Thanks OEJ + for tracking down the issue and submitting the patch. (closes + issue #19053) Reported by: oej Tested by: oej Review: + https://reviewboard.asterisk.org/r/1158/ ........ + ................ ................ + +2011-05-03 20:45 +0000 [r316293] Russell Bryant + + * channels/chan_unistim.c, main/udptl.c, main/fskmodem_float.c, + main/rtp_engine.c, /, res/res_musiconhold.c, apps/app_ices.c, + apps/app_followme.c, main/config.c, main/channel.c, main/cdr.c, + channels/chan_phone.c, funcs/func_enum.c, main/manager.c, + channels/chan_skinny.c, apps/app_minivm.c, main/features.c, + main/plc.c, res/res_agi.c, apps/app_amd.c, main/pbx.c, + res/res_fax.c, formats/format_wav.c, apps/app_festival.c, + channels/chan_agent.c, apps/app_originate.c, apps/app_queue.c, + codecs/lpc10/dyptrk.c, include/asterisk/linkedlists.h, + main/file.c, main/audiohook.c, pbx/pbx_config.c, main/asterisk.c, + main/dsp.c, res/res_calendar.c, apps/app_voicemail.c: Merged + revisions 316265 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) + | 5 lines Fix a bunch of compiler warnings generated by gcc + 4.6.0. Most of these are -Wunused-but-set-variable, but there + were a few others mixed in here, as well. ........ + +2011-05-03 19:22 +0000 [r316240] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_pri.c: Merged revisions 316224 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316224 | rmudgett | 2011-05-03 14:18:30 -0500 (Tue, 03 May 2011) + | 16 lines The dahdi_hangup() call does not clean up the channel + fully. After dahdi_hangup() has supposedly hungup an ISDN channel + there is still traffic on the S0-bus because the channel was not + cleaned up fully. Shuffled the hangup code to include some + missing cleanup. Also fixed some code formatting in the area. I + think the primary missing clean up code was the call to + tone_zone_play_tone() to turn off any active tones on the + channel. (closes issue #19188) Reported by: jg1234 Patches: + issue19188_v1.8.patch uploaded by rmudgett (license 664) Tested + by: jg1234 ........ + +2011-05-03 19:00 +0000 [r316216-316218] David Vossel + + * /, channels/chan_sip.c: Merged revisions 316217 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316217 | dvossel | 2011-05-03 13:59:06 -0500 (Tue, 03 May 2011) + | 9 lines Never put the Require: timer header in an Invite. This + has already been discussed and should have been resolved earlier. + View revsion 285565's log for more information about why it is + important to not put timer in the Require header. (closes issue + #18704) Reported by: mfrager ........ + + * /, res/res_odbc.c: Merged revisions 316215 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316215 | dvossel | 2011-05-03 13:49:48 -0500 (Tue, 03 May 2011) + | 9 lines Fixes a random crash (NULL reference) in res_odbc.c. + (closes issue #19180) Reported by: pruiz Patches: tmp.diff + uploaded by pruiz (license 1152) Tested by: pruiz, seanbright + ........ + +2011-05-03 18:23 +0000 [r316213] Sean Bright + + * main/manager.c, /: Merged revisions 316206 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r316206 | seanbright | 2011-05-03 14:17:36 -0400 (Tue, 03 May + 2011) | 8 lines If we aren't interested in events, don't generate + the FullyBooted event on AMI login. (closes issue #19089) + Reported by: bklang Patches: issue19089-1.8-r316204.patch + uploaded by seanbright (license 71) Tested by: seanbright + ........ + +2011-05-02 19:15 +0000 [r316095] Tilghman Lesher + + * funcs/func_curl.c, /: Merged revisions 316094 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r316094 | tilghman | 2011-05-02 14:09:55 -0500 + (Mon, 02 May 2011) | 15 lines Merged revisions 316093 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r316093 | tilghman | 2011-05-02 14:04:36 -0500 (Mon, 02 May 2011) + | 8 lines More possible crashes based upon invalid inputs. + (closes issue #18161) Reported by: wdoekes Patches: + 20110301__issue18161.diff.txt uploaded by tilghman (license 14) + Tested by: wdoekes ........ ................ + +2011-05-02 15:58 +0000 [r316054] Paul Belanger + + * apps/app_meetme.c: Formatting change, remove red blobs + +2011-04-27 19:15 +0000 [r315895] Matthew Nicholson + + * /, channels/chan_sip.c, channels/sip/reqresp_parser.c: Merged + revisions 315894 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315894 | mnicholson | 2011-04-27 14:14:27 -0500 + (Wed, 27 Apr 2011) | 28 lines Merged revisions 315893 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315893 | mnicholson | 2011-04-27 14:03:05 -0500 + (Wed, 27 Apr 2011) | 21 lines Merged revisions 315891 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315891 | mnicholson | 2011-04-27 13:57:56 -0500 (Wed, 27 Apr + 2011) | 14 lines Fix our compliance with RFC 3261 section 18.2.2. + This change optimizes the free_via() function and removes some + redundant null checking. It also fixes compliance with RFC 3261 + section 18.2.2 by always using the port specified in the Via + header for routing responses (even when maddr is not set). Also + the htons() function is now used when setting the port. + Additional documentation comments have been added in various + places to make the logic in the code clearer. (closes issue + #18951) Reported by: jmls Patches: + issue18951_set_proper_port_from_via.patch uploaded by wdoekes + (license 717) (modified) ........ ................ + ................ + +2011-04-27 17:51 +0000 [r315855-315856] David Vossel + + * apps/app_confbridge.c: Makes the new ConfBridge join and leave + sounds be used by default rather than beep and beeperr. + + * main/channel.c: Clears exception flag during ast_read when + func_jitterbuffer is enabled + +2011-04-27 15:56 +0000 [r315811] Russell Bryant + + * /, main/asterisk.c: Merged revisions 315810 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r315810 | russell | 2011-04-27 10:55:48 -0500 (Wed, 27 Apr 2011) + | 2 lines Set the copyright year to 2011 in the startup message. + ........ + +2011-04-27 12:37 +0000 [r315766] Leif Madsen + + * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 315765 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r315765 | lmadsen | 2011-04-27 07:36:17 -0500 (Wed, 27 Apr 2011) + | 4 lines Enable Russian core sound selection in menuselect. + (closes issue #18724) Reported by: pbxware ........ + +2011-04-26 23:10 +0000 [r315670-315675] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 315673 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315673 | twilson | 2011-04-26 15:56:19 -0700 + (Tue, 26 Apr 2011) | 25 lines Merged revisions 315672 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315672 | twilson | 2011-04-26 15:52:25 -0700 + (Tue, 26 Apr 2011) | 18 lines Merged revisions 315671 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315671 | twilson | 2011-04-26 15:47:56 -0700 (Tue, 26 Apr 2011) + | 11 lines Make sure unregistering a peer unlinks it from the + peer container Instead of mostly copying the code from + expire_register, just use the function that "does the right + thing". (closes issue #16033) Reported by: kkm Patches: + 016033-tilgman-fixed-refcount.diff uploaded by kkm (license 888) + Tested by: kkm, tilghman, twilson ........ ................ + ................ + + * channels/chan_sip.c: Make sure to create the caps structure for + autocreated peers Because crashing is bad. + + * apps/app_dial.c, main/features.c, apps/app_queue.c: Merged + revisions 315644 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315644 | twilson | 2011-04-26 14:39:01 -0700 + (Tue, 26 Apr 2011) | 32 lines Merged revisions 315643 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315643 | twilson | 2011-04-26 14:27:44 -0700 + (Tue, 26 Apr 2011) | 25 lines Merged revisions 315596 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) + | 18 lines Allow transfer loops without allowing forwarding loops + We try to avoid the situation where two phones may be forwarded + to each other causing an infinite loop by storing each dialed + interface in a channel datastore and checking the list before + dialing out. This works, but currently breaks situations like A + calls B, A transfers B to C, B transfers C to A, and A transfers + C to B. Since human interaction is happening here and not an + automated forwarding loop, it should be allowed. This patch + removes the dialed_interfaces datastore when a call is bridged (a + suggestion from the brilliant mmichelson). If a call is being + bridged, it should be safe to assume that we aren't stuck in a + loop. Since we are now handling this is the bridge code, the + previous attempts at handling it in app_dial and app_queue are + removed. Review: https://reviewboard.asterisk.org/r/1195/ + ........ ................ ................ + +2011-04-26 22:18 +0000 [r315649] Richard Mudgett + + * main/pbx.c, /: Merged revisions 315645 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r315645 | rmudgett | 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) + | 21 lines The 'e' special extension fails to trigger in at least + two cases. The 'e' extension is a fall back for the 'i', 't', or + 'T' extensions if any of them do not exist. Many of the places + the 'e' extension was supposed to be invoked fail because the + priority was set wrong. There were two places where the 'e' + extension was not even checked for fall back. * Made invoke the + 'e' extension similarly to the previous 'i', 't', or 'T' + extension check and added the 'e' extension as a fall back to the + two missing locations. * Prioritized and optimized some hangup + tests associated with the 'e' extension. (closes issue #19136) + Reported by: kshumard Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/1196/ ........ + +2011-04-26 19:38 +0000 [r315504] Tilghman Lesher + + * include/asterisk/select.h, /: Merged revisions 315503 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315503 | tilghman | 2011-04-26 14:32:50 -0500 + (Tue, 26 Apr 2011) | 28 lines Merged revisions 315502 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315502 | tilghman | 2011-04-26 14:22:52 -0500 + (Tue, 26 Apr 2011) | 21 lines Merged revisions 315501 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315501 | tilghman | 2011-04-26 14:18:46 -0500 (Tue, 26 Apr 2011) + | 14 lines Fix the bounds-checking code. The code that set the + bit within the select bitfield was correct, but the + bounds-checking code was not. The change to that line uses the + new _bitsize macro for clarity. Also, FD_ZERO macro did not + zero-out anything but the first word of the bitfield, so this + could have caused problems with modules using that macro with the + expanded bitfield. (closes issue #18773) Reported by: jamicque + Patches: 20110423__issue18773.diff.txt uploaded by tilghman + (license 14) Tested by: chris-mac ........ ................ + ................ + +2011-04-26 18:02 +0000 [r315453] Richard Mudgett + + * apps/app_dial.c, /: Merged revisions 315452 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r315452 | rmudgett | 2011-04-26 13:00:34 -0500 (Tue, 26 Apr 2011) + | 1 line Add missing set of name valid flag when dialing. + ........ + +2011-04-26 17:41 +0000 [r315447] Russell Bryant + + * channels/chan_local.c, /: Merged revisions 315446 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r315446 | russell | 2011-04-26 12:40:23 -0500 (Tue, 26 + Apr 2011) | 14 lines chan_local: resolve a deadlock. This patch + resolves a fairly complex deadlock that can occur with the + combination of chan_local and a dialplan switch, such as dynamic + realtime extensions, which pulls autoservice into the picture + when doing a dialplan lookup. (closes issue #18818) Reported by: + nic Patches: issue18818.patch uploaded by jthurman (license 614) + 18818.v1.txt uploaded by russell (license 2) Tested by: nic, + jthurman, kterzi, steve-howes, sysreq, IshMalik ........ + +2011-04-26 02:21 +0000 [r315395] Paul Belanger + + * /, pbx/pbx_config.c: Merged revisions 315394 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315394 | pabelanger | 2011-04-25 22:18:50 -0400 + (Mon, 25 Apr 2011) | 14 lines Merged revisions 315393 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r315393 | pabelanger | 2011-04-25 22:17:43 -0400 (Mon, 25 Apr + 2011) | 7 lines Add back CLI command 'dialplan save' (closes + issue #19140) Reported by: lmadsen Patches: + __20110419_dialplan_save.patch.txt uploaded by lmadsen (license + 10) ........ ................ + +2011-04-25 21:55 +0000 [r315350] Richard Mudgett + + * /, channels/chan_mgcp.c: Merged revisions 315349 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r315349 | rmudgett | 2011-04-25 16:49:00 -0500 (Mon, 25 + Apr 2011) | 9 lines When using MGCP realtime gateway definitions, + random crashes occur. Fixed incorrect linked list node removal + for realtime gateways. (closes issue #18291) Reported by: + nahuelgreco Patches: dangling-pointers-when-pruning.patch + uploaded by nahuelgreco (license 162) ........ + +2011-04-25 19:40 +0000 [r315214-315260] Russell Bryant + + * /, formats/format_wav.c: Merged revisions 315259 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315259 | russell | 2011-04-25 14:37:32 -0500 + (Mon, 25 Apr 2011) | 24 lines Merged revisions 315258 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315258 | russell | 2011-04-25 14:31:44 -0500 + (Mon, 25 Apr 2011) | 17 lines Merged revisions 315257 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315257 | russell | 2011-04-25 14:28:41 -0500 (Mon, 25 Apr 2011) + | 10 lines Be more flexible with unknown chunks in wav files. + This patch makes format_wav ignore unknown chunks instead of + erroring out on them. (closes issue #18306) Reported by: jhirsch + Patches: wav_skip_unknown_blocks.diff uploaded by jhirsch + (license 1156) ........ ................ ................ + + * /, channels/chan_sip.c: Merged revisions 315213 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315213 | russell | 2011-04-25 14:04:28 -0500 + (Mon, 25 Apr 2011) | 14 lines Merged revisions 315212 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r315212 | russell | 2011-04-25 14:00:24 -0500 (Mon, 25 Apr 2011) + | 7 lines Don't link non-cached realtime peers into the + peers_by_ip container. (closes issue #18924) Reported by: wdoekes + Patches: issue18924_uncached_realtime_peers_leak-1.6.2.17.patch + uploaded by wdoekes (license 717) ........ ................ + +2011-04-25 07:17 +0000 [r315054] Alec L Davis + + * channels/chan_local.c, /: Merged revisions 315053 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r315053 | alecdavis | 2011-04-25 19:14:32 +1200 + (Mon, 25 Apr 2011) | 23 lines Merged revisions 315052 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r315052 | alecdavis | 2011-04-25 19:11:12 +1200 + (Mon, 25 Apr 2011) | 16 lines Merged revisions 315051 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r315051 | alecdavis | 2011-04-25 19:06:29 +1200 (Mon, 25 Apr + 2011) | 11 lines chan_local:check_bridge() misplaced misplaced + ast_mutex_unlock if !p->chan->_bridge->_softhangup path isn't + followed, brigde remains locked. (closes issue #19176) Reported + by: alecdavis Patches: bug19176.diff.txt uploaded by alecdavis + (license 585) ........ ................ ................ + +2011-04-22 23:01 +0000 [r315002] Alec L Davis + + * channels/chan_dahdi.c, /: Merged revisions 315001 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r315001 | alecdavis | 2011-04-23 10:59:18 +1200 (Sat, 23 + Apr 2011) | 12 lines chan_dahdi: Can't return to normal ring + after distinctive ring on FXS clear a previous distinctivering + pattern before each new call (closes issue #18985) Reported by: + bromont Patches: bug18985.diff.txt uploaded by alecdavis (license + 585) Tested by: alecdavis, bromont ........ + +2011-04-22 21:33 +0000 [r314960] Matthew Nicholson + + * /, channels/chan_agent.c: Merged revisions 314959 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314959 | mnicholson | 2011-04-22 16:20:08 -0500 + (Fri, 22 Apr 2011) | 24 lines Merged revisions 314958 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r314958 | mnicholson | 2011-04-22 15:49:45 -0500 + (Fri, 22 Apr 2011) | 17 lines Merged revisions 311203,314908 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar + 2011) | 4 lines Don't hold the pvt lock while streaming a file. + ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48 + -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and + the app threads from using the asterisk channel at the same time. + ABE-2756 ........ ................ ................ + +2011-04-22 14:49 +0000 [r314824] Tzafrir Cohen + + * channels/chan_unistim.c, /, res/res_fax_spandsp.c: Merged + revisions 314779 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314779 | tzafrir | 2011-04-22 16:59:43 +0300 (ו', 22 אפר 2011) | + 2 lines Fix a few typos (shown by Lintian) ........ + +2011-04-22 14:08 +0000 [r314781] Russell Bryant + + * /, res/res_agi.c: Merged revisions 314780 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314780 | russell | 2011-04-22 09:02:23 -0500 + (Fri, 22 Apr 2011) | 18 lines Merged revisions 314778 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r314778 | russell | 2011-04-22 08:58:03 -0500 (Fri, 22 Apr 2011) + | 11 lines Initialize buffers in getvar and getvarfull. + Initialize the buffers used to hold the result from GET VARIABLE + or GET VARIABLE FULL. The bug report shows func_read returning + garbage in the result. It assumed that the buffer passed in was + initialized, like many other functions do. In the more common + code path (through the dialplan), it is initialized, so just + initialize it here too. (closes issue #19050) Reported by: johnz + ........ ................ + +2011-04-21 22:53 +0000 [r314733-314735] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Implement AMI action PRIShowSpans. PRIShowSpans works like the + AMI action DAHDIShowChannels but for PRI spans. It is similar to + the CLI command "pri show spans". (closes issue #15980) Reported + by: dwery + + * channels/sig_pri.c: Simplify sig_pri.c:build_status(). + + * channels/chan_dahdi.c, /: Merged revisions 314732 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r314732 | rmudgett | 2011-04-21 17:38:44 -0500 (Thu, 21 + Apr 2011) | 1 line Correct DAHDIShowChannels XML documentation. + ........ + +2011-04-21 18:32 +0000 [r314666] Matthew Nicholson + + * main/manager.c, /, channels/chan_sip.c, channels/chan_skinny.c, + main/http.c, configs/sip.conf.sample, configs/skinny.conf.sample, + channels/sip/include/sip.h, configs/http.conf.sample: Merged + revisions 314628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314628 | mnicholson | 2011-04-21 13:24:05 -0500 + (Thu, 21 Apr 2011) | 27 lines Merged revisions 314620 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r314620 | mnicholson | 2011-04-21 13:22:19 -0500 + (Thu, 21 Apr 2011) | 20 lines Merged revisions 314607 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr + 2011) | 14 lines Added limits to the number of unauthenticated + sessions TCP based protocols are allowed to have open + simultaneously. Also added timeouts for unauthenticated sessions + where it made sense to do so. Unrelated, the manager interface + now properly checks if the user has the "system" privilege before + executing shell commands via the Originate action. AST-2011-005 + AST-2011-006 (closes issue #18787) Reported by: kobaz (related to + issue #18996) Reported by: tzafrir ........ ................ + ................ + +2011-04-21 18:11 +0000 [r314598] David Vossel + + * configs/confbridge.conf.sample (added), apps/confbridge (added), + bridges/bridge_softmix.c, UPGRADE.txt, + include/asterisk/channel.h, res/res_musiconhold.c, CHANGES, + apps/confbridge/conf_config_parser.c (added), main/channel.c, + include/asterisk/bridging_technology.h, + bridges/bridge_builtin_features.c, + apps/confbridge/include/confbridge.h (added), apps/Makefile, + include/asterisk/bridging_features.h, + include/asterisk/bridging.h, include/asterisk/dsp.h, + apps/app_confbridge.c, apps/confbridge/include (added), + main/bridging.c, main/dsp.c: New HD ConfBridge conferencing + application. Includes a new highly optimized and customizable + ConfBridge application capable of mixing audio at sample rates + ranging from 8khz-192khz. Review: + https://reviewboard.asterisk.org/r/1147/ + +2011-04-21 00:29 +0000 [r314551] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 314550 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314550 | twilson | 2011-04-20 17:23:04 -0700 + (Wed, 20 Apr 2011) | 13 lines Merged revisions 314549 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r314549 | twilson | 2011-04-20 17:17:34 -0700 (Wed, 20 Apr 2011) + | 6 lines Don't allocate more space than necessary for a sip_pkt + This extra allocation is a hold-over from when pkt->data was a + character array. Now that it is an allocated string, just + allocate enough for the sip_pkt. ........ ................ + +2011-04-20 20:52 +0000 [r314509] David Vossel + + * main/channel.c, main/abstract_jb.c, funcs/func_jitterbuffer.c + (added), include/asterisk/channel.h, CHANGES, + include/asterisk/abstract_jb.h: Introduction of the JITTERBUFFER + dialplan function. Review: + https://reviewboard.asterisk.org/r/1157/ + +2011-04-20 19:56 +0000 [r314471] Shaun Ruffell + + * codecs/codec_dahdi.c: codec_dahdi: DAHDI still advertises formats + using the old bitfields. Previously, the DAHDI format bit fields + matched up with the Asterisk bitfields. Since the Asterisk codec + bit fields were replaced in r306010, codec_dahdi needs to contain + the formats itself. In the future, the DAHDI formats should + either change to something other than bitfields, or the bitfields + need to move from include/dahdi/kernel.h to include/dahdi/user.h. + Signed-off-by: Shaun Ruffell + +2011-04-20 16:55 +0000 [r314418] Richard Mudgett + + * /, include/asterisk/frame.h: Merged revisions 314417 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r314417 | rmudgett | 2011-04-20 11:54:02 -0500 (Wed, 20 + Apr 2011) | 1 line AST_CONTROL_XXX comment changes. ........ + +2011-04-20 16:37 +0000 [r314415] David Vossel + + * codecs/codec_resample.c: Fixes error with frame datalen being + calculated from samples when this is not allwaya accurate. + +2011-04-20 05:28 +0000 [r314359] Terry Wilson + + * main/lock.c, /: Merged revisions 314358 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314358 | twilson | 2011-04-19 22:25:15 -0700 (Tue, 19 Apr 2011) + | 4 lines Initialize track pointer ast_reentrancy_init checks to + see if it is NULL before initializing with calloc ........ + +2011-04-19 15:42 +0000 [r314204-314252] Leif Madsen + + * main/tcptls.c, /: Merged revisions 314251 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314251 | lmadsen | 2011-04-19 10:42:10 -0500 (Tue, 19 Apr 2011) + | 8 lines Use SSLv23_client_method instead of old SSLv2 only. + (closes issue #19095) (closes issue #19138) Reported by: tzafrir + Patches: no_ssl2.diff uploaded by tzafrir (license 46) Tested by: + russell, chazzam ........ + + * /, funcs/func_channel.c: Merged revisions 314206 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314206 | lmadsen | 2011-04-19 09:28:15 -0500 + (Tue, 19 Apr 2011) | 14 lines Merged revisions 314205 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19 Apr 2011) + | 6 lines Remove duplicate documentation from func_channel.c + (closes issue #18970) Reported by: IgorG Patches: + func_channel.c.doc.diff uploaded by IgorG (license 20) ........ + ................ + + * apps/app_dial.c, /: Merged revisions 314203 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r314203 | lmadsen | 2011-04-19 09:24:25 -0500 + (Tue, 19 Apr 2011) | 15 lines Merged revisions 314202 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r314202 | lmadsen | 2011-04-19 09:23:39 -0500 (Tue, 19 Apr 2011) + | 7 lines Update seconds to milliseconds in ast_verb output. + (closes issue #19084) Reported by: smurfix Patches: + app_dial.patch uploaded by smurfix (license 547) Tested by: + lmadsen, smurfix ........ ................ + +2011-04-19 08:22 +0000 [r314158] Olle Johansson + + * apps/app_meetme.c: Add explanation of strange flag setup in + app_meetme (stolen from Mark's message to asterisk-dev) + +2011-04-18 19:48 +0000 [r314079-314116] 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: Problems with ISDN MWI to phones. The + "controlling user number" is always the number of the voice mail + box which is identical with the subscriber number itself. This + number which is listed in the ISDN phone MWI menu cannot be + called back to contact the voice mail box. The controlling user + number should be made configurable. JIRA ABE-2738 JIRA SWP-2846 + + * /, res/res_agi.c: Merged revisions 314069 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314069 | rmudgett | 2011-04-18 11:10:10 -0500 (Mon, 18 Apr 2011) + | 22 lines The AsyncAGI command loop is lax in the value it + returns for the return status. * Return correct status: + SUCCESS/FAILED/HANGUP. Previously, abnormal exits from the + command loop such as hangup would return SUCCESS. * The "asyncagi + break" command now returns SUCCESS and is now the only way to + break the command loop with that status. Previously, it returned + FAILED. * The AMI event AsyncAGI End is no longer sent if the + AsyncAGI Start event is not sent. Previously, this happened + because of an error setting up the AGI pipes. * All executed AGI + commands now get an AsyncAGI Exec result event. Previously, if + the command returned failure (because of hangup), the command + loop just exited with FAILURE and did not send the AsyncAGI Exec + result event. * Makes sure that the channel frame queue is empty + on hangup. Review: https://reviewboard.asterisk.org/r/1183/ + ........ + + * apps/app_dial.c, /: Merged revisions 314068 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314068 | rmudgett | 2011-04-18 11:02:12 -0500 (Mon, 18 Apr 2011) + | 7 lines Unclear code in app_dial.c. Make code formatting clear. + (closes issue #19134) Reported by: oej ........ + +2011-04-18 16:22 +0000 [r314018-314078] David Vossel + + * /, channels/chan_sip.c: Merged revisions 314067 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314067 | dvossel | 2011-04-18 10:23:45 -0500 (Mon, 18 Apr 2011) + | 22 lines Remove the need for deadlock avoidance in chan_sip + do_monitor. Deadlock avoidance between the sip pvt and the + pvt->owner is very difficult. Now that channel's are ao2 objects, + this complication is no longer necessary. It turns out the pvt's + msg queue only exists because of deadlock avoidance (when + deadlock avoidance fails msgs were added to a queue to be + processed later), so this goes away as well. The technique used + in the new sip_lock_pvt_full() function should be used as a + template for replacing all locations where deadlock avoidance + occurs between a channel tech_pvt and the pvt's owner. My hope is + that this will begin a reversal of the invalid channel driver + locking architecture we have been using for so long. This patch + also resolves an issue where the pvt->owner gets unlocked during + processing the msg queue. (closes issue #18690) Reported by: + dvossel Review: https://reviewboard.asterisk.org/r/1182/ ........ + + * main/rtp_engine.c, /, channels/chan_sip.c, + include/asterisk/rtp_engine.h: Merged revisions 314017 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) + | 17 lines sip codec negotiation of dynamic rtp payloads error + fix This patch fixes how chan_sip handles dynamic rtp payload + types it does not understand. At the moment if a dynamic + payload's mime type does not match one we understand, the payload + does not get removed from our payload table. As a result of this, + the payload is set to whatever dynamic codec we use internally + for that payload number on outgoing INVITES. This is incorrect. + This patch fixes this by properly checking the rtpmap set + function's return code to make sure it was found. The function + can return both -1 and -2 depending on the source of the + mismatch. We were just checking -1 explicitly. Review: + https://reviewboard.asterisk.org/r/1169/ ........ + +2011-04-17 09:28 +0000 [r313980] Damien Wedhorn + + * channels/chan_skinny.c: Consolidate all new call calls to run + through new setsubstate_ringout. (closes issue #17907) Reported + by: wedhorn Patches: cleanup.stateringout.diff uploaded by + wedhorn (license 30) Tested by: salecha, wedhorn + +2011-04-17 01:28 +0000 [r313907-313944] Alexandr Anikin + + * addons/chan_ooh323.c: fix compile error from r313907 + + * addons/chan_ooh323.c: fix trivial error with set_max_datagram on + pvt->udptl + +2011-04-15 15:20 +0000 [r313867] Jonathan Rose + + * /, main/cli.c: Merged revisions 313860 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r313860 | jrose | 2011-04-15 10:08:05 -0500 + (Fri, 15 Apr 2011) | 17 lines Merged revisions 313859 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r313859 | jrose | 2011-04-15 09:58:37 -0500 (Fri, 15 Apr 2011) | + 10 lines Fix a Tab Completion bug that occurs due to multiple + matches on a substring. Makes word_match function in cli.c repeat + a search for a command string until a proper match is found or + the string is searched to the last point. (closes issue #17494) + Reported by: ffossard Review: + https://reviewboard.asterisk.org/r/1180/ ........ + ................ + +2011-04-14 21:53 +0000 [r313822] Terry Wilson + + * res/res_rtp_asterisk.c: Sets video mark bit on format field + correctly This fixes a regression in the media architecture + change where video frames did not have their video mark set + correctly. dvossel wrote this. twilson kindly committed this, + mmichelson found the bug. + +2011-04-14 21:02 +0000 [r313606-313781] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 313780 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r313780 | rmudgett | 2011-04-14 15:59:56 -0500 (Thu, 14 + Apr 2011) | 20 lines Leftover debug messages unconditionally sent + to the console. Executing Dial(DAHDI/1/18475551212,300,) with the + echotraining config option enabled outputs the following debug + messages unconditionally: Dialing T1847555121 on 1 Dialing www2w + on 1 * Made debug messages in my_dial_digits() normal debug + messages that do not get output unless enabled. * Reworded some + debug messages in my_dial_digits() to be clearer. * Replace + strncpy() with ast_copy_string() in my_dial_digits() which does + the same job better. (closes issue #18847) Reported by: + vmikhelson Tested by: rmudgett ........ + + * CREDITS, main/ccss.c, configs/ccss.conf.sample: Add Device State + Information CCSS for Generic Devices. Add Asterisk Device State + information and callbacks to the Call Completion Supplemental + Services for generic agents. There are currently not many devices + that have native support for CCSS. Even as the devices become + available there may be other reasons why one may choose to not + take advantage of the native abilities and stick with the generic + implementation. The generic implementation is quite capable and + could be greatly enhanced by adding device state capabilities. A + phone could then subscribe to the device state with a BLF key in + conjunction with Asterisk hints. The advantages of the device + state information would allow a single button to: request CCSS, + cancel a CCSS request, and display the current state of a CCSS + request. For example, you may have a single button that when not + lit, there is no active CCSS request. When you press that button, + the dialplan can query the DEVICE_STATE() associated with that + caller to determine whether they should be calling + CallCompletionRequest() or CallCompletionCancel(). If there is + currently a pending request, then the dialplan would cancel it. + This also has the advantage of showing the true state of a + request, which is an asynchronous call, even when + CallCompletionRequest() thinks it was successful. The actual + request could ultimately fail. Once lit, further feedback can be + provided to the caller about the current state of their request + since it will be updated by the CCSS State Machine as + appropriate. The DEVICE_STATE mapping is configurable since the + BLF being used on a given phone type may vary. The idea is to + allow some level of customization as to the phone's behavior. As + an example, you may want the BLF key to go solid once you have + requested a callback. You may then want the LED to blink + (typically ringing) when either the callback is in process, which + is a visual indication that the incoming call is the desired + callback. You may want it to blink when the callee is ready but + you are busy, giving you a visual indication that the target is + available as you may want to get off the line so that the + callback can be successful. Device state information is sent back + via the ast_devstate_prov_add() callback for any generic CCSS + device as it traverses through the state machine. You simply + provide a map between CC_STATE values and the corresponding + AST_DEVICE state values. You could then generate hints against + these states similar to what is possible today with Custom + Devstates or MeetMe states. For example, you may have an + extension 3000 that is currently associated with device SIP/3000. + You could then create a feature code for that extension that may + look something like: exten => *823000,hint,ccss:sip/3000 You + would then subscribe a BLF button to *823000 which would point to + the dialplan that handled CCSS requests/cancels using the + available DEVICE_STATE() information about ccss:sip/3000 to make + the decision about what to do. (closes issue #18788) Reported by: + p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p + lindheimer (license 558) Modified with final reviewboard + comments. Tested by: p_lindheimer, loloski Review: + https://reviewboard.asterisk.org/r/1105/ + + * /, res/res_agi.c: Merged revisions 313700 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313700 | rmudgett | 2011-04-13 17:52:47 -0500 (Wed, 13 Apr 2011) + | 5 lines Revert flushing stale AsyncAGI commands from -r313615. + It looks like it was intentional to leave any commands or + in-flight commands in the queue in case Async AGI is run again on + the call. ........ + + * /, res/res_agi.c: Merged revisions 313658 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313658 | rmudgett | 2011-04-13 12:47:43 -0500 (Wed, 13 Apr 2011) + | 2 lines Miscellaneous AGI diagnostic message cleanup and code + optimization. ........ + + * /, res/res_agi.c: Merged revisions 313615 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313615 | rmudgett | 2011-04-13 12:18:49 -0500 (Wed, 13 Apr 2011) + | 5 lines * Add missing channel lock to handle_cli_agi_add_cmd(). + * Flush any Async AGI commands left over from earlier Async AGI + control of the call. ........ + + * main/channel.c, /, res/res_agi.c: Merged revisions 313588 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 + (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 + (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) + | 41 lines Asterisk does not hangup a channel after endpoint + hangs up. If the call that the dialplan started an AGI script for + is hungup while the AGI script is in the middle of a command then + the AGI script is not notified of the hangup. There are many AGI + Exec commands that this can happen with. The reported + applications have been: Background, Wait, Read, and Dial. Also + the AGI Get Data command. * Don't wait on the Asterisk channel + after it has hung up. The channel is likely to never need + servicing again. * Restored the AGI script's ability to return + the AGI_RESULT_HANGUP value in run_agi(). It previously only + could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the + DeadAGI and AGI applications were merged. (closes issue #17954) + Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by + rmudgett (license 664) issue17954_v1.6.2.patch uploaded by + rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett + (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue + #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 + (closes issue #18935) Reported by: nvitaly Tested by: astmiv, + rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby + Tested by: rmudgett JIRA SWP-2727 Review: + https://reviewboard.asterisk.org/r/1165/ ........ + ................ ................ + +2011-04-13 15:49 +0000 [r313528] Leif Madsen + + * configs/iax.conf.sample, configs/users.conf.sample, + channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, + channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/chan_iax2.c, channels/sip/include/sip.h: Add + 'description' field for CLI and Manager output (closes issue + #19076) Reported by: lmadsen Patches: + __20110408-channel-description.txt uploaded by lmadsen (license + 10) Tested by: lmadsen Review: + https://reviewboard.asterisk.org/r/1163/ + +2011-04-13 15:23 +0000 [r313527] Richard Mudgett + + * /, apps/app_dumpchan.c: Merged revisions 313517 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313517 | rmudgett | 2011-04-12 17:35:53 -0500 (Tue, 12 Apr 2011) + | 12 lines Bring the dumpchan application inline with "core show + channel". * Added fields that are in "core show channel" to + dumpchan output. * Fixed reuse of formatbuf before the previous + string stored there was used by snprintf. All output strings now + have their own buffer. * Adjusted the buffer sizes to not be so + abusive of the stack now that there are more buffers. Change + requested by oej. ........ + +2011-04-12 21:59 +0000 [r313482] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooGkClient.c, + addons/ooh323c/src/ooLogChan.h, addons/chan_ooh323.c, + addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h, + addons/ooh323c/src/ooGkClient.h, addons/ooh323c/src/ooports.c, + addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/ooh323ep.h, + addons/ooh323c/src/ootypes.h, addons/ooh323c/src/ooLogChan.c, + addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooSocket.h, + addons/ooh323c/src/ooq931.c: IPv6 support for chan_ooh323 IPv6 + support for ooh323, bindaddr, peers and users ip can be IPv4 or + IPv6 addr correction for multi-homed mode (0.0.0.0 or :: + bindaddr) can work in dual 6/4 mode with :: bindaddr gatekeeper + mode isn't supported in v6 mode while (issue #18278) Reported by: + may213 Patches: ipv6-ooh323.patch uploaded by may213 (license + 454) Review: https://reviewboard.asterisk.org/r/1004/ + +2011-04-12 18:53 +0000 [r313437-313438] Jonathan Rose + + * /: blocking fix from 313436 that was already made in this commit + + * channels/chan_dahdi.c, /: Merged revisions 313435 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 also + went ahead and fixed the problem it introduces before committing. + ........ r313435 | jrose | 2011-04-12 13:44:44 -0500 (Tue, 12 Apr + 2011) | 1 line fixing stupid mistake with putting code before + variable declaration ........ Merged revisions 313433 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | + 14 lines reload Chan_dahdi memory leak caused by variables + chan_dahdi reloading with variables set via setvar in + chan_dahdi.conf would stay in the dahdi_pvt structs for + individual channels (causing them to just continue adding the new + ones to the list) and also there was a memory leak causes by the + conf objects. This patch resolves both of these by using + ast_variables_destroy during the loading process. (closes issue + #17450) Reported by: nahuelgreco Patches: patch.diff uploaded by + jrose (license 1225) Tested by: tilghman, jrose Review: + https://reviewboard.asterisk.org/r/1170/ ........ ........ + ........ + +2011-04-11 23:20 +0000 [r313367-313383] Richard Mudgett + + * apps/app_dial.c, /: Merged revisions 313368-313369 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r313368 | rmudgett | 2011-04-11 18:03:02 -0500 (Mon, 11 + Apr 2011) | 2 lines Backport a restructuring change from trunk to + make the next change stand out. ........ r313369 | rmudgett | + 2011-04-11 18:08:02 -0500 (Mon, 11 Apr 2011) | 13 lines Frames + from the inbound channel should go to all outbound channels in + app_dial.c. In app_dial.c:wait_for_answer() frames from the + inbound channel should be sent to all outbound channels instead + of only if there is just one outbound channel. Control frames + like AST_CONTROL_CONNECTED_LINE need to be passed to all of the + the outbound channels. This can happen if a blond transfer is + done by a remote switch on the inbound channel. JIRA AST-443 JIRA + SWP-2730 ........ + + * /, main/cli.c: Merged revisions 313366 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313366 | rmudgett | 2011-04-11 17:27:25 -0500 (Mon, 11 Apr 2011) + | 2 lines Added "Connected Line ID" and "Connected Line ID Name" + to "core show channel" output. ........ + +2011-04-11 19:39 +0000 [r313280] Leif Madsen + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 313279 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r313279 | lmadsen | 2011-04-11 14:36:40 -0500 + (Mon, 11 Apr 2011) | 21 lines Merged revisions 313278 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r313278 | lmadsen | 2011-04-11 14:33:03 -0500 + (Mon, 11 Apr 2011) | 14 lines Merged revisions 313277 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r313277 | lmadsen | 2011-04-11 14:30:20 -0500 (Mon, 11 Apr 2011) + | 6 lines Fix detection of OpenSSL 1.0 (closes issue #19093) + Reported by: tzafrir Patches: detect_openssl_10.diff uploaded by + tzafrir (license 46) ........ ................ ................ + +2011-04-11 15:47 +0000 [r313191] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 313190 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r313190 | rmudgett | 2011-04-11 10:40:30 -0500 + (Mon, 11 Apr 2011) | 39 lines Merged revisions 313189 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r313189 | rmudgett | 2011-04-11 10:32:53 -0500 + (Mon, 11 Apr 2011) | 32 lines Merged revisions 313188 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) + | 25 lines Stuck channel using FEATD_MF if caller hangs up at the + right time. The cause was actually a caller hanging up just at + the end of the Feature Group D DTMF tones that setup the call. + The reason for this is a "guard timer" that's implemented using + ast_safe_sleep(100). If the caller happens to hang up AFTER the + final tone of the DTMF string but BEFORE the end of that + ast_safe_sleep(), then ast_safe_sleep() will return non-zero. + This causes the code to bounce to the end of ss_thread(), but it + does NOT tear down the call properly. This should be a rare + occurrence because the caller has to hang up at EXACTLY the right + time. Nonetheless, it was happening quite regularly on the + reporter's system. It's not easily reproducible, unless you + purposely increase the guard-time to 2000 or more. Once you do + that, you can reproduce it every time by watching the DTMF debug + and hanging up just as it ends. Simply add an ast_hangup() before + goto quit. (closes issue #15671) Reported by: jcromes Patches: + issue15671.patch uploaded by pabelanger (license 224) Tested by: + jcromes ........ ................ ................ + +2011-04-09 21:00 +0000 [r313143] Alexandr Anikin + + * addons/chan_ooh323.c, /: Merged revisions 313142 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r313142 | may | 2011-04-10 00:56:17 +0400 (Sun, 10 Apr + 2011) | 3 lines fix trivial bug in ooh323_indicate on + AST_CONTROL_SRC... check p->rtp is not null ........ + +2011-04-08 16:17 +0000 [r313100] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, + channels/sig_ss7.h, channels/sig_pri.c, channels/sig_ss7.c: Add + private lock deadlock avoidance callback to PRI and SS7. Factor + out the equivalent function for analog. + +2011-04-07 13:42 +0000 [r313049] Jonathan Rose + + * /, main/features.c: Merged revisions 313048 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r313048 | jrose | 2011-04-07 08:35:33 -0500 + (Thu, 07 Apr 2011) | 16 lines Merged revisions 313047 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r313047 | jrose | 2011-04-07 08:23:01 -0500 (Thu, 07 Apr 2011) | + 9 lines Makes parking lots clear and rebuild properly when + features reload is invoked from CLI Before, default parkinglot in + context parkedcalls with ext 700 would always be present and when + reload was invoked, the previous parkinglots would not be + cleared. (closes issue #18801) Reported by: mickecarlsson Review: + https://reviewboard.asterisk.org/r/1161/ ........ + ................ + +2011-04-07 10:30 +0000 [r313003-313005] Alec L Davis + + * /, channels/sig_pri.c: Merged revisions 313001 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r313001 | alecdavis | 2011-04-07 22:19:31 +1200 (Thu, 07 Apr + 2011) | 13 lines Fix ISDN calling subaddr User Specified Odd/Even + Flag Calculation of the Odd/Even flag was wrong. Implement + correct algo, and set odd/even=0 if data would be truncated. Only + allow automatic calculation of the O/E flag, don't let dialplan + influence. (closes issue #19062) Reported by: festr Patches: + bug19062.diff2.txt uploaded by alecdavis (license 585) Tested by: + festr, alecdavis, rmudgett ........ + + * apps/app_voicemail.c: app_voicemail: close_mailbox change + LOG_WARNING to LOG_NOTICE + +2011-04-05 18:47 +0000 [r312868-312950] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: + Merged revisions 312949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r312949 | rmudgett | 2011-04-05 13:45:24 -0500 (Tue, 05 Apr 2011) + | 6 lines Crash if ISDN span layer 1 is down on initial load. + Regression from -r312575 B channel shifting during negotiation. * + Also combine updating the alarm flag with clearing the resetting + flag. ........ + + * /, channels/chan_sip.c: Merged revisions 312889 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r312889 | rmudgett | 2011-04-05 11:19:35 -0500 (Tue, 05 Apr 2011) + | 5 lines Add 416 response to OPTIONS packet. RFC3261 Section + 11.2 says the response code to an OPTIONS packet needs to be the + same as if it were an INVITE. ........ + + * /, channels/chan_sip.c: Merged revisions 312866 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r312866 | rmudgett | 2011-04-05 10:38:14 -0500 (Tue, 05 Apr 2011) + | 15 lines Responding to OPTIONS packet with 404 because Asterisk + not looking for "s" extension. The get_destination() function was + not using the "s" extension when the request URI did not specify + an extension. This is a regression caused when the URI parsing + code was extracted into parse_uri(). Made get_destination() + substitute the "s" extension when the parsed URI results in an + empty string. (closes issue #18348) Reported by: shmaize Patches: + issue18348_v1.8.patch uploaded by rmudgett (license 664) Tested + by: shmaize ........ + +2011-04-05 14:16 +0000 [r312767] Matthew Nicholson + + * main/manager.c, /, configs/manager.conf.sample: Merged revisions + 312766 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r312766 | mnicholson | 2011-04-05 09:14:50 -0500 + (Tue, 05 Apr 2011) | 22 lines Merged revisions 312764 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r312764 | mnicholson | 2011-04-05 09:13:07 -0500 + (Tue, 05 Apr 2011) | 15 lines Merged revisions 312761 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r312761 | mnicholson | 2011-04-05 09:10:34 -0500 (Tue, 05 Apr + 2011) | 8 lines Limit the number of unauthenticated manager + sessions and also limit the time they have to authenticate. + AST-2011-005 (closes issue #18996) Reported by: tzafrir Tested + by: mnicholson ........ ................ ................ + +2011-04-05 13:55 +0000 [r312756] Jonathan Rose + + * apps/app_meetme.c: Minor change to 'L' option for meetme to + include some verb statements for the option. + +2011-04-04 19:31 +0000 [r312716] Richard Mudgett + + * channels/sig_pri.c: Remove the channel parameter from + sig_pri_handle_subcmds(). It was only used in a debug message and + may not be correct anyway. + +2011-04-04 17:37 +0000 [r312678-312680] Jonathan Rose + + * pbx/pbx_config.c: In handle_cli_dialplan_add_extension, const + char pointer *into_context is used instead of a->argv[5] to + improve readability. + + * CHANGES, pbx/pbx_config.c: Makes 'dialplan add extension' create + the specified context if it does not already exist. If the user + invokes 'dialplan add extension' into a non-existing context, the + context will be created and a message informing the user of the + context being created will be issued in cli. (closes issue + #17431) Reported by: leearcher Patches: context_auto_create.diff + uploaded by kobaz (license 834) Tested by: leearcher, kobaz, + jrose + +2011-04-04 16:17 +0000 [r312579] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: + Merged revisions 312575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r312575 | rmudgett | 2011-04-04 11:10:50 -0500 + (Mon, 04 Apr 2011) | 52 lines Merged revisions 312574 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r312574 | rmudgett | 2011-04-04 11:00:02 -0500 + (Mon, 04 Apr 2011) | 45 lines Merged revisions 312573 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) + | 38 lines Issues with ISDN calls changing B channels during call + negotiations. The handling of the PROCEEDING message was not + using the correct call structure if the B channel was changed. + (The same for PROGRESS.) The call was also not hungup if the new + B channel is not provisioned or is busy. * Made all call + connection messages (SETUP_ACKNOWLEDGE, PROCEEDING, PROGRESS, + ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are + using the correct structure and B channel. If there is any + problem with the operations then the call is now hungup with an + appropriate cause code. * Made miscellaneous messages + (INFORMATION, FACILITY, NOTIFY) find the correct structure by + looking for the call and not using the channel ID. NOTIFY is an + exception with versions of libpri before v1.4.11 because a call + pointer is not available for Asterisk to use. * Made all hangup + messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find the correct + structure by looking for the call and not using the channel ID. + (closes issue #18313) Reported by: destiny6628 Tested by: + rmudgett JIRA SWP-2620 (closes issue #18231) Reported by: + destiny6628 Tested by: rmudgett JIRA SWP-2924 (closes issue + #18488) Reported by: jpokorny JIRA SWP-2929 JIRA AST-437 (The + issues fixed here are most likely causing this JIRA issue.) JIRA + DAHDI-406 JIRA LIBPRI-33 (Stuck resetting flag likely fixed) + ........ ................ ................ + +2011-04-01 23:17 +0000 [r312462-312510] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 312509 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r312509 | rmudgett | 2011-04-01 18:15:42 -0500 (Fri, 01 + Apr 2011) | 22 lines When a call going out an NT-PTMP port gets + rejected, Asterisk crashes. If a call is sent to an ISDN phone + that rejects the call with RELEASE_COMPLETE(cause: call + reject(21), or busy(17)) Asterisk crashes. I could not get my + setup to crash. However, I could see the possibility from a race + condition between queuing an AST_CONTROL_BUSY to the core and + then queueing an AST_CONTROL_HANGUP. If the AST_CONTROL_BUSY is + processed before the AST_CONTROL_HANGUP is queued, the + ast_channel could be destroyed out from under chan_misdn. Avoid + this particular crash scenario by not queueing the + AST_CONTROL_HANGUP if the AST_CONTROL_BUSY was queued. (closes + issue #18408) Reported by: wimpy Patches: issue18408_v1.8.patch + uploaded by rmudgett (license 664) Tested by: rmudgett, wimpy + JIRA SWP-2679 ........ + + * /, main/ccss.c: Merged revisions 312461 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) + | 25 lines CallCompletionRequest()/CallCompletionCancel() exit + non-zero if fail. The + CallCompletionRequest()/CallCompletionCancel() dialplan + applications exit nonzero on normal failure conditions. The + nonzero exit causes the dialplan to hangup immediately. The + dialplan author has no opportunity to report success/failure to + the user. * Made always return zero so the dialplan can continue. + * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and + CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively. + Also documented the values set. * Reduced the warning about no + core instance in CallCompletionCancel() to a debug message. It is + a normal event and should not be output at the WARNING level. + (closes issue #18763) Reported by: p_lindheimer Patches: + ccss.patch uploaded by p lindheimer (license 558) Modified Tested + by: p_lindheimer, rmudgett JIRA SWP-3042 ........ + +2011-04-01 17:28 +0000 [r312384-312423] Jonathan Rose + + * channels/chan_dahdi.c: Fixing bad line break from 312384 + + * channels/chan_dahdi.c, include/asterisk/dsp.h, CHANGES, + main/dsp.c: New Feature for chan_dahdi. 4 length pattern + matching. In chan_dahdi.conf, the user can now use length 4 + patterns in addition to the usual length 2 patterns. The s ntax + remains the same and the method used to track the pattern history + will only change when using the length 4 patterns. (closes issue + SWP-3250) Code: jrose rmudgett + +2011-04-01 10:59 +0000 [r312289] Tilghman Lesher + + * include/asterisk/select.h, /, addons/cdr_mysql.c, + main/asterisk.c: Merged revisions 312286,312288 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r312286 | tilghman | 2011-04-01 05:44:33 -0500 + (Fri, 01 Apr 2011) | 2 lines Reload must react correctly against + a possibly changed table, so dropping the conditional reload + flag. ................ r312288 | tilghman | 2011-04-01 05:58:45 + -0500 (Fri, 01 Apr 2011) | 21 lines Merged revisions 312287 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r312287 | tilghman | 2011-04-01 05:51:24 -0500 + (Fri, 01 Apr 2011) | 14 lines Merged revisions 312285 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) + | 7 lines Found some leaking file descriptors while looking at + ast_FD_SETSIZE dead code. (issue #18969) Reported by: oej + Patches: 20110315__issue18969__14.diff.txt uploaded by tilghman + (license 14) ........ ................ ................ + +2011-04-01 09:08 +0000 [r312118-312212] Alec L Davis + + * /, apps/app_voicemail.c: Merged revisions 312211 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r312211 | alecdavis | 2011-04-01 22:03:11 +1300 + (Fri, 01 Apr 2011) | 36 lines Merged revisions 312210 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r312210 | alecdavis | 2011-04-01 21:47:29 +1300 + (Fri, 01 Apr 2011) | 29 lines Merged revisions 312174 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r312174 | alecdavis | 2011-04-01 21:29:49 +1300 (Fri, 01 Apr + 2011) | 23 lines voicemail: get real last_message_index and + count_messages, ODBC resequence change last_message_index to read + the max msgnum stored in the database change count_messages to + actually count the number of messages. last_message_index change: + This fixed overwriting of the last message if msgnum=0 was + missing. Previously every incoming message would overwrite + msgnum=1. count_messages change: allows us to detect when + requencing is required in opneA_mailbox. resequence enabled for + ODBC storage: Assists with fixing up corrupt databases with gaps, + but only when a user actively opens there mailboxes. (closes + issue #18692,#18582,#19032) Reported by: elguero Patches: based + on odbc_resequence_mailbox2.1.diff uploaded by elguero (license + 37) Tested by: elguero, nivek, alecdavis Review: + https://reviewboard.asterisk.org/r/1153/ ........ + ................ ................ + + * /, apps/app_voicemail.c: Merged revisions 312117 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r312117 | alecdavis | 2011-04-01 20:32:12 +1300 + (Fri, 01 Apr 2011) | 29 lines Merged revisions 312103 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r312103 | alecdavis | 2011-04-01 20:25:54 +1300 + (Fri, 01 Apr 2011) | 22 lines Merged revisions 312070 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r312070 | alecdavis | 2011-04-01 19:46:56 +1300 (Fri, 01 Apr + 2011) | 16 lines app_voicemail: close_mailbox needs to respect + additional messages while mailbox is open. close_mailbox leave + gaps in message sequence if messages are deleted and new messages + arrive during this time, this is because the shuffle down to slot + 0, only shuffles the number of pre-existing messages when mailbox + is opened, ignoring new arrivals. Fix: in close_mailbox + re-evaluate number of messages before the shuffle, this then + includes new arrivals. Happens on filebased or ODBC storage. + (issues #19032,#18582,#18692,#18998) Reported by: + alecdavis,tootai,afosorio Review: + https://reviewboard.asterisk.org/r/1153/ ........ + ................ ................ + +2011-03-31 20:12 +0000 [r311981-312023] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 312022 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r312022 | rmudgett | 2011-03-31 15:11:40 -0500 (Thu, 31 + Mar 2011) | 14 lines chan_misdn segfaults when DEBUG_THREADS is + enabled. The segfault happens because jb->mutexjb is + uninitialized from the ast_malloc(). The internals of + ast_mutex_init() were assuming a nonzero value meant mutex + tracking initialization had already happened. Recent changes to + mutex tracking code to reduce excessive memory consumption + exposed this uninitialized value. Converted misdn_jb_init() to + use ast_calloc() instead of ast_malloc(). Also eliminated + redundant zero initialization code in the routine. (closes issue + #18975) Reported by: irroot ........ + + * include/asterisk/channel.h: Fix function reference in comment. + +2011-03-31 06:44 +0000 [r311931] Tilghman Lesher + + * /, configs/cdr_mysql.conf.sample: Merged revisions 311930 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311930 | tilghman | 2011-03-31 01:43:18 -0500 (Thu, 31 Mar 2011) + | 6 lines Incorrect default example; the field is actually + internally named "clid", not "callerid". (closes issue #19040) + Reported by: wcselby Tested by: tilghman ........ + +2011-03-30 01:57 +0000 [r311875] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 311874 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r311874 | rmudgett | 2011-03-29 20:56:05 -0500 (Tue, 29 + Mar 2011) | 1 line Update some setup_dahdi_int() comments. + ........ + +2011-03-29 08:33 +0000 [r311806] Tilghman Lesher + + * cel/cel_odbc.c, /: Merged revisions 311799 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311799 | tilghman | 2011-03-29 02:08:39 -0500 (Tue, 29 Mar 2011) + | 7 lines Remove extraneous check from integer-type fields. + (closes issue #19027) Reported by: mlehner Review: + https://reviewboard.asterisk.org/r/1149/ ........ + +2011-03-28 22:00 +0000 [r311752] Russell Bryant + + * /, apps/app_voicemail.c: Merged revisions 311751 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r311751 | russell | 2011-03-28 17:00:01 -0500 (Mon, 28 + Mar 2011) | 2 lines Cross-reference VoiceMail() and + VoiceMailMain() in the xml docs. ........ + +2011-03-27 21:49 +0000 [r311688] Alexandr Anikin + + * addons/chan_ooh323.c, /: Merged revisions 311687 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r311687 | may | 2011-03-28 01:47:13 +0400 (Mon, 28 Mar + 2011) | 2 lines correct return values in ooh323_indicate for + AST_CONTROL_T38_PARAMETERS ........ + +2011-03-23 21:55 +0000 [r311613-311616] Brett Bryant + + * /, apps/app_meetme.c: Merged revisions 311615 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311615 | bbryant | 2011-03-23 17:54:11 -0400 (Wed, 23 Mar 2011) + | 8 lines This patch fixes a bug with MeetMe behavior where the + 'P' option for always prompting for a pin is ignored for the + first caller. (closes issue #18070) Reported by: mav3rick Review: + https://reviewboard.asterisk.org/r/1132/ ........ + + * /, channels/sip/reqresp_parser.c: Merged revisions 311612 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311612 | bbryant | 2011-03-23 17:45:46 -0400 (Wed, 23 Mar 2011) + | 9 lines Fix a possible crash in sip/reqresp_parser.c that is + caused by a possible null value. (closes issue #18821) Reported + by: cmaj Patches: + patch-reqresp_parser_sip_uri_domain_cmp_c_locale-crash-1.8.3-rc2.diff.tx + uploaded by cmaj (license 830) ........ + +2011-03-23 02:51 +0000 [r311559] Terry Wilson + + * /, channels/sip/reqresp_parser.c: Merged revisions 311558 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311558 | twilson | 2011-03-22 19:24:53 -0700 (Tue, 22 Mar 2011) + | 5 lines Don't use static declared buf in parse_name_andor_addr + This function isn't used anywhere yet, but we definitely don't + want to keep the same value for buf between calls to the + function. ........ + +2011-03-22 15:26 +0000 [r311498] David Vossel + + * /, apps/app_meetme.c: Merged revisions 311497 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r311497 | dvossel | 2011-03-22 10:25:24 -0500 + (Tue, 22 Mar 2011) | 9 lines Merged revisions 311496 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r311496 | dvossel | 2011-03-22 10:24:45 -0500 (Tue, 22 + Mar 2011) | 2 lines Fixes memory leak in MeetMe AMI action + ........ ................ + +2011-03-18 19:05 +0000 [r311427] Jonathan Rose + + * CHANGES, apps/app_followme.c: Adds an option to FollowMe that + isn't useful for the bug it was made to solve. Still, due to the + nature of FollowMe, it makes sense to have this option since it + keeps apps bound to channels that would otherwise go away from + being lost. + +2011-03-18 16:27 +0000 [r311385] David Vossel + + * codecs/codec_resample.c: Remove libresample dependency from + codec_resample.c + +2011-03-18 16:24 +0000 [r311373] Jonathan Rose + + * /, channels/chan_sip.c, res/res_fax.c, res/res_jabber.c: Merged + revisions 311352 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311352 | jrose | 2011-03-18 11:19:05 -0500 (Fri, 18 Mar 2011) | + 10 lines Changes some print statements/events to use a blank + string in place of NULL if the string in question is NULL. This + is supposed to improve Solaris compatibility since Solaris goes + berserk when trying to output NULL strings. (closes issue #18759) + Reported by: bklang Patches: null-strings.patch uploaded by + bklang (license 919) ........ + +2011-03-18 16:03 +0000 [r311343] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 311342 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311342 | mnicholson | 2011-03-18 11:02:50 -0500 (Fri, 18 Mar + 2011) | 2 lines Properly populate the LOCALSTATIONID channel + variable. ........ + +2011-03-18 03:00 +0000 [r311296-311298] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 311297 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311297 | rmudgett | 2011-03-17 21:59:05 -0500 (Thu, 17 Mar 2011) + | 12 lines Race condition when ISDN CallRerouting/CallDeflection + invoked. The queued AST_CONTROL_BUSY could sometimes be processed + before the call_forward dial string is recognized. * Moved + setting the call_forwarding dial string after sending a response + to the initiator and just queue an empty frame to wake up the + media thread instead of an AST_CONTROL_BUSY. * Added check for + empty rerouting/deflection number and respond with an error. + ........ + + * apps/app_dial.c, /: Merged revisions 311295 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r311295 | rmudgett | 2011-03-17 21:22:07 -0500 + (Thu, 17 Mar 2011) | 35 lines Merged revision 310986 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r310986 | rmudgett | 2011-03-16 13:56:28 -0500 (Wed, + 16 Mar 2011) | 28 lines Dial() o option broke when connected line + feature added. The patch restores the o option behavior and adds + the ability to specify the CallerID. The Dial o and f options are + complementary to each other. The o option stores the CallerID on + the outgoing channel as the channel's CallerID. The f option + forces the CallerID sent by the outgoing channel. o(x) - The + argument 'x' is optional. If not present, then specify that the + CallerID that was present on the *calling* channel be stored as + the CallerID on the *called* channel. This was the behavior of + Asterisk 1.0 and earlier. If present, then specify the CallerID + stored on the *called* channel. Note that o(${CALLERID(all)}) is + similar to option o without parameters. f(x) - The argument 'x' + is optional and its presence changes the behavior of this option. + If not present, then force the outgoing CallerID on a + call-forward or deflection to the dialplan extension for this + Dial() using a dialplan 'hint'. For example, some PSTNs do not + allow CallerID to be set to anything other than the numbers + assigned to you. If present, then force the outgoing CallerID to + 'x'. Patches: jira_abe_2752_dial_fo_options.patch uploaded by + rmudgett (license 664) Tested by: rmudgett JIRA ABE-2752 JIRA + SWP-3096 .......... ................ + +2011-03-17 19:05 +0000 [r311198] Jonathan Rose + + * /, apps/app_chanspy.c: Merged revisions 311197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r311197 | jrose | 2011-03-17 14:03:34 -0500 (Thu, 17 Mar 2011) | + 11 lines This fixes a nasty chanspy bug which was causing a + channel leak every time a spied on channel made a call. In + addition to the above, it makes certain channel destruction + occurs so that applications don't get stuck waiting for datastore + destruction while monitored by chanspy. (closes issue #18742) + Reported by: jkister Tested by: jkister, jcovert, jrose Review: + http://reviewboard.digium.internal/r/106/ ........ + +2011-03-17 15:02 +0000 [r311142] Matthew Nicholson + + * main/manager.c, /: Merged revisions 311141 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r311141 | mnicholson | 2011-03-17 10:00:33 -0500 + (Thu, 17 Mar 2011) | 11 lines Merged revisions 311140 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r311140 | mnicholson | 2011-03-17 09:58:52 -0500 (Thu, 17 Mar + 2011) | 4 lines Don't write items to the manager socket twice. + AST-2011-003 (closes issue 0018987) Reported by: ks-steven + ........ ................ + +2011-03-17 10:51 +0000 [r311051] Alec L Davis + + * /, configs/indications.conf.sample: Merged revisions 311050 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r311050 | alecdavis | 2011-03-17 23:49:41 +1300 + (Thu, 17 Mar 2011) | 24 lines Merged revisions 311049 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r311049 | alecdavis | 2011-03-17 23:45:47 +1300 + (Thu, 17 Mar 2011) | 17 lines Merged revisions 311048 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r311048 | alecdavis | 2011-03-17 23:43:35 +1300 (Thu, 17 Mar + 2011) | 12 lines Remove extra quote in indications.conf Picking + low hanging fruit. (closes issue #18971) Reported by: IgorG + Patches: based on indications.conf.sample.diff uploaded by IgorG + (license 20) Tested by: IgorG ........ ................ + ................ + +2011-03-16 19:51 +0000 [r310941-311001] Terry Wilson + + * main/tcptls.c, /: Merged revisions 310999 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310999 | twilson | 2011-03-16 14:47:59 -0500 + (Wed, 16 Mar 2011) | 18 lines Merged revisions 310998 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r310998 | twilson | 2011-03-16 14:46:36 -0500 (Wed, 16 Mar 2011) + | 11 lines Fix crash on fdopen failure See security advisory + AST-2011-004 (closes issue #18845) Reported by: cmaj Patches: + patch-main-tcptls-1.8.3-rc2-open-session-crash-take2.diff.txt + uploaded by cmaj (license 830) + patch-main-tcptls-1.8.3-rc2-open-session-crash-take3.diff.txt + uploaded by cmaj (license 830) Tested by: cmaj, twilson ........ + ................ + + * main/manager.c, /: Merged revisions 310993 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310993 | twilson | 2011-03-16 14:26:57 -0500 + (Wed, 16 Mar 2011) | 11 lines Merged revisions 310992 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r310992 | twilson | 2011-03-16 14:23:03 -0500 (Wed, 16 Mar 2011) + | 4 lines Don't keep trying to write to a closed connection See + security advisory AST-2011-003. ........ ................ + + * /, main/features.c: Merged revisions 310902 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310902 | twilson | 2011-03-16 12:19:57 -0500 + (Wed, 16 Mar 2011) | 43 lines Merged revisions 310889 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r310889 | twilson | 2011-03-16 12:03:27 -0500 + (Wed, 16 Mar 2011) | 36 lines Merged revisions 310888 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) + | 29 lines Don't delay DTMF in core bridge while listening for + DTMF features This patch is mostly the work of Olle Johansson. I + did some cleanup and added the silence generating code if + transmit_silence is set. When a channel listens for DTMF in the + core bridge, the outbound DTMF is not sent until we have received + DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds + of DTMF to Asterisk, which sends no audio for those 4 seconds. + Some products see this delay and the time skew on RTP packets + that results and start ignoring the audio that is sent afterward. + With this change, the DTMF_BEGIN frame is inspected and checked. + If it matches a feature code, we wait for DTMF_END and activate + the feature as before. If transmit_silence=yes in asterisk.conf, + silence is sent if we paritally match a multi-digit feature. If + it doesn't match a feature, the frame is forwarded along with the + DTMF_END without delay. By doing it this way, DTMF is not + delayed. (closes issue #15642) Reported by: jasonshugart Patches: + issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license + 396) Tested by: globalnetinc, jde (closes issue #16625) Reported + by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/ + Review: https://reviewboard.asterisk.org/r/1125/ ........ + ................ ................ + +2011-03-15 01:49 +0000 [r310835] Tilghman Lesher + + * addons/chan_ooh323.c, /: Merged revisions 310834 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r310834 | tilghman | 2011-03-14 20:48:25 -0500 (Mon, 14 + Mar 2011) | 2 lines Fix branch compile. ........ + +2011-03-15 01:36 +0000 [r310833] Alec L Davis + + * /, main/utils.c: Merged revisions 310781 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310781 | alecdavis | 2011-03-15 14:00:55 +1300 (Tue, 15 Mar + 2011) | 10 lines core show locks: display ThreadID in hexadecimal + Allow easier cross referencing of thread ID's with GDB backtraces + (closes issue #18968) Reported by: alecdavis Patches: + bug18968.diff.txt uploaded by alecdavis (license 585) ........ + +2011-03-14 21:51 +0000 [r310735] Alexandr Anikin + + * addons/chan_ooh323.c, addons/ooh323c/src/ooCapability.c, /, + addons/ooh323c/src/ooCalls.h: Merged revisions 310734 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 (closes + issue #18693) ........ r310734 | may | 2011-03-15 00:45:53 +0300 + (Tue, 15 Mar 2011) | 12 lines Introduce t.38 parameters control + functionality not full but enough for Send/RcvFax support + Introduce t.38 controls between asterisk core and channel/proto + layers. Not all parameters are transferred from proto layers but + *Fax apps tested and work ok. (issue #18693) Reported by: + benngard2 Patches: issue-18693.patch uploaded by may213 (license + 454) ........ + +2011-03-14 16:55 +0000 [r310637] Richard Mudgett + + * /, main/callerid.c: Merged revisions 310636 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310636 | rmudgett | 2011-03-14 11:50:59 -0500 + (Mon, 14 Mar 2011) | 39 lines Merged revisions 310635 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r310635 | rmudgett | 2011-03-14 11:47:54 -0500 + (Mon, 14 Mar 2011) | 32 lines Merged revisions 310633 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r310633 | rmudgett | 2011-03-14 11:38:24 -0500 (Mon, 14 Mar 2011) + | 25 lines "Caller*ID failed checksum" on Wildcard TDM2400P and + TDM410 The last character in the caller id message is getting a + framing error. The checksum is the last character in the message. + A framing error in the checksum could be because: 1) The sender + did not send a full stop bit. 2) The sender cut off the FSK + carrier too soon. 3) The sender opted to send zero of the + specified zero to 10 trailing mark bits and round-off errors in + the code resulted in the code not being where it thought it was + in the demodulated bit stream. Bit 8 of 'b' is set when parity + error. Bit 9 of 'b' is set when framing error. Made ignore the + framing and parity error bits if the errored character is the + checksum. We can tolerate a framing/parity error there. The + checksum character validates the message. (closes issue #18474) + Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek + (license 636) (with modifications) Tested by: nivek ........ + ................ ................ + +2011-03-14 15:40 +0000 [r310547-310588] Jonathan Rose + + * /, funcs/func_volume.c: Merged revisions 310587 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310587 | jrose | 2011-03-14 10:27:57 -0500 + (Mon, 14 Mar 2011) | 15 lines Merged revisions 310585 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r310585 | jrose | 2011-03-14 08:56:22 -0500 (Mon, 14 Mar 2011) | + 8 lines Adds 'p' as an option to func_volume. When it is on, the + old behavior with DTMF controlling volume adjustment will be + enforced. When it is off, DTMF will not be processed by the + function. Programmed by Jonathan Rose Reviewed by David Vossel, + Leif Madsen, and Russell Bryant + http://reviewboard.digium.internal/r/93/ ........ + ................ + + * main/audiohook.c: Fixes null reference bug introduced by audio + hook changes that affects various OS distributions. Thanks David. + +2011-03-12 20:42 +0000 [r310416-310500] Tilghman Lesher + + * /, pbx/pbx_ael.c: Merged revisions 310462 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310462 | tilghman | 2011-03-12 14:27:54 -0600 + (Sat, 12 Mar 2011) | 45 lines Merged revisions 310448 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r310448 | tilghman | 2011-03-12 14:24:54 -0600 + (Sat, 12 Mar 2011) | 38 lines Recorded merge of revisions 310435 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r310435 | tilghman | 2011-03-12 14:22:07 -0600 (Sat, 12 Mar 2011) + | 31 lines Add AELSub, which provides a stable entry point into + AEL subroutines. This commit needs some explanation, given that + we're adding a new application into an existing release branch. + This is generally a violation of our release policy, except in + very limited circumstances, and I believe this is one of those + circumstances. The problem that this solves is one of the sanity + of using multiple dialplan languages to define a dialplan. In the + case of the reporter, he or she is using AEL is define + subroutines, while using Realtime extensions to invoke those + subroutines. While you can do this, it's based upon the reality + of AEL using actual dialplan extensions; however, there is no + guarantee that the details of _how_ AEL is compiled into + extensions will remain stable. In fact, at the time of this + commit, it has already changed twice, once in a fundamental way. + Now normally, a new application would only be added to trunk. + However, this application is explicitly to create a stable + user-level API between versions, and adding it to trunk only will + not solve the user's problem of switching between 1.6.2 and 1.8, + nor will it help anybody switching from 1.8 to 1.10. Therefore, + it needs to go into existing release branches. For the sake of + consistency, and also because one of the changes was between 1.4 + and 1.6.x, I am also electing to commit this to 1.4. (closes + issue #18910) Reported by: alexandrekeller Patches: + 20110304__issue18919__1.6.2.diff.txt uploaded by tilghman + (license 14) 20110304__issue18919__1.4.diff.txt uploaded by + tilghman (license 14) Tested by: alexandrekeller ........ + ................ ................ + + * /, funcs/func_odbc.c: Merged revisions 310415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310415 | tilghman | 2011-03-12 14:05:46 -0600 + (Sat, 12 Mar 2011) | 14 lines Merged revisions 310414 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r310414 | tilghman | 2011-03-12 13:51:23 -0600 (Sat, 12 Mar 2011) + | 7 lines Transactional handles should be used for the insertbuf, + if available. Also, fix a possible resource leak. (closes issue + #18943) Reported by: irroot ........ ................ + +2011-03-11 18:54 +0000 [r310373] Jonathan Rose + + * include/asterisk/audiohook.h, main/audiohook.c, CHANGES, + apps/app_mixmonitor.c: Mix Monitor: Now with r and t options. + +2011-03-11 15:09 +0000 [r310332] Kevin P. Fleming + + * Makefile, configure, codecs/gsm/Makefile, configure.ac, + makeopts.in, codecs/lpc10/Makefile: Use "-march=native" when + possible. Recent versions of GCC have a tuning option value of + 'native', which causes the compiler to optimize the build for the + CPU the compile is performed on. Since most people are building + Asterisk on the machine they plan to run it on, the configure + script and build system will now use this value unless a + different value is specified by the user in CFLAGS when the + configure script is executed. In addition, this value will be + used for building the GSM and LPC10 codecs as well, in preference + to the logic that has been in their Makefiles forever to optimize + for certain types of CPUs. + +2011-03-11 06:56 +0000 [r310288] Alec L Davis + + * main/rtp_engine.c, /: Merged revisions 310287 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310287 | alecdavis | 2011-03-11 19:47:44 +1300 (Fri, 11 Mar + 2011) | 17 lines remote_bridge_loop: prevent segfault when after + transfer of IAX2 of DAHDI call If the channel condition is one of + the following after breaking out of the loop, don't try to + update_peer (where x = 0/1) 1). ZOMBIE 2). cx->tech_pvt != pvtx + 3). gluex != ast_rtp_instance_get_glue(cx->tech->type)) (closes + issue #18781) Reported by: alecdavis Patches: bug18781.diff3.txt + uploaded by alecdavis (license 585) Tested by: alecdavis, ZX81 + Review: https://reviewboard.asterisk.org/r/1128/ ........ + +2011-03-10 16:09 +0000 [r310241] Terry Wilson + + * main/manager.c, /, res/res_phoneprov.c: Merged revisions 310240 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310240 | twilson | 2011-03-10 10:05:45 -0600 (Thu, 10 Mar 2011) + | 13 lines Add \r\n to remaining http headers passed to + ast_http_send r309204 changed the behavior of ast_http_send. It + now requires headers to be passed with trailing \r\n. This change + updates the remaining instances in the code that did not pass the + \r\n. (closes issue #18186) Reported by: nivaldomjunior Patches: + res_phoneprov.c.diff uploaded by lathama (license 1028) + manager.diff.txt uploaded by twilson (license 396) Tested by: + lathama ........ + +2011-03-10 15:28 +0000 [r310238] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 310231 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310231 | mmichelson | 2011-03-10 09:17:04 -0600 (Thu, 10 Mar + 2011) | 9 lines Be more tolerant of what URI we accept for call + completion PUBLISH requests. (closes issue #18946) Reported by: + GeorgeKonopacki Patches: 18946.patch uploaded by mmichelson + (license 60) Tested by: GeorgeKonopacki ........ + +2011-03-10 05:54 +0000 [r310143] Tilghman Lesher + + * res/res_config_odbc.c, /, funcs/func_odbc.c, + apps/app_voicemail.c: Merged revisions 310142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r310142 | tilghman | 2011-03-09 23:53:29 -0600 + (Wed, 09 Mar 2011) | 19 lines Merged revisions 310141 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r310141 | tilghman | 2011-03-09 23:51:37 -0600 + (Wed, 09 Mar 2011) | 12 lines Merged revisions 310140 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) + | 5 lines Initialize column size to 0 to deal with a potential + UnixODBC bug on 64-bit systems. (closes issue #18295) Reported + by: pruiz ........ ................ ................ + +2011-03-08 20:34 +0000 [r310089] Jonathan Rose + + * /, channels/sip/dialplan_functions.c: Merged revisions 310088 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310088 | jrose | 2011-03-08 14:19:32 -0600 (Tue, 08 Mar 2011) | + 9 lines Returns with an error notice if CHANNEL function of SIP + channel is read without arguments. (Closes issue #18653) Reported + by: wuwu Patches: diff.patch uploaded by jrose (license 1225) + Tested by: jrose ........ + +2011-03-08 18:19 +0000 [r310045] Terry Wilson + + * /, res/res_calendar.c: Merged revisions 310039 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r310039 | twilson | 2011-03-08 10:10:50 -0800 (Tue, 08 Mar 2011) + | 11 lines Spelling fix in "calendar show calendar" + s/Cartegories/Catagories/ (closes issue #18931) Reported by: + pdugas Patches: res_calendar.c.patch uploaded by pdugas (license + 1222) Review: [full review board URL with trailing slash] + ........ + +2011-03-08 16:46 +0000 [r309996] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 309994 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r309994 | rmudgett | 2011-03-08 10:37:02 -0600 (Tue, 08 Mar 2011) + | 1 line Make pri parameter description consistent. ........ + +2011-03-07 22:16 +0000 [r309859] Jonathan Rose + + * /, apps/app_mixmonitor.c: Merged revisions 309858 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309858 | jrose | 2011-03-07 16:07:25 -0600 + (Mon, 07 Mar 2011) | 22 lines Merged revisions 309857 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r309857 | jrose | 2011-03-07 16:04:44 -0600 + (Mon, 07 Mar 2011) | 15 lines Merged revisions 309856 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) | + 8 lines Bug fix for MixMonitor involving filenames with '.' not + in the extension Closes issue #18391) Reported by: pabelanger + Patches: bugfix.patch uploaded by jrose (license 1225) Tested by: + jrose ........ ................ ................ + +2011-03-07 01:01 +0000 [r309809] Tilghman Lesher + + * channels/chan_dahdi.c, /, configure, + include/asterisk/autoconfig.h.in, main/ast_expr2f.c, + configure.ac, main/ast_expr2.fl: Merged revisions 309808 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309808 | tilghman | 2011-03-06 18:54:42 -0600 + (Sun, 06 Mar 2011) | 14 lines Merged revisions 309251 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) + | 7 lines Revert previous 2 commits, and instead conditionally + redefine the same macro used in flex 2.5.35 that clashed with our + workaround. Not surprisingly, the workaround was exactly the same + code as was provided by the Flex maintainers, albeit in two + different places, in different macros. This should fix the + FreeBSD builds, which have an older version of Flex. ........ + ................ + +2011-03-07 00:14 +0000 [r309766] Mark Michelson + + * /, configs/sip.conf.sample: Merged revisions 309765 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r309765 | mmichelson | 2011-03-06 18:13:36 -0600 (Sun, + 06 Mar 2011) | 3 lines Indicate that Asterisk uses the Allow + header to determine if MESSAGE requests should be sent. ........ + +2011-03-05 17:53 +0000 [r309721] Moises Silva + + * channels/chan_dahdi.c, /: Merged revisions 309720 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r309720 | moy | 2011-03-05 12:44:30 -0500 (Sat, 05 Mar + 2011) | 6 lines Fix caller id passed to openr2_chan_make_call + (closes issue #18894) Reported by: malufrj Tested by: moy + ........ + +2011-03-05 10:30 +0000 [r309679] Tilghman Lesher + + * /, main/asterisk.c: Merged revisions 309678 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309678 | tilghman | 2011-03-05 04:29:30 -0600 + (Sat, 05 Mar 2011) | 14 lines Merged revisions 309677 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309677 | tilghman | 2011-03-05 04:28:24 -0600 (Sat, 05 Mar 2011) + | 7 lines Missed part of the conversion when we started passing + ppid to astcanary. (closes issue #18850) Reported by: viraptor + Patches: canary_ppid.patch uploaded by viraptor (license 543) + ........ ................ + +2011-03-04 23:22 +0000 [r309640] Terry Wilson + + * configs/calendar.conf.sample, include/asterisk/calendar.h, + CHANGES, res/res_calendar.c: Add setvar option to calendaring + Adding the setvar option with variable substitution on the value + allows things like setting the outbound caller id name to the + summary of a calendar event, etc. Values could be chained + together as they are appended in order to do some scripting if + necessary. Review: https://reviewboard.asterisk.org/r/1134/ + +2011-03-04 19:38 +0000 [r309493-309587] Matthew Nicholson + + * /, pbx/pbx_lua.c: Merged revisions 309585 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309585 | mnicholson | 2011-03-04 13:38:25 -0600 + (Fri, 04 Mar 2011) | 9 lines Merged revisions 309584 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r309584 | mnicholson | 2011-03-04 13:37:13 -0600 (Fri, + 04 Mar 2011) | 2 lines Restore mysterious lua_pushvalue() call + removed in r309494. The mystery has been solved. ........ + ................ + + * /, pbx/pbx_lua.c: Merged revisions 309542 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309542 | mnicholson | 2011-03-04 13:00:33 -0600 + (Fri, 04 Mar 2011) | 11 lines Merged revisions 309541 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309541 | mnicholson | 2011-03-04 12:59:20 -0600 (Fri, 04 Mar + 2011) | 4 lines Check for errors from fseek() when loading config + file, properly abort on errors from fread(), and supply a + traceback for errors generated when loading the config file. + Also, prepend a newline to traceback output so that the main + error message is on it's own line. ........ ................ + + * /, pbx/pbx_lua.c: Merged revisions 309495 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309495 | mnicholson | 2011-03-04 12:10:23 -0600 + (Fri, 04 Mar 2011) | 9 lines Merged revisions 309494 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r309494 | mnicholson | 2011-03-04 11:55:57 -0600 (Fri, + 04 Mar 2011) | 2 lines remove mysterious lua_pushvalue() that is + never used ........ ................ + + * pbx/pbx_lua.c, configs/extensions.lua.sample: Add support for + defining hints from pbx_lua (closes issue #16024) Reported by: + mnicholson + +2011-03-04 17:40 +0000 [r309491] Russell Bryant + + * channels/chan_nbs.c: Fix a buglet that prevented chan_nbs from + loading (and subsequently stopped Asterisk). In passing, convert + the return codes to be the proper AST_MODULE_LOAD_* constants. + +2011-03-04 16:00 +0000 [r309449] Matthew Nicholson + + * /, pbx/pbx_lua.c: Merged revisions 309448 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r309448 | mnicholson | 2011-03-04 09:59:25 -0600 (Fri, 04 Mar + 2011) | 8 lines Export global symbols from pbx_lua to allow + modules to be loaded. Fixes a regression introduced in r278132. + (closes issue #18671) Reported by: Igels Patches: + pbx_lua_global_symbols1.diff uploaded by mnicholson (license 96) + Tested by: Igels ........ + +2011-03-04 15:28 +0000 [r309446] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /, + funcs/func_channel.c, channels/sig_pri.c, UPGRADE-1.8.txt: Merged + revisions 309445 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) + | 46 lines Get real channel of a DAHDI call. Starting with + Asterisk v1.8, the DAHDI channel name format was changed for ISDN + calls to: DAHDI/i/[:]- + There were several reasons that the channel name had to change. + 1) Call completion requires a device state for ISDN phones. The + generic device state uses the channel name. 2) Calls do not + necessarily have B channels. Calls placed on hold by an ISDN + phone do not have B channels. 3) The B channel a call initially + requests may not be the B channel the call ultimately uses. + Changes to the internal implementation of the Asterisk master + channel list caused deadlock problems for chan_dahdi if it needed + to change the channel name. Chan_dahdi no longer changes the + channel name. 4) DTMF attended transfers now work with ISDN + phones because the channel name is "dialable" like the chan_sip + channel names. For various reasons, some people need to know + which B channel a DAHDI call is using. * Added + CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and + CHANNEL(dahdi_type) so the dialplan can determine the B channel + currently in use by the channel. Use CHANNEL(no_media_path) to + determine if the channel even has a B channel. * Added AMI event + DAHDIChannel to associate a DAHDI channel with an Asterisk + channel so AMI applications can passively determine the B channel + currently in use. Calls with "no-media" as the DAHDIChannel do + not have an associated B channel. No-media calls are either on + hold or call-waiting. (closes issue #17683) Reported by: mrwho + Tested by: rmudgett (closes issue #18603) Reported by: arjankroon + Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett + (license 664) Tested by: stever28, rmudgett ........ + +2011-03-04 01:52 +0000 [r309404] David Ruggles + + * /, apps/app_externalivr.c: Merged revisions 309403 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309403 | diruggles | 2011-03-03 20:50:44 -0500 + (Thu, 03 Mar 2011) | 23 lines Merged revisions 309356 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r309356 | diruggles | 2011-03-03 19:42:28 -0500 + (Thu, 03 Mar 2011) | 16 lines Merged revisions 309355 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar + 2011) | 9 lines fix small memory leak fix small memory leak + caused by a string allocation that wasn't freed (closes issue + #18907) Reported by: andy11 Patches: + asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 + (license 1224) ........ ................ ................ + +2011-03-02 21:08 +0000 [r309209-309300] Jason Parker + + * main/channel.c: Add HangupRequest manager event, to specify + when/where a channel gets hung up. (closes issue #18226) Reported + by: clegall_proformatique Patches: + asterisk_1.8_293157_hanguprequests.svn.patch uploaded by clegall + proformatique (license 1139) + + * /, channels/chan_sip.c: Merged revisions 309256 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309256 | qwell | 2011-03-02 13:54:20 -0600 + (Wed, 02 Mar 2011) | 15 lines Merged revisions 309255 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309255 | qwell | 2011-03-02 13:53:47 -0600 (Wed, 02 Mar 2011) | + 8 lines Fix usage of "hasvoicemail=yes" and "mailbox=" in + users.conf for SIP. Since it's a duplicate, nothing is going to + be done, so delme doesn't need to be set at all. Strangely, when + this was added, this was being set to 1 in 1.6, and 0 in trunk. + (issue AST-439) ........ ................ + + * /, main/http.c: Merged revisions 309204 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r309204 | qwell | 2011-03-01 16:25:44 -0600 (Tue, 01 Mar 2011) | + 7 lines Fix consistency of CRLFs on HTTP headers that get sent + out. (closes issue #18186) Reported by: nivaldomjunior Patches: + 18186-httpheadernewline.diff uploaded by qwell (license 4) + ........ + +2011-03-01 21:57 +0000 [r309127-309171] Richard Mudgett + + * /, funcs/func_channel.c: Merged revisions 309170 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r309170 | rmudgett | 2011-03-01 15:57:26 -0600 (Tue, 01 + Mar 2011) | 7 lines Document CHANNEL(keypad_digits) and + CHANNEL(no_media_path). * Added XML documentation for + CHANNEL(keypad_digits) and CHANNEL(no_media_path). * Tweaked XML + documentation for CHANNEL(reversecharge). ........ + + * channels/sig_analog.c, /: Merged revisions 309126 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r309126 | rmudgett | 2011-03-01 12:44:05 -0600 (Tue, 01 + Mar 2011) | 16 lines Chan_dahdi does not retain CID when + detecting DTMF CID without polarity reversal. Looks like an + unintended change when sig_analog.c was extracted from + chan_dahdi.c. Removed useless conditional around needed code and + fixed resulting compiler warning. (closes issue #18667) Reported + by: enegaard Patches: issue18667.patch uploaded by enegaard + (license 1197) Tested by: enegaard JIRA SWP-2965 ........ + +2011-03-01 16:22 +0000 [r309090] David Vossel + + * /, channels/chan_sip.c: Merged revisions 309084 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309084 | dvossel | 2011-03-01 10:09:11 -0600 + (Tue, 01 Mar 2011) | 15 lines Merged revisions 309083 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309083 | dvossel | 2011-03-01 10:05:25 -0600 (Tue, 01 Mar 2011) + | 9 lines Fixes thread blocking issue in the sip TCP/TLS + implementation. (closes issue #18497) Reported by: vois Patches: + issues_18497.diff uploaded by dvossel (license 671) Tested by: + vois, rossbeer, kowalma, Freddi_Fonet ........ ................ + +2011-02-28 11:16 +0000 [r308992-309036] Tilghman Lesher + + * /, configure, include/asterisk/autoconfig.h.in, + main/ast_expr2f.c, configure.ac, main/ast_expr2.fl: Merged + revisions 309035 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r309035 | tilghman | 2011-02-28 05:10:28 -0600 + (Mon, 28 Feb 2011) | 15 lines Merged revisions 309033-309034 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) + | 4 lines A later version of flex already includes the fwrite + workaround code, which if used twice causes a compilation error. + Detect whether Flex will compile without the workaround; if so, + suppress our workaround code. ........ r309034 | tilghman | + 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines Clarify + meaning, removing double negative (stupid!) ........ + ................ + + * /, funcs/func_odbc.c: Merged revisions 308991 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308991 | tilghman | 2011-02-28 03:33:22 -0600 + (Mon, 28 Feb 2011) | 14 lines Merged revisions 308990 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r308990 | tilghman | 2011-02-28 03:32:22 -0600 (Mon, 28 Feb 2011) + | 7 lines Statements updating zero rows may return SQL_NO_DATA. + This is fine; it's handled. (closes issue #18815) Reported by: + irroot Patches: func_odbc.insert_nodata.patch uploaded by irroot + (license 52) ........ ................ + +2011-02-25 18:58 +0000 [r308946] Alec L Davis + + * /, channels/chan_sip.c: Merged revisions 308945 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308945 | alecdavis | 2011-02-26 07:52:53 +1300 (Sat, 26 Feb + 2011) | 21 lines Fix Deadlock with attended transfer of SIP call + Call path sip_set_rtp_peer (locks chan then pvt) + transmit_reinvite_with_sdp try_suggested_sip_codec + pbx_builtin_getvar_helper (locks p->owner) But by the time + p->owner lock was attempted, seems as though chan and p->owner + were different. So in sip_set_rtp_peer, lock pvt first then lock + p->owner using deadlocking methods. (closes issue #18837) + Reported by: alecdavis Patches: bug18837-trunk.diff3.txt uploaded + by alecdavis (license 585) Tested by: alecdavis, Irontec, ZX81, + cmaj Review: [https://reviewboard.asterisk.org/r/1126/] ........ + +2011-02-24 21:43 +0000 [r308904] Richard Mudgett + + * main/channel.c, /: Merged revisions 308903 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308903 | rmudgett | 2011-02-24 15:38:41 -0600 (Thu, 24 Feb 2011) + | 9 lines Invalid read in ast_channel_set_caller_event(). + Valgrind reported that ast_channel_set_caller_event() was reading + data from a freed buffer when using the pre_set structure. + Rearange things to pre-calculate the name and number pointer + before updating the caller party structure to see if the name or + number was changed. ........ + +2011-02-24 17:59 +0000 [r308816] Terry Wilson + + * main/manager.c, /: Merged revisions 308815 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308815 | twilson | 2011-02-24 11:57:18 -0600 + (Thu, 24 Feb 2011) | 26 lines Merged revisions 308814 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r308814 | twilson | 2011-02-24 11:54:49 -0600 + (Thu, 24 Feb 2011) | 19 lines Merged revisions 308813 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r308813 | twilson | 2011-02-24 11:42:16 -0600 (Thu, 24 Feb 2011) + | 12 lines Don't broadcast FullyBooted to every AMI connection + The FullyBooted event should not be sent to every AMI connection + every time someone connects via AMI. It should only be sent to + the user who just connected. (closes issue #18168) Reported by: + FeyFre Patches: bug0018168.patch uploaded by FeyFre (license + 1142) Tested by: FeyFre, twilson ........ ................ + ................ + +2011-02-24 15:10 +0000 [r308724] Matthew Nicholson + + * main/udptl.c, /: Merged revisions 308723 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308723 | mnicholson | 2011-02-24 09:06:14 -0600 + (Thu, 24 Feb 2011) | 16 lines Merged revisions 308722 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r308722 | mnicholson | 2011-02-24 08:59:41 -0600 + (Thu, 24 Feb 2011) | 9 lines Merged revisions 308721 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r308721 | mnicholson | 2011-02-24 08:54:56 -0600 (Thu, + 24 Feb 2011) | 2 lines silence gcc 4.2 compiler warning ........ + ................ ................ + +2011-02-24 03:49 +0000 [r308680] Terry Wilson + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 308679 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308679 | twilson | 2011-02-23 21:41:34 -0600 + (Wed, 23 Feb 2011) | 15 lines Merged revisions 308678 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r308678 | twilson | 2011-02-23 21:38:22 -0600 (Wed, 23 Feb 2011) + | 8 lines Use remotesecret to authenticate with a remote party + The remotesecret option was only being used for outbound + registration and not for placing calls. This patch uses + remotesecret on outbound calls if it is set, otherwise secret is + still used. Review: https://reviewboard.asterisk.org/r/1107/ + ........ ................ + +2011-02-23 23:55 +0000 [r308623-308624] Richard Mudgett + + * main/translate.c: Fix compiler warning. + + * /, channels/sig_pri.c: Merged revisions 308622 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308622 | rmudgett | 2011-02-23 17:38:04 -0600 (Wed, 23 Feb 2011) + | 9 lines sig_pri_new_ast_channel() should return NULL when + new_ast_channel() fails. (closes issue #18874) Reported by: cmaj + Patches: + patch-sig_pri-crash-possible-null-channel-pointer.diff.txt + uploaded by cmaj (license 830) JIRA SWP-3172 ........ + +2011-02-22 23:04 +0000 [r308582] David Vossel + + * main/format.c, funcs/func_speex.c, main/frame.c, + main/rtp_engine.c, include/asterisk/silk.h (added), + codecs/speex/fixed_generic.h (added), bridges/bridge_softmix.c, + channels/chan_gtalk.c, bridges/bridge_multiplexed.c, + channels/chan_iax2.c, main/format_pref.c, codecs/speex/resample.c + (added), main/channel.c, funcs/func_pitchshift.c, + include/asterisk/audiohook.h, channels/chan_skinny.c, + main/format_cap.c, funcs/func_volume.c, codecs/speex (added), + codecs/codec_resample.c, include/asterisk/format.h, + codecs/speex/arch.h (added), include/asterisk/frame.h, + include/asterisk/rtp_engine.h, codecs/speex/stack_alloc.h + (added), main/bridging.c, apps/app_jack.c, + configs/codecs.conf.sample, res/res_rtp_asterisk.c, + formats/format_attr_silk.c (added), channels/chan_sip.c, + main/translate.c, main/slinfactory.c, codecs/codec_speex.c, + include/asterisk/_private.h, CHANGES, + codecs/speex/speex_resampler.h (added), res/res_mutestream.c, + include/asterisk/format_cap.h, codecs/Makefile, + channels/chan_jingle.c, main/data.c, channels/iax2.h, + main/audiohook.c, apps/app_chanspy.c, apps/app_mixmonitor.c, + main/asterisk.c, include/asterisk/slinfactory.h, + include/asterisk/translate.h, codecs/speex/resample_sse.h + (added), include/asterisk/time.h: Media Project Phase2: SILK + 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio + ConfBridge, and other stuff -Functional changes 1. Dynamic global + format list build by codecs defined in codecs.conf 2. SILK 8khz, + 12khz, 16khz, and 24khz with custom attributes defined in + codecs.conf 3. Negotiation of SILK attributes in chan_sip. 4. + SPEEX 32khz with translation 5. SLINEAR 8khz, 12khz, 24khz, + 32khz, 44.1khz, 48khz, 96khz, 192khz with translation using + codec_resample.c 6. Various changes to RTP code required to + properly handle the dynamic format list and formats with + attributes. 7. ConfBridge now dynamically jumps to the best + possible sample rate. This allows for conferences to take + advantage of HD audio (Which sounds awesome) 8. Audiohooks are no + longer limited to 8khz audio, and most effects have been updated + to take advantage of this such as Volume, DENOISE, PITCH_SHIFT. + 9. codec_resample now uses its own code rather than depending on + libresample. -Organizational changes Global format list is moved + from frame.c to format.c Various format specific functions moved + from frame.c to format.c Review: + https://reviewboard.asterisk.org/r/1104/ + +2011-02-22 15:33 +0000 [r308527] Andrew Latham + + * main/http.c: Use ast_debug for console logging Guessed the log + levels based on info that level 3 is the soft roof. Can we create + a page / document to define the levels? + +2011-02-21 15:04 +0000 [r308417] Matthew Nicholson + + * main/udptl.c, /: Merged revisions 308416 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308416 | mnicholson | 2011-02-21 09:02:20 -0600 + (Mon, 21 Feb 2011) | 19 lines Merged revisions 308414 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r308414 | mnicholson | 2011-02-21 09:00:22 -0600 + (Mon, 21 Feb 2011) | 12 lines Merged revisions 308413 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r308413 | mnicholson | 2011-02-21 08:57:15 -0600 (Mon, 21 Feb + 2011) | 5 lines Properly check the bounds of arrays when decoding + UDPTL packets. Also, remove broken support for receiving UDPTL + packets larger than 16k. That shouldn't ever happen anyway. + AST-2011-002 FAX-281 ........ ................ ................ + +2011-02-21 14:14 +0000 [r308372] Andrew Latham + + * main/http.c: Add HTTP URI Debug logging and update notice enable + reporting of the request URI / URL in debugging change funny + debug note to a serious note. + +2011-02-21 13:58 +0000 [r308371] Tzafrir Cohen + + * main/pbx.c: fix a memory leak in device state The callback + handle_statechange (pbx.c) fails to release its data pointer, + leaking memory in the process. Reported by: tzafrir Patches: + 18735_pbx_free_callback.diff uploaded by tzafrir (license 46) + Review: https://reviewboard.asterisk.org/r/1110/ + +2011-02-19 14:07 +0000 [r308331] Andrew Latham + + * main/http.c: Add CSS MIME Type Modern browsers are checking for + the MIME Type of pages and in some cases will not load a file if + the type is wrong. + +2011-02-19 11:03 +0000 [r308289] Tilghman Lesher + + * utils, /: Merged revisions 308288 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308288 | tilghman | 2011-02-19 05:02:49 -0600 (Sat, 19 Feb 2011) + | 2 lines A few more (copies of) files to ignore in this + directory. ........ + +2011-02-18 00:11 +0000 [r308243] Alexandr Anikin + + * addons/chan_ooh323.c, /, addons/ooh323cDriver.c, + addons/ooh323cDriver.h: Merged revisions 308242 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308242 | may | 2011-02-18 03:07:20 +0300 (Fri, 18 Feb 2011) | 3 + lines added g729onlyA option for announce only AnnexA g.729 codec + in h.323 capabilities. Option can be global or per user/peer. + ........ + +2011-02-17 20:21 +0000 [r308205] Richard Mudgett + + * channels/chan_dahdi.c: Add more verbage to CLI command 'pri show + channels' usage. + +2011-02-16 22:02 +0000 [r308157] Paul Belanger + + * /, addons/ooh323c/src/ooSocket.c: Merged revisions 308150 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308150 | pabelanger | 2011-02-16 15:21:17 -0500 (Wed, 16 Feb + 2011) | 2 lines Fix FreeBSD builds. ........ + +2011-02-16 08:06 +0000 [r308099] Alexandr Anikin + + * /, addons/ooh323c/src/ooSocket.c: Merged revisions 308098 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r308098 | may | 2011-02-16 10:57:22 +0300 (Wed, 16 Feb 2011) | 2 + lines ifdef __linux__ keepalive variables also ........ + +2011-02-15 23:34 +0000 [r308013] Jason Parker + + * /, apps/app_queue.c: Merged revisions 308010 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r308010 | qwell | 2011-02-15 17:34:03 -0600 + (Tue, 15 Feb 2011) | 24 lines Merged revisions 308007 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r308007 | qwell | 2011-02-15 17:33:24 -0600 + (Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | + 10 lines Fix regression that changed behavior of queues when + ringing a queue member. This reverts r298596, which was to fix a + highly bizarre and contrived issue with a queue member that + called into his own queue being transferred back into his own + queue. I couldn't reproduce that issue in any way. I think one of + the other recent transfer fixes actually fixed this. (closes + issue #18747) Reported by: vrban ........ ................ + ................ + +2011-02-15 23:07 +0000 [r307969] Alexandr Anikin + + * addons/ooh323c/src/ooSocket.c: include tcp keepalive socket calls + only on linux, freebsd and others don't have these options on + sockets. + +2011-02-15 21:42 +0000 [r307963-307964] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_pri.c: + Add CLI "pri show channels" command. List the current mapping of + DAHDI B channels to Asterisk channel names and which calls are on + hold or call-waiting. Calls on hold or call-waiting are not + associated with any B channel. JIRA LIBPRI-27 JIRA SWP-2547 + + * apps/app_dial.c, /: Merged revisions 307962 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) + | 1 line Don't crash when forcing caller id. ........ + +2011-02-15 18:09 +0000 [r307927] David Vossel + + * channels/chan_phone.c: Fixes compile error in chan_phone for big + endian + +2011-02-15 16:18 +0000 [r307883] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /, + channels/chan_sip.c, main/ccss.c, channels/sig_pri.c, + include/asterisk/ccss.h: Merged revisions 307879 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r307879 | rmudgett | 2011-02-15 10:13:55 -0600 (Tue, 15 + Feb 2011) | 37 lines No response sent for SIP CC + subscribe/resubscribe request. Asterisk does not send a response + if we try to subscribe for call completion after we have received + a 180 Ringing. You can only subscribe for call completion when + the call has been cleared. When we receive the 180 Ringing, for + this call, its call-completion state is 'CC_AVAILABLE'. If we + then send a subscribe message to Asterisk, it trys to change the + call-completion state to 'CC_CALLER_REQUESTED'. Because this is + an invalid state change, it just ignores the message. The only + state Asterisk will accept our subscribe message is in the + 'CC_CALLER_OFFERED' state. Asterisk will go into the + 'CC_CALLER_OFFERED' when the SIP client clears the call by + sending a CANCEL. Asterisk should always send a response. Even if + its a negative one. The fix is to allow for the CCSS core to + notify a CC agent that a failure has occurred when CC is + requested. The "ack" callback is replaced with a "respond" + callback. The "respond" callback has a parameter indicating + either a successful response or a specific type of failure that + may need to be communicated to the requester. (closes issue + #18336) Reported by: GeorgeKonopacki Tested by: mmichelson, + rmudgett JIRA SWP-2633 (closes issue #18337) Reported by: + GeorgeKonopacki Tested by: mmichelson JIRA SWP-2634 ........ + +2011-02-15 07:03 +0000 [r307751-307838] Tilghman Lesher + + * /, funcs/func_odbc.c: Merged revisions 307837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r307837 | tilghman | 2011-02-15 01:02:45 -0600 + (Tue, 15 Feb 2011) | 15 lines Merged revisions 307836 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r307836 | tilghman | 2011-02-15 01:01:37 -0600 (Tue, 15 Feb 2011) + | 8 lines Need to retrieve the rows affected before using the + associated variable. (closes issue #18795) Reported by: irroot + Patches: 20110211__issue18795.diff.txt uploaded by tilghman + (license 14) Tested by: tilghman ........ ................ + + * /, res/res_odbc.c: Merged revisions 307793 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r307793 | tilghman | 2011-02-14 14:16:55 -0600 + (Mon, 14 Feb 2011) | 15 lines Merged revisions 307792 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r307792 | tilghman | 2011-02-14 14:10:28 -0600 (Mon, 14 Feb 2011) + | 8 lines Increment usage count at first reference, to avoid a + race condition with many threads creating connections all at + once. (issue #18156) Reported by: asgaroth Patches: + 20110214__issue18156.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman ........ ................ + + * addons/chan_ooh323.c, addons/ooh323c/src/ooCmdChannel.c: Making + trunk compile again. + + * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 307750 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) + | 23 lines Calling a gosub routine defined in AEL from Dial/Queue + ceased to work. A bug in AEL did not distinguish between the "s" + extension generated by AEL and an "s" extension that was required + to exist by the chan_dahdi (or another channel) that was not + supplied with a starting extension. Therefore, AEL made incorrect + assumptions about what commands were permissable in the context. + This was fixed by making AEL generate a different extension name. + However, Dial and Queue make additional assumptions about the + name of the default gosub extension. Therefore, they needed to be + brought into line with a "macro" rendered by AEL (as a gosub), + without breaking traditional dialplans written without the aid of + AEL. Related to (issue #18480) Reported by: nivek (closes issue + #18729) Reported by: kkm Patches: 20110209__issue18729.diff.txt + uploaded by tilghman (license 14) + 018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888) + Tested by: kkm ........ + +2011-02-13 10:50 +0000 [r307677-307713] Alexandr Anikin + + * addons/ooh323c/src/ooLogChan.c, + addons/ooh323c/src/ooCmdChannel.c: lc not found - it's warning, + not error, change malloc to ast_calloc again + + * addons/chan_ooh323.c, addons/ooh323cDriver.c: change malloc to + ast_calloc calls to prevent crash of asterisk + +2011-02-10 22:43 +0000 [r307537] Jason Parker + + * contrib/init.d/rc.debian.asterisk, /, main/asterisk.c: Merged + revisions 307536 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r307536 | qwell | 2011-02-10 16:39:30 -0600 + (Thu, 10 Feb 2011) | 22 lines Merged revisions 307535 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r307535 | qwell | 2011-02-10 16:35:49 -0600 + (Thu, 10 Feb 2011) | 15 lines Merged revisions 307534 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | + 8 lines Remove color when executing commands via a remote + console. Essentially this makes '-x' imply '-n' on rasterisk. + This was done in a different and incomplete way previously, which + I'm reverting here. (issue #18776) Reported by: alecdavis + ........ ................ ................ + +2011-02-10 17:45 +0000 [r307468] Mark Michelson + + * /, configs/ccss.conf.sample: Merged revisions 307467 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r307467 | mmichelson | 2011-02-10 11:44:42 -0600 (Thu, + 10 Feb 2011) | 5 lines Fix a gaffe in the CCSS sample + configuration. Discovered by Philippe Lindheimer and pointed out + on #asterisk-dev ........ + +2011-02-10 17:12 +0000 [r307433] David Vossel + + * channels/chan_sip.c, main/format_cap.c, + include/asterisk/format_cap.h: Fixes bug in chan_sip where + nativeformats are not set correctly. The nativeformats field was + being overwritten when it should have been appended too. This + caused some format capabilities to be lost briefly and some log + warnings to be output. + +2011-02-10 13:29 +0000 [r307396] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c, + addons/ooh323c/src/ooh245.c, addons/ooh323c/src/ooCalls.h, + addons/ooh323c/src/ooSocket.c, addons/ooh323c/src/oochannels.c, + addons/ooh323c/src/ooStackCmds.c, addons/ooh323c/src/ooq931.c: + Corrections for properly work with H.323v2 (older) endpoints and + other small fixes. Interpret remote side H.225 version. + Corrections for H.323v2 endpoints: don't start TCS and MSD before + connect, don't start TCS and MSD by accepting H.245 connection, + start TCS and MSD by StartH245 facility message. Other fixes: fix + non zeroended remoteDisplayName issue, small fixes in call + clearing by closing H.245 connection, tcp keepalive introduced on + TCP connections (now is hardcoded, will be configurable in the + future), don't force H.245tunneling if FastStart is active, don't + send Alerting singal more than once per call. (closes issue + #18542) Reported by: vmikhelson Patches: issue18542-final-3.patch + uploaded by may213 (license 454) Tested by: vmikhelson + +2011-02-09 22:48 +0000 [r307359] Jeff Peeler + + * apps/app_meetme.c, CHANGES: Add new manager action + MeetmeListRooms. From the submitter: I've added a new manager + action to list only the active conferences on an Asterisk system. + It shows the same data displayed when you run a 'meetme list' on + the Asterisk CLI. (closes issue #17905) Reported by: rcasas + Patches: app_meetme.c.patch uploaded by rcasas (license 641) + Review: https://reviewboard.asterisk.org/r/874/ + +2011-02-09 21:46 +0000 [r307315] Andrew Latham + + * contrib/init.d/rc.debian.asterisk: Disable color during running + test (closes issue #18776) Reported by: alecdavis Patches: + ast_deb_init.diff uploaded by lathama (license 1028) Tested by: + andrel, lathama + +2011-02-09 21:08 +0000 [r307229-307274] Jeff Peeler + + * /, main/astobj2.c: Merged revisions 307273 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307273 | jpeeler | 2011-02-09 15:06:33 -0600 (Wed, 09 Feb 2011) + | 8 lines Add missing debug info for ao2_link for use with + REF_DEBUG in ao2 callback. (closes issue #18758) Reported by: + rgagnon Patches: branch-1.8-r306540-astobj-fix.diff uploaded by + rgagnon (license 1202) trunk-r306540-astobj-fix.diff uploaded by + rgagnon (license 1202) ........ + + * main/features.c, CHANGES: Allow parkedmusicclass to be settable + for non-default parking lots. (closes issue #17946) Reported by: + bluecrow76 Patches: + asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff + + * /, main/features.c: Merged revisions 307228 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r307228 | jpeeler | 2011-02-09 13:52:51 -0600 + (Wed, 09 Feb 2011) | 17 lines Merged revisions 307227 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r307227 | jpeeler | 2011-02-09 13:52:12 -0600 (Wed, 09 Feb 2011) + | 11 lines Make sure to set parking dial context for non-default + parking lots. Since parking_con_dial isn't settable, set all + parking lots to "park-dial". (closes issue #17946) Reported by: + bluecrow76 Patches: + asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by + bluecrow76 (license 270) modified by me ........ ................ + +2011-02-09 19:17 +0000 [r307192] Tzafrir Cohen + + * main/loader.c: clarify warning when no loadable module support + Clarify warning message when LOADABLE_MODULES is disabled but we + still try to load a module. + +2011-02-09 05:53 +0000 [r307143] Tilghman Lesher + + * main/lock.c, /: Merged revisions 307142 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307142 | tilghman | 2011-02-08 23:39:39 -0600 (Tue, 08 Feb 2011) + | 3 lines Initialize tracking variable in structure properly. + Fixes a memory leak. (Reported by The_Boy_Wonder on IRC, fixed by + me.) ........ + +2011-02-08 21:24 +0000 [r307097] Jason Parker + + * /, main/logger.c: Merged revisions 307092 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307092 | qwell | 2011-02-08 15:24:01 -0600 (Tue, 08 Feb 2011) | + 9 lines Fix issue with verbose messages not showing on remote + console. This code was reworked recently, and since the + logchannel list hadn't been created yet at this point, and it was + a verbose message, it was being dropped on the floor. Now it'll + continue on to where it should be handled. (closes issue #18580) + Reported by: pabelanger ........ + +2011-02-08 21:18 +0000 [r307071] Mark Michelson + + * /, main/ccss.c: Merged revisions 307065 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r307065 | mmichelson | 2011-02-08 15:13:08 -0600 (Tue, 08 Feb + 2011) | 6 lines Add a couple of useful channel variables for the + CC recall macro. CC_EXTEN and CC_CONTEXT will allow you to + determine the channel and context that will be called when the + recall occurs. ........ + +2011-02-08 20:42 +0000 [r307061] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 306979 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306979 | twilson | 2011-02-08 12:18:08 -0800 + (Tue, 08 Feb 2011) | 16 lines Merged revisions 306973 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306973 | twilson | 2011-02-08 12:14:09 -0800 + (Tue, 08 Feb 2011) | 9 lines Merged revisions 306972 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 + Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with + pedantic=yes ........ ................ ................ + +2011-02-08 20:31 +0000 [r307041] Andrew Latham + + * /, doc/asterisk.8, configs/asterisk.conf.sample, + configs/voicemail.conf.sample, doc/asterisk.sgml: 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 + +2011-02-08 19:42 +0000 [r306867-306968] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 306967 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306967 | jpeeler | 2011-02-08 13:41:42 -0600 + (Tue, 08 Feb 2011) | 16 lines Merged revisions 306966 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306966 | jpeeler | 2011-02-08 13:41:21 -0600 + (Tue, 08 Feb 2011) | 9 lines Merged revisions 306965 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08 + Feb 2011) | 1 line fix this line again ........ ................ + ................ + + * /, apps/app_voicemail.c: Merged revisions 306962 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306962 | jpeeler | 2011-02-08 13:25:38 -0600 + (Tue, 08 Feb 2011) | 22 lines Merged revisions 306961 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306961 | jpeeler | 2011-02-08 13:25:10 -0600 + (Tue, 08 Feb 2011) | 15 lines Merged revisions 306960 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) + | 9 lines Backup file storing message duration is not used with + IMAP_STORAGE, remove code. The message duration is stored in the + body of the email when using IMAP_STORAGE, so nothing needs to + happen with the backup file. (closes issue #18718) Reported by: + kerframil ........ ................ ................ + + * /, apps/app_voicemail.c: Merged revisions 306866 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306866 | jpeeler | 2011-02-08 10:21:45 -0600 + (Tue, 08 Feb 2011) | 16 lines Merged revisions 306865 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306865 | jpeeler | 2011-02-08 10:21:25 -0600 + (Tue, 08 Feb 2011) | 9 lines Merged revisions 306864 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08 + Feb 2011) | 1 line make this safer and fully correct, pointed out + by Steve Davis ........ ................ ................ + +2011-02-08 02:05 +0000 [r306827] Andrew Latham + + * doc/asterisk.sgml: Documentation Updates. Start updates to the + man pages. (issue #16505) Reported by: tzafrir Tested by: lathama + +2011-02-08 00:43 +0000 [r306755-306793] Richard Mudgett + + * configs/chan_dahdi.conf.sample: Define the MCID acronym in + chan_dahdi.conf.sample. + + * channels/sig_pri.h: Use correct conditional for MCID send. + + * channels/sig_pri.h, main/channel.c, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, funcs/func_frame_trace.c, + main/features.c, CHANGES, channels/sig_pri.c, + include/asterisk/frame.h: Pass a MCID request to the bridged + channel. Pass a MCID request to the bridged channel so the + bridged channel can send it to the network. The ability to send + the MCID request on an ISDN span is enabled with the new + chan_dahdi.conf mcid_send option. JIRA SWP-2845 JIRA ABE-2736 + +2011-02-07 22:46 +0000 [r306670-306675] Terry Wilson + + * /, main/features.c: Merged revisions 306674 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306674 | twilson | 2011-02-07 14:43:22 -0800 + (Mon, 07 Feb 2011) | 24 lines Merged revisions 306673 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306673 | twilson | 2011-02-07 14:40:20 -0800 + (Mon, 07 Feb 2011) | 17 lines Merged revisions 306672 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) + | 10 lines Don't try to pickup a call in the middle of a + masquerade If A calls B which doesn't answer and C & D both try + to do a call pickup, it is possible for ast_pickup_call to answer + the call, then fail to masquerade one of the calls because the + other one is already in the process of masquerading. This patch + checks to see if the channel is in the process of masquerading + before call before selecting it for a pickup. Review: + https://reviewboard.asterisk.org/r/1094/ ........ + ................ ................ + + * /, channels/chan_sip.c: Merged revisions 306619 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306619 | twilson | 2011-02-07 14:15:27 -0800 + (Mon, 07 Feb 2011) | 24 lines Merged revisions 306618 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306618 | twilson | 2011-02-07 13:59:54 -0800 + (Mon, 07 Feb 2011) | 17 lines Merged revisions 306617 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) + | 10 lines Don't allow a REFER w/replaces to replace its own + dialog Asterisk currently accepts a REFER with a Refer-To with an + embedded Replaces header that matches the dialog of the REFER. + This would be a situation like A calls B, A calls C, A transfers + B to A, which is just silly. This patch makes the transfer fail + instead of making Asterisk freak out and forget to hang other + channels up. Review: https://reviewboard.asterisk.org/r/1093/ + ........ ................ ................ + +2011-02-07 17:55 +0000 [r306576] Mark Michelson + + * /, main/ccss.c: Merged revisions 306575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r306575 | mmichelson | 2011-02-07 11:36:56 -0600 (Mon, 07 Feb + 2011) | 9 lines Rearrange a bit of code in the generic CC recall + operation. By waiting to call the callback macro after the + CC_INTERFACES, extension, priority, and context have been set, + this information can be accessed more easily within the callback + macro. Reported by Philippe Lindheimer. ........ + +2011-02-07 16:33 +0000 [r306541] David Vossel + + * channels/chan_sip.c: Fixes use of ast_format_cap_append where + ast_format_cap_copy is necessary. + +2011-02-05 22:16 +0000 [r306499] Alexandr Anikin + + * addons/chan_ooh323.c: fix trivial issue after dvossel patch, + initial zero fill user and peer structure before cap structure + allocated. + +2011-02-05 02:55 +0000 [r306464] Richard Mudgett + + * channels/chan_dahdi.c: Ignore voice frames in chan_dahdi native + bridging. Hardware is handling them. + +2011-02-04 22:37 +0000 [r306432] Jeff Peeler + + * main/manager.c: Send manager event for blackfilter only if it + DOES NOT match. The logic got reversed, oops. Works properly now + when multiple blackfilters are present. (closes issue #18283) + Reported by: telecos82 Patches: ast_managereventfilter.patch + uploaded by telecos82 (license 687) + +2011-02-04 20:30 +0000 [r306396] Richard Mudgett + + * apps/app_dial.c, channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, configure, + include/asterisk/autoconfig.h.in, configure.ac, CHANGES, + channels/sig_pri.c: Add ISDN display ie text handling options to + chan_dahdi.conf. The display ie handling can be controlled + independently in the send and receive directions with the + following options: * Block display text data. * Use display text + in SETUP/CONNECT messages for name. * Use display text for COLP + name updates (FACILITY/NOTIFY as appropriate). * Pass arbitrary + display text during a call. Sent in INFORMATION messages. + Received from any message that the display text was not used as a + name. If the display options are not set then the options default + to legacy behavior. The arbitrary display text is exchanged + between bridged channels using the AST_FRAME_TEXT frame type. To + send display text from the dialplan use the SendText() + application when the arbitrary display text option is enabled. + JIRA SWP-2688 JIRA ABE-2693 + +2011-02-04 19:24 +0000 [r306359] Jason Parker + + * /, apps/app_queue.c: Merged revisions 306356 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306356 | qwell | 2011-02-04 13:24:29 -0600 + (Fri, 04 Feb 2011) | 16 lines Merged revisions 306346 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | + 9 lines Don't fallthrough to 'unknown' in the 'ringing' case. + This could cause improper exits from the queue. (closes issue + #18499) Reported by: zaltar Patches: app_queue.patch uploaded by + zaltar (license 1148) ........ ................ + +2011-02-04 19:09 +0000 [r306325-306326] Richard Mudgett + + * tests/test_format_api.c: Fix compiler warning. + + * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 306324 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r306324 | rmudgett | 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011) + | 9 lines Don't send redirecting updates to the caller if the + dialplan forked the call. Each fork in the dial could be + redirected and confuse the caller. For ISDN the DivLeg1 and + DivLeg3 messages would get confused because ISDN redirects calls + in sequence not in parallel. * Also fixed a formatting + inconsistency in app_dial.c and make a warning message more + useful about what frame type could not be written. ........ + +2011-02-04 18:16 +0000 [r306258-306292] Paul Belanger + + * utils/extconf.c: Revert changes to extconf.c It seems extconf.c + already defines some local ast_debug() functions. Theses should + be removed and replaced with logger.h. A patch will be added to + reviewboard shortly. + + * cel/cel_radius.c, addons/chan_ooh323.c, apps/app_meetme.c, + main/say.c, channels/chan_gtalk.c, main/taskprocessor.c, + res/res_http_post.c, res/res_musiconhold.c, channels/chan_iax2.c, + res/res_jabber.c, pbx/pbx_loopback.c, main/channel.c, + channels/chan_dahdi.c, pbx/pbx_spool.c, main/manager.c, + res/res_smdi.c, channels/chan_skinny.c, main/features.c, + res/res_agi.c, main/http.c, main/logger.c, res/ais/evt.c, + main/app.c, res/res_config_ldap.c, apps/app_rpt.c, + res/res_rtp_asterisk.c, main/pbx.c, channels/chan_sip.c, + apps/app_fax.c, include/asterisk/channel.h, channels/sig_pri.c, + channels/chan_misdn.c, include/asterisk/sched.h, utils/extconf.c, + codecs/codec_ilbc.c, main/audiohook.c, res/res_odbc.c, + main/xmldoc.c, apps/app_voicemail.c: Replace ast_log(LOG_DEBUG, + ...) with ast_debug() (closes issue #18556) Reported by: kkm + Review: https://reviewboard.asterisk.org/r/1071/ + +2011-02-04 16:42 +0000 [r306257] David Vossel + + * codecs/codec_ilbc.c, codecs/ex_ilbc.h: Fix compile error in codec + ilbc translator. + +2011-02-03 23:50 +0000 [r306216] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 306215 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r306215 | jpeeler | 2011-02-03 17:49:28 -0600 (Thu, 03 Feb 2011) + | 20 lines Fix SIP deadlock involving state changes. Once again a + call to pbx_builtin_getvar_helper (and pbx_builtin_setvar_helper) + has caused locking problems. Both of these functions lock the + channel when the channel argument is passed in! In this case, the + suspected problem (the backtrace makes it impossible to tell) was + the private being locked in sip_set_rtp_peer and then: + transmit_reinvite_with_sdp try_suggested_sip_codec + pbx_builtin_getvar_helper (Traced to verify that the fix was only + required in 1.8 and later.) (closes issue #18491) Reported by: + cmaj Patches: chan_sip_fix_deadlocks_bug_18491.txt uploaded by + cmaj (license 830) Tested by: cmaj ........ + +2011-02-03 21:13 +0000 [r306128] Terry Wilson + + * channels/chan_local.c, /: Merged revisions 306127 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306127 | twilson | 2011-02-03 13:03:26 -0800 + (Thu, 03 Feb 2011) | 23 lines Merged revisions 306126 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r306126 | twilson | 2011-02-03 12:56:00 -0800 + (Thu, 03 Feb 2011) | 16 lines Merged revisions 306119 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) + | 9 lines Set hangup cause in local_hangup When a call involves a + local channel (like SIP -> Local -> SIP), the hangup cause was + not being set. This resulted in SIP channels sometimes getting a + 503 error instead of a 486 when the far side sent a busy. In + Asterisk 1.8+ this also can cause issues with CCSS that involve a + local channel. This patch sets the hangupcause for one side of + the local channel to the other in local_hangup for outbound + calls. ........ ................ ................ + +2011-02-03 20:51 +0000 [r306125] Jeff Peeler + + * /, main/features.c: Merged revisions 306124 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r306124 | jpeeler | 2011-02-03 14:50:48 -0600 + (Thu, 03 Feb 2011) | 17 lines Merged revisions 306123 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r306123 | jpeeler | 2011-02-03 14:49:48 -0600 (Thu, 03 Feb 2011) + | 10 lines Set exception on channel in parking thread when + POLLPRI event detected. This is done just to make the code be + equivalent to the old select code. As noted in 303106 the same + issue was already fixed in this branch, but the exception was not + set on the channel in the case of POLLPRI. The reason that this + did not cause a problem here is because in 122923 the check in + __ast_read to check the exception flag was removed. (related to + #18637) ........ ................ + +2011-02-03 18:37 +0000 [r306086] Jason Parker + + * main/frame.c: Modify alignment of 'core show codecs', since the + ID is no longer a huge int. + +2011-02-03 18:12 +0000 [r306010-306053] David Vossel + + * main/frame.c: Fixes output of "core show codecs" to display image + types correctly. + + * apps/app_dahdibarge.c, channels/chan_local.c, main/frame.c, + apps/app_record.c, apps/app_alarmreceiver.c, + bridges/bridge_softmix.c, formats/format_sln16.c, + apps/app_ices.c, bridges/bridge_multiplexed.c, + channels/chan_iax2.c, main/astobj2.c, res/res_rtp_multicast.c, + channels/chan_dahdi.c, include/asterisk/bridging_technology.h, + funcs/func_pitchshift.c, pbx/pbx_spool.c, + include/asterisk/audiohook.h, channels/chan_skinny.c, + channels/sip/include/globals.h, apps/app_dumpchan.c, + formats/format_pcm.c, formats/format_h263.c, main/bridging.c, + codecs/ex_ulaw.h, channels/sip/include/sip.h, main/pbx.c, + codecs/codec_g722.c, formats/format_wav.c, codecs/codec_g726.c, + formats/format_ogg_vorbis.c, bridges/bridge_simple.c, + include/asterisk/channel.h, apps/app_talkdetect.c, + channels/iax2-parser.c, include/asterisk/format_cap.h (added), + apps/app_speech_utils.c, channels/iax2-parser.h, main/data.c, + funcs/func_channel.c, main/audiohook.c, codecs/codec_dahdi.c, + include/asterisk/frame_defs.h, formats/format_g726.c, + apps/app_mixmonitor.c, main/asterisk.c, res/res_calendar.c, + apps/app_voicemail.c, channels/chan_vpb.cc, addons/format_mp3.c, + formats/format_sln.c, apps/app_dictate.c, codecs/ex_g722.h, + codecs/codec_gsm.c, codecs/ex_g726.h, channels/chan_gtalk.c, + include/asterisk/abstract_jb.h, main/channel.c, apps/app_mp3.c, + codecs/codec_resample.c, formats/format_h264.c, + formats/format_siren14.c, apps/app_rpt.c, channels/chan_mgcp.c, + codecs/codec_lpc10.c, channels/chan_sip.c, codecs/ex_lpc10.h, + include/asterisk/format_pref.h (added), codecs/codec_alaw.c, + res/res_adsi.c, tests/test_format_api.c (added), + apps/app_originate.c, channels/chan_jingle.c, + formats/format_vox.c, main/abstract_jb.c, + include/asterisk/bridging.h, main/callerid.c, main/file.c, + apps/app_sms.c, formats/format_g723.c, main/dsp.c, main/format.c + (added), main/udptl.c, main/rtp_engine.c, addons/chan_ooh323.c, + codecs/codec_adpcm.c, apps/app_test.c, addons/chan_ooh323.h, + include/asterisk/speech.h, codecs/ex_adpcm.h, codecs/ex_alaw.h, + formats/format_wav_gsm.c, include/asterisk/data.h, + codecs/ex_gsm.h, main/indications.c, main/format_pref.c (added), + main/cli.c, main/features.c, include/asterisk/mod_format.h, + apps/app_amd.c, addons/ooh323cDriver.c, channels/chan_alsa.c, + formats/format_jpeg.c, addons/ooh323cDriver.h, + formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c, + formats/format_g719.c, channels/h323/ast_h323.cxx, + channels/chan_bridge.c, apps/app_echo.c, apps/app_fax.c, + codecs/codec_speex.c, include/asterisk/slin.h, + channels/chan_agent.c, channels/iax2-provision.c, + codecs/ex_speex.h, channels/chan_misdn.c, + include/asterisk/image.h, channels/iax2.h, codecs/codec_ilbc.c, + apps/app_chanspy.c, res/res_fax_spandsp.c, + include/asterisk/slinfactory.h, include/asterisk/translate.h, + channels/chan_unistim.c, channels/chan_multicast_rtp.c, + main/ccss.c, apps/app_meetme.c, res/res_musiconhold.c, + apps/app_followme.c, formats/format_siren7.c, + formats/format_ilbc.c, include/asterisk/file.h, + include/asterisk/callerid.h, channels/chan_phone.c, main/dial.c, + main/manager.c, main/format_cap.c (added), + funcs/func_frame_trace.c, res/res_agi.c, main/app.c, + apps/app_confbridge.c, include/asterisk/format.h (added), + main/image.c, include/asterisk/rtp_engine.h, + include/asterisk/frame.h, addons/chan_mobile.c, + apps/app_parkandannounce.c, apps/app_jack.c, + res/res_clioriginate.c, res/res_rtp_asterisk.c, + apps/app_nbscat.c, codecs/codec_a_mu.c, res/res_fax.c, + apps/app_festival.c, apps/app_waitforsilence.c, + include/asterisk/astobj2.h, main/slinfactory.c, main/translate.c, + channels/chan_console.c, channels/h323/chan_h323.h, + channels/chan_oss.c, channels/chan_usbradio.c, + channels/chan_h323.c, codecs/codec_ulaw.c, + include/asterisk/pbx.h, channels/chan_nbs.c, + formats/format_g729.c: Asterisk media architecture conversion - + no more format bitfields This patch is the foundation of an + entire new way of looking at media in Asterisk. The code present + in this patch is everything required to complete phase1 of my + Media Architecture proposal. For more information about this + project visit the link below. + https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal + The primary function of this patch is to convert all the usages + of format bitfields in Asterisk to use the new format and + format_cap APIs. Functionally no change in behavior should be + present in this patch. Thanks to twilson and russell for all the + time they spent reviewing these changes. Review: + https://reviewboard.asterisk.org/r/1083/ + +2011-02-03 16:13 +0000 [r305988] Andrew Latham + + * phoneprov/snom-mac.xml (added), configs/phoneprov.conf.sample: + res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support + (issue #18713) Reported by: lathama Patches: snom_dir.diff + uploaded by lathama (license 1028) Tested by: lathama + +2011-02-03 00:29 +0000 [r305939] Richard Mudgett + + * main/channel.c, main/manager.c, /, channels/chan_sip.c, + apps/app_sendtext.c: Merged revisions 305923 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r305923 | rmudgett | 2011-02-02 18:24:40 -0600 + (Wed, 02 Feb 2011) | 24 lines Merged revisions 305889 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r305889 | rmudgett | 2011-02-02 18:15:07 -0600 + (Wed, 02 Feb 2011) | 17 lines Merged revisions 305888 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) + | 8 lines Minor AST_FRAME_TEXT related issues. * Include the null + terminator in the buffer length. When the frame is queued it is + copied. If the null terminator is not part of the frame buffer + length, the receiver could see garbage appended onto it. * Add + channel lock protection with ast_sendtext(). * Fixed AMI SendText + action ast_sendtext() return value check. ........ + ................ ................ + +2011-02-02 20:06 +0000 [r305845] Tilghman Lesher + + * /, funcs/func_env.c: Merged revisions 305844 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r305844 | tilghman | 2011-02-02 14:05:43 -0600 (Wed, 02 Feb 2011) + | 5 lines Eliminate a file descriptor leak when using the FILE() + dialplan function. (closes issue #18731) Reported by: marioabajo + ........ + +2011-02-02 19:30 +0000 [r305759-305843] Andrew Latham + + * configs/iax.conf.sample, funcs/func_enum.c, + configs/dundi.conf.sample, funcs/func_callcompletion.c, /, + configs/mgcp.conf.sample, configs/iaxprov.conf.sample, + configs/unistim.conf.sample, apps/app_externalivr.c, + configs/sip.conf.sample, configs/skinny.conf.sample, + configs/h323.conf.sample, configs/sla.conf.sample, + apps/app_voicemail.c: Replacing doc/* and asterisk.pdf with wiki + links Adding links to http(s)://wiki.asterisk.org + + * configs/chan_dahdi.conf.sample, /, configs/extconfig.conf.sample, + configs/res_snmp.conf.sample, main/ast_expr2f.c, + res/res_timing_dahdi.c, configs/ccss.conf.sample, + configs/sip.conf.sample, configs/skinny.conf.sample, + main/config.c, configs/h323.conf.sample, configs/sla.conf.sample, + main/ast_expr2.fl, res/res_srtp.c: Replacing doc/* with wiki + links Adding links to http(s)://wiki.asterisk.org + + * /, channels/chan_sip.c: Replace link to old doc with new wiki + page. Link to + https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions + +2011-02-01 22:48 +0000 [r305693] Jason Parker + + * /, channels/chan_iax2.c: Merged revisions 305692 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r305692 | qwell | 2011-02-01 16:48:16 -0600 (Tue, 01 Feb + 2011) | 7 lines Reverse sense of an error test when reading from + astdb. (closes issue #18545) Reported by: jcovert Patches: + chan_iax2.c.patch uploaded by jcovert (license 551) ........ + +2011-02-01 21:16 +0000 [r305650] Andrew Latham + + * configs/sip.conf.sample: SIP Configuration Documentation sip show + settings reports qualifyfreq in milliseconds. sip.conf configures + qualifyfreg in seconds. + +2011-02-01 19:27 +0000 [r305604] Brett Bryant + + * cel/cel_pgsql.c, /: Merged revisions 305603 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r305603 | bbryant | 2011-02-01 14:23:20 -0500 (Tue, 01 Feb 2011) + | 4 lines Add a possible solution to a customer problem with + reloading cel_pgsql.so quickly. ........ + +2011-02-01 18:03 +0000 [r305561] Andrew Latham + + * /: doc/tex dir removed, but corresponding entries still exists + Update README, CHANGES, and Makefile. Direct users to + http://wiki.asterisk.org for documentation or to the AST.txt and + AST.pdf included in the tarball. (closes issue #18443) Reported + by: bas Patches: changes.diff uploaded by lathama (license 1028) + readme.diff uploaded by lathama (license 1028) Tested by: lathama + bas + +2011-02-01 17:05 +0000 [r305474] Jason Parker + + * /, res/res_musiconhold.c: Merged revisions 305473 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r305473 | qwell | 2011-02-01 11:04:23 -0600 + (Tue, 01 Feb 2011) | 23 lines Merged revisions 305472 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r305472 | qwell | 2011-02-01 11:02:09 -0600 + (Tue, 01 Feb 2011) | 16 lines Merged revisions 305471 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb 2011) | + 9 lines Close file descriptor for timing source when a MOH class + gets destroyed. (closes issue #18457) Reported by: mcallist + Patches: 18457-closetimer.diff uploaded by qwell (license 4) + 18457-closetimer_trunk.diff uploaded by qwell (license 4) Tested + by: qwell, loloski ........ ................ ................ + +2011-02-01 16:05 +0000 [r305433] Brett Bryant + + * apps/app_confbridge.c: Add's two features to confbridge: + confbridge kick, and confbridge list. (closes issue #14389) + (closes issue #18007) Reported by: jcollie Patches: + 0001-Fix-up-bridging-module-so-that-menuselect-works.patch + uploaded by jcollie (license 412) + 0002-Add-confbridge-list-and-confbridge-kick-CLI-comm.patch + uploaded by jcollie (license 412) Tested by: file Review: + https://reviewboard.asterisk.org/r/1084/ + +2011-02-01 00:07 +0000 [r305344] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 305343 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r305343 | rmudgett | 2011-01-31 18:01:09 -0600 + (Mon, 31 Jan 2011) | 21 lines Merged revisions 305342 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r305342 | rmudgett | 2011-01-31 17:50:10 -0600 + (Mon, 31 Jan 2011) | 14 lines Merged revisions 305341 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) + | 7 lines Obtain the pri lock for PRI queue counters. Need to + obtain the pri lock when calling pri_dump_info_str() to avoid a + reentrancy problem when calculating the Q.921 Q count statistic. + JIRA AST-484 ........ ................ ................ + +2011-01-31 23:08 +0000 [r305132-305255] Jason Parker + + * apps/app_dial.c, /, channels/chan_sip.c: Merged revisions 305254 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r305254 | qwell | 2011-01-31 17:07:00 -0600 + (Mon, 31 Jan 2011) | 24 lines Merged revisions 305253 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r305253 | qwell | 2011-01-31 16:59:34 -0600 + (Mon, 31 Jan 2011) | 17 lines Merged revisions 305252 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | + 10 lines Prevent a crash when dialing a technology with no + destination (ex: Dial(SIP/)) chan_iax2 and other channel drivers + already had code to prevent this. The attempt that app_dial was + making to prevent it was not correct, so I fixed that. (closes + issue #18371) Reported by: gbour Patches: 18371.patch uploaded by + gbour (license 1162) ........ ................ ................ + + * main/tcptls.c, /, configs/sip.conf.sample: Merged revisions + 305247 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r305247 | qwell | 2011-01-31 16:25:23 -0600 (Mon, 31 Jan 2011) | + 7 lines Add alternative name for config option. The SIP sample + configuration had "tlscadir" as the option name, but chan_sip + used the more correct "tlscapath". Now both are accepted. + Discovered (sort of) by a user on IRC in #asterisk ........ + + * /, res/res_musiconhold.c: Merged revisions 305198 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r305198 | qwell | 2011-01-31 15:30:44 -0600 (Mon, 31 Jan + 2011) | 2 lines Fix compile error. pseudofd no longer exists. + ........ + + * /, res/res_musiconhold.c: Merged revisions 305131 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r305131 | qwell | 2011-01-31 15:00:25 -0600 + (Mon, 31 Jan 2011) | 16 lines Merged revisions 305130 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r305130 | qwell | 2011-01-31 14:59:37 -0600 + (Mon, 31 Jan 2011) | 9 lines Merged revisions 305129 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan + 2011) | 2 lines Set file descriptors to -1 on creation, so that + we don't see weirdness later. ........ ................ + ................ + +2011-01-31 13:57 +0000 [r305084] Andrew Latham + + * main/http.c: Asterisk HTTP response Content-type Address content + type for BSD and other platforms (closes issue #18456) Reported + by: alexo Patches: asterisk18_http.patch uploaded by alexo + (license 1175) Tested by: alexo + +2011-01-31 07:52 +0000 [r304951-305041] Tilghman Lesher + + * /, include/asterisk/lock.h: Merged revisions 305040 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r305040 | tilghman | 2011-01-31 01:51:40 -0600 (Mon, 31 + Jan 2011) | 2 lines Use the non-specific API aliases, to avoid a + problem with building the utils directory. ........ + + * /, apps/app_voicemail.c: Merged revisions 304985 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304985 | tilghman | 2011-01-31 01:27:13 -0600 + (Mon, 31 Jan 2011) | 16 lines Merged revisions 304978 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304978 | tilghman | 2011-01-31 01:25:14 -0600 + (Mon, 31 Jan 2011) | 9 lines Merged revisions 304952 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31 + Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined. + ........ ................ ................ + + * main/lock.c, /, main/heap.c, main/utils.c, + include/asterisk/lock.h, .cleancount: Merged revisions 304950 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r304950 | tilghman | 2011-01-31 00:41:36 -0600 (Mon, 31 Jan 2011) + | 18 lines Change mutex tracking so that it only consumes memory + in the core mutex object when it's actually being used. This + reduces the overall size of a mutex which was 3016 bytes before + this back down to 216 bytes (this is on 64-bit Linux with a + glibc-implemented mutex). The exactness of the numbers here may + vary slightly based upon how mutexes are implemented on a + platform, but the long and short of it is that prior to this + commit, chan_iax2 held down 98MB of memory on a 64-bit system for + nothing more than a table of 32767 locks. After this commit, the + same table occupies a mere 7MB of memory. (closes issue #18194) + Reported by: job Patches: 20110124__issue18194.diff.txt uploaded + by tilghman (license 14) Tested by: tilghman Review: + https://reviewboard.asterisk.org/r/1066 ........ + +2011-01-30 00:22 +0000 [r304913] Andrew Latham + + * funcs/func_callcompletion.c, /, apps/app_externalivr.c, + apps/app_queue.c, apps/app_voicemail.c, funcs/func_realtime.c, + res/res_calendar.c: Add Function and Application Relationships to + documentation Add and extend the see-also sections to the + documentation for applications and functions in an effort to + expand the online documentation of the wiki. Also check for and + update any links to moved documentation in the doc folder. + +2011-01-29 23:10 +0000 [r304639-304867] Sean Bright + + * /, res/res_config_ldap.c: Merged revisions 304866 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304866 | seanbright | 2011-01-29 18:07:18 -0500 + (Sat, 29 Jan 2011) | 14 lines Merged revisions 304865 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304865 | seanbright | 2011-01-29 18:05:25 -0500 (Sat, 29 Jan + 2011) | 7 lines Plug some memory leaks in the LDAP realtime + driver. (closes issue #18435) Reported by: zaltar Patches: + res_config_ldap.patch uploaded by zaltar (license 1148) ........ + ................ + + * /, apps/app_meetme.c: Merged revisions 304777 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304777 | seanbright | 2011-01-29 13:09:37 -0500 + (Sat, 29 Jan 2011) | 22 lines Merged revisions 304776 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan + 2011) | 15 lines If we fail to allocate our announcement objects, + make sure we don't leak objects. The majority of this patch was + committed already in r304726 and r304729. (issue #18225) Reported + by: kenji (issue #18444) Reported by: junky (closes issue #18343) + Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz + (license 834) ........ ................ + + * /, apps/app_meetme.c: Merged revisions 304774 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304774 | seanbright | 2011-01-29 12:54:43 -0500 + (Sat, 29 Jan 2011) | 16 lines Merged revisions 304773 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan + 2011) | 9 lines When we pass the S() or L() options to MeetMe, + make sure that we honor C as well. Without this patch, if the + user was kicked from the conference via the S() or L() mechanism, + we would just hang up on them even if we also passed C (continue + in dialplan when kicked). With this patch we honor the C flag in + those cases. (closes issue #17317) Reported by: var ........ + ................ + + * /, apps/app_meetme.c: Merged revisions 304730 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304730 | seanbright | 2011-01-29 12:15:27 -0500 + (Sat, 29 Jan 2011) | 22 lines Merged revisions 304729 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan + 2011) | 15 lines Make sure that we unref the correct object when + ejecting the most recent caller. Currently, when we kick the last + user to enter, we decrement our own reference count which results + in a crash when we kick another user or when we exit the + conference ourselves. This will fix #18225 in 1.8 and trunk, but + that particular bug does not exist in 1.6.2. (closes issue + #18225) Reported by: kenji Patches: issue18225.patch uploaded by + seanbright (license 71) Tested by: seanbright ........ + ................ + + * /, apps/app_meetme.c: Merged revisions 304727 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304727 | seanbright | 2011-01-29 11:28:27 -0500 + (Sat, 29 Jan 2011) | 16 lines Merged revisions 304726 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan + 2011) | 9 lines Fix user reference leak in MeetMe. We were + unlinking the user from the conferences user container, but not + decrementing the reference count of the user as well, resulting + in a leak. (closes issue #18444) Reported by: junky Tested by: + seanbright ........ ................ + + * /, apps/app_meetme.c: Merged revisions 304683 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304683 | seanbright | 2011-01-28 17:54:23 -0500 + (Fri, 28 Jan 2011) | 16 lines Merged revisions 304659,304682 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri, 28 Jan + 2011) | 5 lines Don't leak references if we can't create a pseudo + channel for mixing in MeetMe. If there was a problem allocating a + pseudo channel when building our meetme, we weren't destroying + our user container or destroying the mutexes that we created. + ........ r304682 | seanbright | 2011-01-28 17:38:05 -0500 (Fri, + 28 Jan 2011) | 2 lines Revert part of the previous commit that + snuck in. ........ ................ + + * /, main/acl.c: Merged revisions 304638 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r304638 | seanbright | 2011-01-28 15:19:08 -0500 (Fri, 28 Jan + 2011) | 11 lines Restore some conditionals that we lost in + r277814. There are some cases where ast_append_ha() is called + with a NULL instead of a valid int pointer. So if we get a NULL, + don't try to dereference it. (closes issue #18162) Reported by: + imcdona Patches: issue0018162.patch uploaded by pabelanger + (license 224) Tested by: enegaard ........ + +2011-01-27 20:09 +0000 [r304600] Brett Bryant + + * res/res_config_pgsql.c: Patch that fixes the "realtime show pgsql + cache" command crash when giving a table name, because of the use + of an uninitialized variable. Fixes an error introduced in + r300882. (closes issue #18605) Reported by: romain_proformatique + Patches: res_config_pgsql_fix.patch uploaded by romain + proformatique (license 975) Tested by: romain_proformatique + +2011-01-27 20:07 +0000 [r304599] Kevin P. Fleming + + * res/res_fax.c: Fix bug with 'F' option for ReceiveFAX and + SendFAX. Skipping the call to set_t38_fax_caps() caused the FAX + session details to not be marked as supporting audio FAX + either... the function's name is a bit misleading. This patch + restores the single bit of non-T.38 behavior from that function + when audio mode is forced. + +2011-01-27 19:12 +0000 [r304555] Richard Mudgett + + * /, main/ccss.c: Merged revisions 304554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r304554 | rmudgett | 2011-01-27 13:08:14 -0600 (Thu, 27 Jan 2011) + | 4 lines Warning message if CALLCOMPLETION(cc_callback_macro or + cc_agent_dialstring) are empty. Test if the value pointer is not + NULL instead of not ast_strlen_zero(). ........ + +2011-01-27 17:03 +0000 [r304463-304467] Jason Parker + + * /, configure, configure.ac: Merged revisions 304466 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304466 | qwell | 2011-01-27 11:03:01 -0600 + (Thu, 27 Jan 2011) | 23 lines Merged revisions 304465 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304465 | qwell | 2011-01-27 11:01:24 -0600 + (Thu, 27 Jan 2011) | 16 lines Merged revisions 304464 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) | + 9 lines Fix default prefix=/usr regression on non-Linux systems. + This partially reverts a change made in branches/1.4/ r267759, + which will cause issue #17013 to be reopened. This issue was + pointed out by a user on #asterisk, who helpfully discovered that + paths were being set incorrectly. To truly understand what was + wrong, one should run: svn diff --force -c + configure ........ ................ ................ + + * /, configure: Merged revisions 304462 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304462 | qwell | 2011-01-27 10:48:44 -0600 + (Thu, 27 Jan 2011) | 16 lines Merged revisions 304461 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304461 | qwell | 2011-01-27 10:48:00 -0600 + (Thu, 27 Jan 2011) | 9 lines Merged revisions 304460 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan + 2011) | 1 line Rerun bootstrap.sh with no changes, so that it is + more obvious what my next commit changes. ........ + ................ ................ + +2011-01-27 15:57 +0000 [r304422] Kevin P. Fleming + + * res/res_fax.c: Rename the SendFAX/ReceiveFAX 'force audio' + option. The recently added option to disable usage of T.38 for a + single session should have been named 'F' for 'force audio', + since that is really what the user is asking to happen (and it's + a positive option instead of a negative option that way). + +2011-01-27 00:06 +0000 [r304385] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, channels/sig_pri.c: Merged from + revision 304341 + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r304341 | rmudgett | 2011-01-26 16:38:39 -0600 (Wed, + 26 Jan 2011) | 7 lines Add connected line chan_dahdi.conf + pricpndialplan option. * Added from_channel value to + prilocaldialplan option. JIRA ABE-2731 JIRA SWP-2842 .......... + +2011-01-26 23:41 +0000 [r304384] Jeff Peeler + + * apps/app_followme.c: Add option to followme to delay answer until + ready to bridge call. Followme answers an incoming call if it + hasn't already been answered and starts MOH. Some poorly designed + autodialers see the answer and start playing their message to the + hold music. The 'N' option has been added to indicate ringing and + not answer until the call is accepted. (closes issue #18479) + Reported by: ianc Patches: trunk_followme.diff uploaded by ianc + (license 998) + +2011-01-26 22:39 +0000 [r304342] Kevin P. Fleming + + * res/res_fax.c: Add ability to disable T.38 usage for specific + SendFAX/ReceiveFAX sessions. Sometimes during troubleshooting it + can be useful to disable T.38 usage in order to narrow down a + problem. This patch adds an 'n' option to SendFAX and ReceiveFAX + so that can be done without having to disable T.38 usage entirely + for the peer that Asterisk is communicating with. (inspired by + trying to assist Bryant Zimmerman on asterisk-users) + +2011-01-26 22:27 +0000 [r304340] Jeff Peeler + + * /, main/features.c: Merged revisions 304339 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304339 | jpeeler | 2011-01-26 16:27:30 -0600 + (Wed, 26 Jan 2011) | 9 lines Merged revisions 304338 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r304338 | jpeeler | 2011-01-26 16:26:37 -0600 (Wed, 26 + Jan 2011) | 2 lines Change delimiter used internally for + GOTO_ON_BLINDXFR to commas to match 76703. ........ + ................ + +2011-01-26 21:03 +0000 [r304252] Mark Michelson + + * main/udptl.c, /: Merged revisions 304250 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304250 | mmichelson | 2011-01-26 15:02:10 -0600 + (Wed, 26 Jan 2011) | 9 lines Merged revisions 304242 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed, + 26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in + ast_udptl ........ ................ + +2011-01-26 20:44 +0000 [r304246] Matthew Nicholson + + * main/netsock2.c, /, channels/chan_sip.c, + channels/sip/reqresp_parser.c, include/asterisk/netsock2.h, + channels/sip/include/sip.h, + channels/sip/include/reqresp_parser.h: Merged revisions 304245 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304245 | mnicholson | 2011-01-26 14:43:27 -0600 + (Wed, 26 Jan 2011) | 20 lines Merged revisions 304244 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304244 | mnicholson | 2011-01-26 14:42:16 -0600 + (Wed, 26 Jan 2011) | 13 lines Merged revisions 304241 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan + 2011) | 6 lines This patch modifies chan_sip to route responses + to the address the request came from. It also modifies chan_sip + to respect the maddr parameter in the Via header. ABE-2664 + Review: https://reviewboard.asterisk.org/r/1059/ ........ + ................ ................ + +2011-01-26 20:25 +0000 [r304195] Sean Bright + + * /, configs/queues.conf.sample: Merged revisions 304186 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304186 | seanbright | 2011-01-26 15:23:48 -0500 + (Wed, 26 Jan 2011) | 16 lines Merged revisions 304181 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304181 | seanbright | 2011-01-26 15:22:47 -0500 + (Wed, 26 Jan 2011) | 9 lines Merged revisions 304159 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed, + 26 Jan 2011) | 1 line Make sure the sample queues.conf is + properly commented. ........ ................ ................ + +2011-01-26 19:58 +0000 [r304152] Matthew Nicholson + + * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions + 303907 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r303907 | mnicholson | 2011-01-25 14:56:12 -0600 (Tue, 25 Jan + 2011) | 2 lines Reimplemented fax session reservation to reverse + the ABI breakage introduced in r297486. ........ + +2011-01-26 19:40 +0000 [r304151] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 304150 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304150 | rmudgett | 2011-01-26 13:39:35 -0600 + (Wed, 26 Jan 2011) | 16 lines Merged revisions 304149 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304149 | rmudgett | 2011-01-26 13:38:38 -0600 + (Wed, 26 Jan 2011) | 9 lines Merged revisions 304148 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed, + 26 Jan 2011) | 2 lines Update documentation for + DAHDISendCallreroutingFacility() application. .......... + ................ ................ + +2011-01-26 01:27 +0000 [r304098] Sean Bright + + * /, main/file.c: Merged revisions 304097 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304097 | seanbright | 2011-01-25 20:26:26 -0500 + (Tue, 25 Jan 2011) | 19 lines Merged revisions 304096 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan + 2011) | 12 lines Per the man page, setvbuf() must be called + before any other operation on an open file. We use setvbuf() to + associate a buffer with a stream, but we have already written to + the open file. This works (by chance) on Linux, but fails on + other platforms, such as OpenSolaris. (closes issue #16610) + Reported by: bklang Patches: setvbuf.patch uploaded by crjw + (license 963) Tested by: bklang, asgaroth, efutch ........ + ................ + +2011-01-25 23:31 +0000 [r304008] Richard Mudgett + + * /, main/features.c: Merged revisions 304007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r304007 | rmudgett | 2011-01-25 17:28:25 -0600 + (Tue, 25 Jan 2011) | 22 lines Merged revisions 304006 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r304006 | rmudgett | 2011-01-25 17:25:32 -0600 + (Tue, 25 Jan 2011) | 15 lines Merged revisions 304005 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011) + | 8 lines DTMF attended transfers sometimes fail for no apparent + reason. The loop in feature_request_and_dial() can exit when + Party C has answered without processing an AST_CONTROL_ANSWER. + Also sometimes an AST_CONTROL_ANSWER never happens even though + Party C has answered. Don't hangup Party C if he is up or we + receive an AST_CONTROL_ANSWER. ........ ................ + ................ + +2011-01-25 22:15 +0000 [r303963] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 303962 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303962 | twilson | 2011-01-25 16:09:01 -0600 + (Tue, 25 Jan 2011) | 30 lines Merged revisions 303960 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303960 | twilson | 2011-01-25 16:02:42 -0600 + (Tue, 25 Jan 2011) | 23 lines Merged revisions 303906 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) + | 16 lines Guard against retransmitting BYEs indefinitely In the + case of an attended transfer (A calls B, A atxfers to C) where A + becomes unreachable before replying to Asterisk's BYE, Asterisk + can sometimes retransmit the BYE indefinitely. This is because + __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0], + SIP_ALREADYGONE and will then transmit a BYE. When this BYE times + out, it will not ever be marked as ALREADYGONE, so when + __sip_autodestruct is called again, we end up starting the cycle + over. This patch adds a call to sip_alreadygone(pkt->owner) in + retrans_pkt in the case of a BYE that has timed out. This should + prevent Asterisk from trying to transmit new BYE messages in the + future. Review: https://reviewboard.asterisk.org/r/1077/ ........ + ................ ................ + +2011-01-25 18:56 +0000 [r303861] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 303860 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303860 | tilghman | 2011-01-25 12:55:27 -0600 + (Tue, 25 Jan 2011) | 12 lines Merged revisions 303858 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r303858 | tilghman | 2011-01-25 12:41:26 -0600 (Tue, 25 Jan 2011) + | 5 lines Fix "sip show user ", so that it actually shows + results, instead of just completing the last entry. (closes issue + #16675) Reported by: pj ........ ................ + +2011-01-25 17:58 +0000 [r303772] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_ss7.h, /, + channels/sig_pri.c, channels/sig_ss7.c: Merged revisions 303771 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303771 | rmudgett | 2011-01-25 11:49:20 -0600 + (Tue, 25 Jan 2011) | 54 lines Merged revisions 303769 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303769 | rmudgett | 2011-01-25 11:42:42 -0600 + (Tue, 25 Jan 2011) | 47 lines Merged revisions 303765 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) + | 40 lines Sending out unnecessary PROCEEDING messages breaks + overlap dialing. Issue #16789 was a good idea. Unfortunately, it + breaks overlap dialing through Asterisk. There is not enough + information available at this point to know if dialing is + complete. The ast_exists_extension(), ast_matchmore_extension(), + and ast_canmatch_extension() calls are not adequate to detect a + dial through extension pattern of "_9!". Workaround is to use the + dialplan Proceeding() application early in non-dial through + extensions. * Effectively revert issue #16789. * Allow outgoing + overlap dialing to hear dialtone and other early media. A + PROGRESS "inband-information is now available" message is now + sent after the SETUP_ACKNOWLEDGE message for non-digital calls. + An AST_CONTROL_PROGRESS is now generated for incoming + SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of + the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent + with the cause codes. * Added better protection from sending out + of sequence messages by combining several flags into a single + enum value representing call progress level. * Added diagnostic + messages for deferred overlap digits handling corner cases. + (closes issue #17085) Reported by: shawkris (closes issue #18509) + Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch + uploaded by rmudgett (license 664) Expanded upon + issue18509_early_media_v1.8_v3.patch to include analog and SS7 + because of backporting requirements. Tested by: wimpy, rmudgett + ........ ................ ................ + +2011-01-25 17:05 +0000 [r303679] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 303678 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303678 | jpeeler | 2011-01-25 11:02:38 -0600 + (Tue, 25 Jan 2011) | 33 lines Merged revisions 303677 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303677 | jpeeler | 2011-01-25 10:59:28 -0600 + (Tue, 25 Jan 2011) | 26 lines Merged revisions 303676 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) + | 20 lines Fix voicemail sequencing for file based storage. A + previous change was made to account for when the number of + voicemail messages exceeds the max limit to be handled properly, + but it caused gaps in the messages to not be properly handled. + This has now been resolved. In later non 1.4 branches, it appears + that resequencing wasn't even occurring due from what appears and + accidental code removal. (closes issue #18498) Reported by: + JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license + 325) (closes issue #18486) Reported by: bluefox Patches: + bug18486.patch uploaded by jpeeler (license 325) ........ + ................ ................ + +2011-01-25 15:52 +0000 [r303638] Matthew Nicholson + + * main/utils.c: Use unsigned char in comparison for UTF8 check to + quiet a compiler warning. + +2011-01-24 20:57 +0000 [r303547-303551] Russell Bryant + + * main/channel.c, main/pbx.c, /, apps/app_meetme.c, + main/features.c, include/asterisk/channel.h: Merged revisions + 303549 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303549 | russell | 2011-01-24 14:51:37 -0600 + (Mon, 24 Jan 2011) | 45 lines Merged revisions 303548 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303548 | russell | 2011-01-24 14:49:53 -0600 + (Mon, 24 Jan 2011) | 38 lines Merged revisions 303546 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) + | 31 lines Fix channel redirect out of MeetMe() and other issues + with channel softhangup. Mantis issue #18585 reports that a + channel redirect out of MeetMe() stopped working properly. This + issue includes a patch that resolves the issue by removing a call + to ast_check_hangup() from app_meetme.c. I left that in my patch, + as it doesn't need to be there. However, the rest of the patch + fixes this problem with or without the change to app_meetme. The + key difference between what happens before and after this patch + is the effect of the END_OF_Q control frame. After END_OF_Q is + hit in ast_read(), ast_read() will return NULL. With the + ast_check_hangup() removed, app_meetme sees this which causes it + to exit as intended. Checking ast_check_hangup() caused + app_meetme to exit earlier in the process, and the target of the + redirect saw the condition where ast_read() returned NULL. + Removing ast_check_hangup() works around the issue in app_meetme, + but doesn't solve the issue if another application did the same + thing. There are also other edge cases where if an application + finishes at the same time that a redirect happens, the target of + the redirect will think that the channel hung up. So, I made some + changes in pbx.c to resolve it at a deeper level. There are + already places that unset the SOFTHANGUP_ASYNCGOTO flag in an + attempt to abort the hangup process. My patch extends this to + remove the END_OF_Q frame from the channel's read queue, making + the "abort hangup" more complete. This same technique was used in + every place where a softhangup flag was cleared. (closes issue + #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: + https://reviewboard.asterisk.org/r/1082/ ........ + ................ ................ + + * contrib/scripts/install_prereq: Add gsm-devel as a package to + install on redhat based systems. + +2011-01-24 18:59 +0000 [r303509] Matthew Nicholson + + * res/res_config_curl.c, include/asterisk/utils.h, + funcs/func_curl.c, channels/chan_sip.c, tests/test_utils.c, + res/res_agi.c, channels/sip/reqresp_parser.c, main/http.c, + main/utils.c, funcs/func_uri.c: According to section 19.1.2 of + RFC 3261: For each component, the set of valid BNF expansions + defines exactly which characters may appear unescaped. All other + characters MUST be escaped. This patch modifies ast_uri_encode() + to encode strings in line with this recommendation. This patch + also adds an ast_escape_quoted() function which escapes '"' and + '\' characters in quoted strings in accordance with section 25.1 + of RFC 3261. The ast_uri_encode() function has also been modified + to take an ast_flags struct describing the set of rules it should + use when escaping characters to allow for it to escape SIP URIs + in addition to HTTP URIs and other types of URIs or variations of + those two URI types in the future. The ast_uri_decode() function + has also been modified to accept an ast_flags struct describing + the set of rules to use when decoding to enable decoding '+' as ' + ' in legacy http URLs. The unit tests for these functions have + also been updated. ABE-2705 Review: + https://reviewboard.asterisk.org/r/1081/ + +2011-01-24 17:21 +0000 [r303468] Jason Parker + + * channels/chan_dahdi.c, /: Merged revisions 303467 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303467 | qwell | 2011-01-24 11:20:03 -0600 + (Mon, 24 Jan 2011) | 22 lines Merged revisions 303285 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 + (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | + 8 lines Reset configuration before parsing users.conf. Some + values configured in chan_dahdi.conf were able to leak in to + users.conf configuration. This was surprising users, and + potentially setting non-sane "defaults". ASTNOW-125 ........ + ................ ................ + +2011-01-22 04:13 +0000 [r303418] Russell Bryant + + * configure, configure.ac: Revert default compiler change. If + someone wishes to do so, it is trivial to set your own default + when running the configure script. + +2011-01-21 23:11 +0000 [r303288-303376] Jason Parker + + * channels/chan_dahdi.c, /: Temporarily revert r303288 + + * channels/chan_dahdi.c, /: Merged revisions 303286 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303286 | qwell | 2011-01-21 15:50:11 -0600 + (Fri, 21 Jan 2011) | 22 lines Merged revisions 303285 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303285 | qwell | 2011-01-21 15:48:09 -0600 + (Fri, 21 Jan 2011) | 15 lines Merged revisions 303284 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | + 8 lines Reset configuration before parsing users.conf. Some + values configured in chan_dahdi.conf were able to leak in to + users.conf configuration. This was surprising users, and + potentially setting non-sane "defaults". ASTNOW-125 ........ + ................ ................ + +2011-01-21 09:09 +0000 [r303198-303235] Tilghman Lesher + + * configure, configure.ac: Really use llvm-gcc, when available. + + * funcs/func_db.c, CHANGES: Add DB_KEYS. Discussion on #asterisk on + 2011-01-19: (02:07:03 PM) boch: i wonder how to cycle all entries + in a tree (02:07:11 PM) leifmadsen: use While() (02:07:17 PM) + leifmadsen: you need to know the tree structure already though + (02:07:36 PM) boch: what you mean? (02:09:02 PM) leifmadsen: you + need to know the structure prior to looping, because you can't + just return the structure from the dialplan (02:09:43 PM) + leifmadsen: the only way I can think of doing that is via + something like writing the output of: asterisk -rx "database + show" to a file, then looping through that to know the structure + of the database and check everything (02:09:59 PM) leifmadsen: + but at that point you're better off just using either a + relational database or an external script (02:10:13 PM) boch: for + example i need to know all entries in the tree (02:10:15 PM) + boch: got it (02:10:20 PM) leifmadsen: exactly (02:10:22 PM) + leifmadsen: that's the problem (02:10:22 PM) boch: thank you + (02:13:09 PM) mateu: yeah, i'm surprised there isn't something + from the dialplan like 'database show family' so one can get all + keys in a family to loop over. (02:15:35 PM) leifmadsen: database + shows everything (02:16:22 PM) mateu: i mean something from the + dial plan that mimics 'database show ' (02:16:41 PM) + leifmadsen: guess no one has found that important enough to + program :) (02:16:52 PM) leifmadsen: at that point you should + probably just use a relational database... (02:17:10 PM) mateu: i + dunno (02:17:16 PM) mateu: seems pretty basic to me. (02:17:16 + PM) leifmadsen: me either (02:17:19 PM) leifmadsen: sure does + (02:17:24 PM) leifmadsen: no one has programmed it though + (02:17:28 PM) ***leifmadsen shrugs (02:17:43 PM) mateu: ok, well + at least we know how it currently stands. thanks leifmadsen + (02:28:52 PM) Corydon76-home: leifmadsen: something like + HASHKEYS() ? (02:30:11 PM) leifmadsen: Corydon76-home: ummm, I + was thinking more like DUNDI_QUERY() and DUNDI_RESULT() (02:30:31 + PM) leifmadsen: although HASHKEYS() might work (02:30:58 PM) + leifmadsen: actually ya, looking at it, similar to HASHKEYS() + (02:31:01 PM) leifmadsen: DBKEYS() I guess? (02:31:45 PM) + Corydon76-home: So with no argument, retrieves families, with an + argument, retrieves keys of that family? (02:34:02 PM) + leifmadsen: ya (02:34:16 PM) leifmadsen: how would you iterate + through layers of them? (02:34:30 PM) leifmadsen: i.e. + family/key/key/key ? (02:34:43 PM) Corydon76-home: Essentially, + yes + +2011-01-20 20:35 +0000 [r303154] Richard Mudgett + + * /, main/ccss.c: Merged revisions 303153 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303153 | rmudgett | 2011-01-20 14:31:20 -0600 + (Thu, 20 Jan 2011) | 22 lines Merged revision 303098 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu, + 20 Jan 2011) | 15 lines CC_INTERFACES does not get built + correctly with local channels. If local channels are used with + CCSS, CC_INTERFACES gets garbage prepended to it so the CC recall + fails. Also CC_INTERFACES gets "&(null)" appended to it. * + Initialize the buffer to eliminate the prepended garbage. * + Filter out the empty interface strings to eliminate the latter. * + Added a diagnostic message if the CC_INTERFACES is ever empty. + JIRA ABE-2740 JIRA SWP-2848 .......... ................ + +2011-01-20 19:58 +0000 [r303108] Shaun Ruffell + + * /, main/features.c: Merged revisions 303107 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303107 | sruffell | 2011-01-20 13:57:31 -0600 + (Thu, 20 Jan 2011) | 23 lines Merged revisions 303106 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011) + | 15 lines main/features: Use POLLPRI when waiting for events on + parked channels. This change resolves a regression in the 1.6.2 + when converting from select to poll. The DAHDI timers use POLLPRI + to indicate that the timer fired, but features was not waiting + for that flag. The result was no audio for MOH when a call was + parked and res_timing_dahdi was in use. This patch is slightly + modified from the one on the mantis issue. It does not set an + exception on the channel if the POLLPRI flag is set. (closes + issue #18262) Reported by: francesco_r Patches: + patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029) + Tested by: francesco_r, rfrantik, one47 ........ ................ + +2011-01-20 17:14 +0000 [r303011] Jeff Peeler + + * /, configs/queues.conf.sample, apps/app_queue.c: Merged revisions + 303009 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r303009 | jpeeler | 2011-01-20 11:10:32 -0600 + (Thu, 20 Jan 2011) | 21 lines Merged revisions 303008 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r303008 | jpeeler | 2011-01-20 11:07:44 -0600 + (Thu, 20 Jan 2011) | 14 lines Merged revisions 303007 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) + | 8 lines Add new queue strategy to preserve behavior for when + queue members moved to ao2. Add queue strategy called "rrordered" + to mimic old behavior from when queue members were stored in a + linked list. ABE-2707 ........ ................ ................ + +2011-01-20 16:12 +0000 [r302922] Russell Bryant + + * /, apps/app_privacy.c: Merged revisions 302921 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302921 | russell | 2011-01-20 10:12:15 -0600 + (Thu, 20 Jan 2011) | 9 lines Merged revisions 302920 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 + Jan 2011) | 2 lines Resolve a compiler warning. ........ + ................ + +2011-01-20 15:46 +0000 [r302919] Leif Madsen + + * apps/app_dial.c, /: Merged revisions 302918 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302918 | lmadsen | 2011-01-20 09:45:39 -0600 + (Thu, 20 Jan 2011) | 16 lines Merged revisions 302917 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) + | 8 lines Option L() is milliseconds, not seconds. > Change the + verbose output of option L() to say milliseconds and not seconds + > as the value is in milliseconds. > > (closes issue #18264) > + Reported by: jacco > Patches: > app_dial_patch.txt uploaded by + lmadsen (license 10) ........ ................ + +2011-01-20 09:07 +0000 [r302879] Tilghman Lesher + + * configure, configure.ac: On systems which have LLVM, use that + compiler. Should result in a massive speed increase. + +2011-01-19 23:57 +0000 [r302838] Russell Bryant + + * main/manager.c, /: Merged revisions 302837 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302837 | russell | 2011-01-19 17:56:48 -0600 (Wed, 19 Jan 2011) + | 2 lines Only check container count if it exists. ........ + +2011-01-19 23:53 +0000 [r302835-302836] Sean Bright + + * main/config.c: Clarify a source comment about configuration + template categories. (closes issue #18578) Reported by: astmiv + Patches: asterisk.main.config.2.patch uploaded by astmiv (license + 1189) + + * /, apps/app_voicemail.c: Merged revisions 302834 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302834 | seanbright | 2011-01-19 18:49:00 -0500 + (Wed, 19 Jan 2011) | 14 lines Merged revisions 302833 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed, 19 Jan + 2011) | 7 lines Support greetingsfolder as documented in + voicemail.conf.sample. (closes issue #17870) Reported by: + edhorton Patches: + __20100816-app_voicemail-greetingsfolder-support.txt uploaded by + lmadsen (license 10) ........ ................ + +2011-01-19 23:33 +0000 [r302832] Paul Belanger + + * /, contrib/scripts/install_prereq: Merged revisions 302831 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302831 | pabelanger | 2011-01-19 18:29:45 -0500 (Wed, 19 Jan + 2011) | 2 lines Add binutils-dev for BETTER_BACKTRACES ........ + +2011-01-19 23:07 +0000 [r302786-302790] Russell Bryant + + * main/manager.c, /: Merged revisions 302789 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302789 | russell | 2011-01-19 17:06:46 -0600 + (Wed, 19 Jan 2011) | 11 lines Merged revisions 302788 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011) + | 4 lines Turn a noisy verbose message into a debug message. This + can drown your console if you're using the AMI over HTTP. + ........ ................ + + * main/manager.c, /: Merged revisions 302785 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) + | 15 lines Resolve a memory leak with the manager interface is + disabled. The intent of this check as it stands in previous + versions of Asterisk was to check if there are any active + sessions. If there were no sessions, then the function would + return immediately and not bother with queueing up the manager + event to be processed. Since the conversion of storing sessions + in an astobj2 container, this check will always pass. I changed + it to go back to checking what was intended. The side effect of + this was that if the AMI is disabled, the manager event queue is + populated anyway, but the code that runs to clear out the queue + never runs. A producer with no consumer is a bad thing. Reported + internally by kmorgan. ........ + +2011-01-19 21:35 +0000 [r302732] Richard Mudgett + + * /, main/features.c: Merged revisions 302713 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302713 | rmudgett | 2011-01-19 15:29:22 -0600 + (Wed, 19 Jan 2011) | 29 lines Merged revisions 302693 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r302693 | rmudgett | 2011-01-19 15:25:41 -0600 + (Wed, 19 Jan 2011) | 22 lines Merged revisions 302671 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) + | 15 lines DTMF transfer plays the wrong sounds for wrong number + or other call failure. * Set the default for features.conf.sample + xferfailsound option to "beeperr" as documented instead of + "pbx-invalid" and corrected the use of it in DTMF blind transfer + (#1). * Improved DTMF blind transfer handling of wrong numbers. + Most of the concerns in this issue were taken care of by the + patch for issue 17999: Issues with DTMF triggered attended + transfers. (closes issue #18379) Reported by: gincantalupo Tested + by: rmudgett ........ ................ ................ + +2011-01-19 21:24 +0000 [r302644-302686] Tilghman Lesher + + * /, include/asterisk/astdb.h: Merged revisions 302680 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302680 | tilghman | 2011-01-19 15:23:31 -0600 + (Wed, 19 Jan 2011) | 16 lines Merged revisions 302675 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r302675 | tilghman | 2011-01-19 15:22:45 -0600 + (Wed, 19 Jan 2011) | 9 lines Merged revisions 302663 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r302663 | tilghman | 2011-01-19 15:20:28 -0600 (Wed, 19 + Jan 2011) | 2 lines Add some API documentation ........ + ................ ................ + + * /, main/app.c: Merged revisions 302634 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302634 | tilghman | 2011-01-19 14:24:57 -0600 + (Wed, 19 Jan 2011) | 22 lines Merged revisions 302599 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011) + | 15 lines Kill zombies. When we ast_safe_fork() with a non-zero + argument, we're expected to reap our own zombies. On a zero + argument, however, the zombies are only reaped when there aren't + any non-zero forked children alive. At other times, we accumulate + zombies. This code is forward ported from res_agi in 1.4, so that + forked children are always reaped, thus preventing an + accumulation of zombie processes. (closes issue #18515) Reported + by: ernied Patches: 20101221__issue18515.diff.txt uploaded by + tilghman (license 14) Tested by: ernied ........ ................ + +2011-01-19 20:15 +0000 [r302601] Jason Parker + + * /, res/res_fax.c: Merged revisions 302600 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302600 | qwell | 2011-01-19 14:14:40 -0600 (Wed, 19 Jan 2011) | + 1 line Fix typo pointed out on asterisk-users list. ........ + +2011-01-19 19:04 +0000 [r302507-302556] Sean Bright + + * /, main/utils.c: Merged revisions 302555 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302555 | seanbright | 2011-01-19 14:03:32 -0500 + (Wed, 19 Jan 2011) | 14 lines Merged revisions 302554 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302554 | seanbright | 2011-01-19 14:02:29 -0500 (Wed, 19 Jan + 2011) | 7 lines Don't call strlen() when we only need to look at + the next character or two. (closes issue #18042) Reported by: + wdoekes Patches: astsvn-inefficient-ast-uri-decode.patch uploaded + by wdoekes (license 717) ........ ................ + + * /, main/features.c: Merged revisions 302552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302552 | seanbright | 2011-01-19 13:54:47 -0500 + (Wed, 19 Jan 2011) | 14 lines Merged revisions 302551 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302551 | seanbright | 2011-01-19 13:54:03 -0500 (Wed, 19 Jan + 2011) | 7 lines Remove an extraneous \r\n at the end of a parking + manager events. (closes issue #18363) Reported by: + clegall_proformatique Patches: + asterisk_1.8_295998_parking_manager_events_format.patch uploaded + by clegall proformatique (license 1139) ........ ................ + + * /, res/res_agi.c: Merged revisions 302549 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302549 | seanbright | 2011-01-19 13:43:11 -0500 + (Wed, 19 Jan 2011) | 17 lines Merged revisions 302548 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan + 2011) | 10 lines Properly handle partial reads from fgets() when + handling AGIs. When fgets() failed with EAGAIN, we were + continually decrementing the available space left in our buffer, + resulting in botched command handling. (closes issue #16032) + Reported by: notahat Patches: agi_buffer_patch2.diff uploaded by + fnordian (license 110) ........ ................ + + * /, main/utils.c: Merged revisions 302505 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302505 | seanbright | 2011-01-19 12:58:11 -0500 + (Wed, 19 Jan 2011) | 14 lines Merged revisions 302504 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302504 | seanbright | 2011-01-19 12:56:32 -0500 (Wed, 19 Jan + 2011) | 7 lines Make sure that h_length is set when we + short-circuit out of ast_gethostbyname. (closes issue #16135) + Reported by: thedavidfactor Patches: utils.patch uploaded by + thedavidfactor (license 903) ........ ................ + +2011-01-19 17:15 +0000 [r302463] Paul Belanger + + * /, res/res_timing_timerfd.c: Merged revisions 302462 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302462 | pabelanger | 2011-01-19 12:09:35 -0500 + (Wed, 19 Jan 2011) | 9 lines Merged revisions 302461 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r302461 | pabelanger | 2011-01-19 12:08:01 -0500 (Wed, + 19 Jan 2011) | 2 lines Handle 'Resource temporarily unavailable' + error more gracefully. ........ ................ + +2011-01-19 15:54 +0000 [r302413-302418] Sean Bright + + * /, configs/extensions.conf.sample: Merged revisions 302417 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302417 | seanbright | 2011-01-19 10:53:20 -0500 + (Wed, 19 Jan 2011) | 16 lines Merged revisions 302416 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302416 | seanbright | 2011-01-19 10:52:44 -0500 (Wed, 19 Jan + 2011) | 9 lines Remove references to priorityjumping from the + sample extensions.conf. Priority jumping was removed from + pbx_config in r68970. (closes issue #18622) Reported by: kshumard + Patches: extensions.conf.sample.patch uploaded by kshumard + (license 92) ........ ................ + + * /, channels/chan_sip.c: Merged revisions 302414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302414 | seanbright | 2011-01-19 10:45:17 -0500 (Wed, 19 Jan + 2011) | 7 lines Initialize an uninitialized variable. (closes + issue #18640) Reported by: jcovert Patches: chan_sip.c.patch + uploaded by jcovert (license 551) ........ + + * channels/chan_local.c, /: Merged revisions 302412 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r302412 | seanbright | 2011-01-19 10:31:39 -0500 (Wed, + 19 Jan 2011) | 10 lines Use appropriate type for requested format + in chan_local. We were passing and storing the requested format + as an int instead of format_t resulting in truncation. (closes + issue #18238) Reported by: whizemen Patches: + 0018238_speex16.patch uploaded by whizemen (license 1143) + ........ + +2011-01-18 22:06 +0000 [r302319] Richard Mudgett + + * /, main/features.c: Merged revisions 302318 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302318 | rmudgett | 2011-01-18 16:04:14 -0600 (Tue, 18 Jan 2011) + | 1 line Use the expanded format type instead of plain int. + ........ + +2011-01-18 21:44 +0000 [r302315] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 302314 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302314 | mnicholson | 2011-01-18 15:43:21 -0600 + (Tue, 18 Jan 2011) | 18 lines Merged revisions 302313 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r302313 | mnicholson | 2011-01-18 15:40:03 -0600 + (Tue, 18 Jan 2011) | 11 lines Merged revisions 302311 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r302311 | mnicholson | 2011-01-18 15:35:03 -0600 (Tue, 18 Jan + 2011) | 4 lines URI encode the user part of the contact header. + ABE-2705 ........ ................ ................ + +2011-01-18 20:40 +0000 [r302270] Jeff Peeler + + * main/pbx.c, /: Merged revisions 302266 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302266 | jpeeler | 2011-01-18 14:19:57 -0600 + (Tue, 18 Jan 2011) | 34 lines Merged revisions 302265 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) + | 27 lines Convert device state callbacks to ao2 objects to fix a + deadlock in chan_sip. Lock scenario presented here: Thread 1 + holds ast_rdlock_contexts &conlock holds handle_statechange hints + holds handle_statechange hint waiting for cb_extensionstate + Locked Here: chan_sip.c line 7428 (find_call) Thread 2 holds + handle_request_do &netlock holds find_call sip_pvt_ptr waiting + for ast_rdlock_contexts &conlock Locked Here: pbx.c line 9911 + (ast_rdlock_contexts) Chan_sip has an established locking order + of locking the sip_pvt and then getting the context lock. So the + as stated by the summary, the operations in thread 2 have been + modified to no longer require the context lock. (closes issue + #18310) Reported by: one47 Patches: statecbs_ao2.mk2.patch + uploaded by one47 (license 23), modified by me Review: + https://reviewboard.asterisk.org/r/1072/ ........ + ................ + +2011-01-18 20:21 +0000 [r302268] Russell Bryant + + * /, main/astobj2.c: Merged revisions 302267 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r302267 | russell | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) + | 5 lines Don't enable AO2_DEBUG by default if AST_DEVMODE is on. + AO2_DEBUG is not important and is causing a false compiler + warning to be generated on my Ubuntu Natty dev box. ........ + +2011-01-18 18:17 +0000 [r302178] Richard Mudgett + + * /, main/features.c: Merged revisions 302174 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r302174 | rmudgett | 2011-01-18 12:11:43 -0600 + (Tue, 18 Jan 2011) | 102 lines Merged revisions 302173 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r302173 | rmudgett | 2011-01-18 12:07:15 -0600 + (Tue, 18 Jan 2011) | 95 lines Merged revisions 302172 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) + | 88 lines Issues with DTMF triggered attended transfers. Issue + #17999 1) A calls B. B answers. 2) B using DTMF dial *2 (code in + features.conf for attended transfer). 3) A hears MOH. B dial + number C 4) C ringing. A hears MOH. 5) B hangup. A still hears + MOH. C ringing. 6) A hangup. C still ringing until + "atxfernoanswertimeout" expires. For v1.4 C will ring forever + until C answers the dead line. (Issue #17096) Problem: When A and + B hangup, C is still ringing. Issue #18395 SIP call limit of B is + 1 1. A call B, B answered 2. B *2(atxfer) call C 3. B hangup, C + ringing 4. Timeout waiting for C to answer 5. Recall to B fails + because B has reached its call limit. Because B reached its call + limit, it cannot do anything until the transfer it started + completes. Issue #17273 Same scenario as issue 18395 but party B + is an FXS port. Party B cannot do anything until the transfer it + started completes. If B goes back off hook before C answers, B + hears ringback instead of the expected dialtone. ********** Note + for the issue #17273 and #18395 fix: DTMF attended transfer works + within the channel bridge. Unfortunately, when either party A or + B in the channel bridge hangs up, that channel is not completely + hung up until the transfer completes. This is a real problem + depending upon the channel technology involved. For chan_dahdi, + the channel is crippled until the hangup is complete. Either the + channel is not useable (analog) or the protocol disconnect + messages are held up (PRI/BRI/SS7) and the media is not released. + For chan_sip, a call limit of one is going to block that endpoint + from any further calls until the hangup is complete. For party A + this is a minor problem. The party A channel will only be in this + condition while party B is dialing and when party B and C are + conferring. The conversation between party B and C is expected to + be a short one. Party B is either asking a question of party C or + announcing party A. Also party A does not have much incentive to + hangup at this point. For party B this can be a major problem + during a blonde transfer. (A blonde transfer is our term for an + attended transfer that is converted into a blind transfer. :)) + Party B could be the operator. When party B hangs up, he assumes + that he is out of the original call entirely. The party B channel + will be in this condition while party C is ringing, while + attempting to recall party B, and while waiting between call + attempts. WARNING: The ATXFER_NULL_TECH conditional is a hack to + fix the problem. It will replace the party B channel technology + with a NULL channel driver to complete hanging up the party B + channel technology. The consequences of this code is that the 'h' + extension will not be able to access any channel technology + specific information like SIP statistics for the call. + ATXFER_NULL_TECH is not defined by default. ********** (closes + issue #17999) Reported by: iskatel Tested by: rmudgett JIRA + SWP-2246 (closes issue #17096) Reported by: gelo Tested by: + rmudgett JIRA SWP-1192 (closes issue #18395) Reported by: + shihchuan Tested by: rmudgett (closes issue #17273) Reported by: + grecco Tested by: rmudgett Review: + https://reviewboard.asterisk.org/r/1047/ ........ + ................ ................ + +2011-01-17 16:38 +0000 [r302006-302048] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 293493 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) + | 14 lines Only offer codecs both sides support for directmedia + When using directmedia, Asterisk needs to limit the codecs + offered to just the ones that both sides recognize, otherwise + they may end up sending audio that the other side doesn't + understand. (closes issue #17403) Reported by: one47 Patches: + sip_codecs_simplified4 uploaded by one47 (license 23) Tested by: + one47, falves11 Review: https://reviewboard.asterisk.org/r/967/ + ........ + + * /, configs/sip.conf.sample: Merged revisions 302005 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r302005 | twilson | 2011-01-17 09:04:59 -0600 (Mon, 17 + Jan 2011) | 2 lines Document "encryption" option in + sip.conf.sample ........ + +2011-01-14 21:13 +0000 [r301947] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 301946 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301946 | rmudgett | 2011-01-14 15:09:57 -0600 (Fri, 14 Jan 2011) + | 13 lines Deadlock between dahdi_request() and pri_dchannel() + processing an incomming call. The sig_pri_new_ast_channel() is + called with the channel private lock held when pri_dchannel() + calls it and no channel private lock held when dahdi_request() + calls it. The use of pri_grab() in sig_pri_new_ast_channel() + could leave the channel private lock held when it returns if the + lock was not held before calling it. Make + sig_pri_new_ast_channel() just lock the PRI span lock instead of + using pri_grab(). It is safe to do this because dahdi_request() + does not have the channel private lock and the deadlock potential + with the PRI span lock is only between pri_dchannel() and other + threads. ........ + +2011-01-14 20:18 +0000 [r301858] Brett Bryant + + * channels/chan_multicast_rtp.c, /: Merged revisions 301851 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301851 | bbryant | 2011-01-14 15:11:55 -0500 (Fri, 14 Jan 2011) + | 6 lines Changing previous revisions 301845/301847 to use + ast_sockaddr_setnull() instead of setting the field manually to + avoid uninitialized data. Review: + https://reviewboard.asterisk.org/r/1076/ ........ + +2011-01-14 20:07 +0000 [r301850] Andrew Latham + + * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to + function documentation. Fix amatuer type mistake + +2011-01-14 19:44 +0000 [r301847] Brett Bryant + + * channels/chan_multicast_rtp.c, /: Merged revisions 301845 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301845 | bbryant | 2011-01-14 14:35:23 -0500 (Fri, 14 Jan 2011) + | 9 lines Fix for a consistent MulticastRTP channel driver crash + due to use of unitilized data. (closes issue #18290) (closes + issue #18602) Reported by: voipgate, wybecom Review: + https://reviewboard.asterisk.org/r/1076/ ........ + +2011-01-14 19:39 +0000 [r301846] Andrew Latham + + * funcs/func_base64.c, /, funcs/func_aes.c: Add relationships to + function documentation. + +2011-01-14 17:34 +0000 [r301791] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 301790 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301790 | jpeeler | 2011-01-14 11:32:52 -0600 (Fri, 14 Jan 2011) + | 42 lines Resolve deadlock involving REFER. Two fixes: 1) One + must always have the private unlocked before calling + pbx_builtin_setvar_helper to not invalidate locking order since + it locks the channel. 2) Unlock the channel before calling + pbx_find_extension, which starts and stops autoservice during the + lookup. The problem scenario as illustrated by the reporter: + Thread: do_monitor ----------------------- handle_request_do + handle_incoming handle_request_refer ast_parking_ext_valid + pbx_find_extension ast_autoservice_stop while (chan_list_state == + as_chan_list_state) { usleep(1000); } Thread: autoservice_run + ----------------------- autoservice_run chan = ast_waitfor_n + ast_waitfor_nandfds ast_waitfor_nandfds_classic / simple / + complex (depending on your system) ast_channel_lock(c[x]); + handle_request_do and schedule_process_request_queue locks the + owner if it exists. The autoservice thread is waiting for the + channel lock, which wasn't ever released since the do_monitor + thread was waiting for autoservice operations to complete. Solved + by unlocking the channel but keeping a reference to guarantee + safety. (closes issue #18403) Reported by: jthurman Patches: + 20110103-blind_deadlock.diff uploaded by jthurman (license 614) + issue18403.patch uploaded by jpeeler (license 325) Tested by: + jthurman ........ + +2011-01-13 17:02 +0000 [r301732] Leif Madsen + + * /, configs/phoneprov.conf.sample: Merged revisions 301731 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301731 | lmadsen | 2011-01-13 11:01:43 -0600 + (Thu, 13 Jan 2011) | 15 lines Merged revisions 301730 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301730 | lmadsen | 2011-01-13 11:01:11 -0600 (Thu, 13 Jan 2011) + | 7 lines Add static entry for split Polycom 332 firmware. + (closes issue #18607) Reported by: cjacobsen Patches: + polycom_331.diff uploaded by cjacobsen (license 1029) Tested by: + lathama ........ ................ + +2011-01-13 16:27 +0000 [r301729] Paul Belanger + + * main/pbx.c, CHANGES: Add dialplan variables for asterisk.conf + directories Review: https://reviewboard.asterisk.org/r/1075/ + +2011-01-12 21:24 +0000 [r301684] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 301683 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301683 | twilson | 2011-01-12 15:19:48 -0600 + (Wed, 12 Jan 2011) | 15 lines Merged revisions 301682 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301682 | twilson | 2011-01-12 15:05:02 -0600 (Wed, 12 Jan 2011) + | 9 lines Don't reject all SUBSCRIBE auth requests When merging + another SUBSCRIBE fix from 1.4, some braces were put in the wrong + place. This patch fixes that. (closes issue #18597) Reported by: + thsgmbh ........ ................ + +2011-01-12 18:52 +0000 [r301596] Matthew Nicholson + + * main/manager.c, /: Merged revisions 301595 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301595 | mnicholson | 2011-01-12 12:51:37 -0600 + (Wed, 12 Jan 2011) | 22 lines Merged revisions 301594 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r301594 | mnicholson | 2011-01-12 12:50:31 -0600 + (Wed, 12 Jan 2011) | 15 lines Removed a usleep(1) that shouldn't + be necessary in session_do, and removed the ms_t member from the + mansession_session structure. Merged revisions 301591 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan + 2011) | 5 lines Don't store the thread id for the manager session + in the structure we pass to the thread for the manager session. + ABE-2543 ........ ................ ................ + +2011-01-12 18:12 +0000 [r301505] Jeff Peeler + + * main/channel.c, /: Merged revisions 301504 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301504 | jpeeler | 2011-01-12 12:12:08 -0600 + (Wed, 12 Jan 2011) | 26 lines Merged revisions 301503 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r301503 | jpeeler | 2011-01-12 12:11:49 -0600 + (Wed, 12 Jan 2011) | 19 lines Merged revisions 301502 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r301502 | jpeeler | 2011-01-12 12:10:42 -0600 (Wed, 12 Jan 2011) + | 12 lines Fix CPU spike when pressing DTMF after agent login. + The problem here is that DTMF was being continuously deferred and + requeued since ast_safe_sleep is called in a loop. There are + serveral other places in the code that sleeps and then loops in a + similar fashion. Because of this fact I opted to not defer DTMF + any more, which will not affect the original fix: + https://reviewboard.asterisk.org/r/674 (closes issue #18130) + Reported by: rgj ........ ................ ................ + +2011-01-12 16:05 +0000 [r301447] David Vossel + + * /, main/file.c: Merged revisions 301446 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301446 | dvossel | 2011-01-12 10:05:12 -0600 (Wed, 12 Jan 2011) + | 2 lines Removal of unused variables so Asterisk will compile. + ........ + +2011-01-12 15:59 +0000 [r301445] Stefan Schmidt + + * Makefile: fix wrong text of rerun menuselect after user interface + warning the warning, if no user interface for menuselect warning + was found is not right. you have to rerun configure before make + menuselect after installing a proper user interface. (closes + issue 0018594) Reported by: Dovid + +2011-01-12 00:27 +0000 [r301403] Tilghman Lesher + + * /, main/file.c: Merged revisions 301402 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301402 | tilghman | 2011-01-11 18:26:39 -0600 (Tue, 11 Jan 2011) + | 7 lines Call execl() directly for a better solution for paths + with spaces. (closes issue #18600) Reported by: ebroad Patches: + 20110111__issue18600__2.diff.txt uploaded by tilghman (license + 14) ........ + +2011-01-11 19:19 +0000 [r301319] Paul Belanger + + * /, configs/extensions.conf.sample: Merged revisions 301311 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301311 | pabelanger | 2011-01-11 14:16:06 -0500 + (Tue, 11 Jan 2011) | 9 lines Merged revisions 301310 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r301310 | pabelanger | 2011-01-11 14:14:31 -0500 (Tue, + 11 Jan 2011) | 2 lines Fix a logic issue when passing context ARG + ........ ................ + +2011-01-11 18:55 +0000 [r301309] Matthew Nicholson + + * /, main/utils.c: Merged revisions 301308 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301308 | mnicholson | 2011-01-11 12:51:40 -0600 + (Tue, 11 Jan 2011) | 18 lines Merged revisions 301307 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r301307 | mnicholson | 2011-01-11 12:42:05 -0600 + (Tue, 11 Jan 2011) | 11 lines Merged revisions 301305 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r301305 | mnicholson | 2011-01-11 12:34:40 -0600 (Tue, 11 Jan + 2011) | 4 lines Prevent buffer overflows in ast_uri_encode() + ABE-2705 ........ ................ ................ + +2011-01-10 22:40 +0000 [r301264] Tilghman Lesher + + * /, main/strcompat.c: Merged revisions 301263 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r301263 | tilghman | 2011-01-10 16:39:31 -0600 (Mon, 10 Jan 2011) + | 8 lines Little endian machines were not converted properly. + (closes issue #18583) Reported by: jcovert Patches: + 20110110__issue18583.diff.txt uploaded by tilghman (license 14) + Tested by: jcovert ........ + +2011-01-09 21:42 +0000 [r301178-301222] Paul Belanger + + * /, configure, configure.ac, autoconf/ast_ext_lib.m4: Merged + revisions 301221 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301221 | pabelanger | 2011-01-09 16:40:34 -0500 + (Sun, 09 Jan 2011) | 21 lines Merged revisions 301220 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan + 2011) | 14 lines SOUND_CACHE_DIR now defaults to empty Sounds + files included in the Asterisk tarball were being ignored and + re-downloaded. Users wanting to cache the files can still + override the setting using the --with-sounds-cache option. + (closes issue #18589) Reported by: pabelanger Patches: + issue18589.patch uploaded by pabelanger (license 224) Tested by: + pabelanger Review: https://reviewboard.asterisk.org/r/1074/ + ........ ................ + + * /, apps/app_verbose.c: Merged revisions 301177 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301177 | pabelanger | 2011-01-08 17:00:12 -0500 + (Sat, 08 Jan 2011) | 14 lines Merged revisions 301176 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan + 2011) | 7 lines Indicate log level argument for Log() is not + optional (closes issue #18586) Reported by: kshumard Patches: + app_verbose.c.patch uploaded by kshumard (license 92) ........ + ................ + +2011-01-08 01:13 +0000 [r301135] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 301134 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r301134 | rmudgett | 2011-01-07 19:11:31 -0600 (Fri, 07 + Jan 2011) | 7 lines The DTMF attended transfer feature cannot + callback a chan_dahdi BRI phone. The DAHDI ISDN channel name is + not dialable. Make a channel name like DAHDI/i3/400-12 dialable + when the sequence number is stripped off of the name. ........ + +2011-01-07 20:53 +0000 [r301091] Jason Parker + + * /, apps/app_meetme.c: Merged revisions 301090 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301090 | qwell | 2011-01-07 14:53:02 -0600 + (Fri, 07 Jan 2011) | 15 lines Merged revisions 301089 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) | + 8 lines Initialize useropts/adminopts in case there is no column + in the realtime DB. (closes issue #18182) Reported by: dimas + Patches: v1-18182.patch uploaded by dimas (license 88) Tested by: + dimas ........ ................ + +2011-01-07 19:58 +0000 [r301048] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 301047 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r301047 | jpeeler | 2011-01-07 13:58:30 -0600 + (Fri, 07 Jan 2011) | 15 lines Merged revisions 301046 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011) + | 8 lines Fix regression causing forwarding voicemails to not + work with file storage. I had actually already fixed this in + 295200 in 1.4 and thought it wasn't missing in the other branches + for some reason. (closes issue #18358) Reported by: cabal95 + ........ ................ + +2011-01-07 18:23 +0000 [r301008] Tilghman Lesher + + * funcs/func_curl.c: Oops, missed the actual decoding part. (closes + issue #18046) Reported by: wdoekes + +2011-01-07 17:24 +0000 [r300959] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 300955 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300955 | jpeeler | 2011-01-07 11:24:14 -0600 + (Fri, 07 Jan 2011) | 21 lines Merged revisions 300951 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r300951 | jpeeler | 2011-01-07 11:23:37 -0600 + (Fri, 07 Jan 2011) | 14 lines Merged revisions 300918 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) + | 7 lines Ensure good bye prompt in voicemail is played at the + correct time. Specifically in the case of timing out but not + leaving voicemail nothing should be heard. And when leaving + voicemail it should be heard. ABE-2647 ........ ................ + ................ + +2011-01-07 07:47 +0000 [r300882] Mark Murawki + + * res/res_config_pgsql.c: Added support for postgres database retry + query on disconnection to res_config_pgsql If your postgres + connection died suddenly in between res_config_pgsql queries, the + next query will fail because the query is executed on a + disconnected/disconnecting handle. The query is abandoned and is + returned from in error. Now we will reconnect and try again if a + query was run on a disconnected connection. (closes issue #18071) + +2011-01-06 17:50 +0000 [r300799-300841] Tilghman Lesher + + * Makefile, funcs/func_curl.c: XML validation + + * funcs/func_curl.c: Add a hashcompat mode called "legacy", which + translates a literal plus sign to a space. (closes issue #18046) + Reported by: wdoekes Patches: 20100930__issue18046.diff.txt + uploaded by tilghman (license 14) + + * /, addons/res_config_mysql.c: Merged revisions 300798 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r300798 | tilghman | 2011-01-06 00:28:18 -0600 (Thu, 06 Jan 2011) + | 8 lines Don't destroy handle not created by use (because the + caller will). (closes issue #18526) Reported by: makoto Patches: + res-config-mysql-include.patch uploaded by makoto (license 38) + Tested by: makoto ........ + +2011-01-06 01:41 +0000 [r300761] David Ruggles + + * Makefile, contrib/scripts/safe_asterisk: update safe_asterisk + script change defaults to make a little more sense. Default log + location is now asterisk log location and default email + notification has been changed to root on the local machine + Review: https://reviewboard.asterisk.org/r/1067/ + +2011-01-05 21:07 +0000 [r300716] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 300714 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300714 | rmudgett | 2011-01-05 14:54:21 -0600 + (Wed, 05 Jan 2011) | 21 lines Merged revision 300711 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r300711 | rmudgett | 2011-01-05 13:43:55 -0600 (Wed, + 05 Jan 2011) | 14 lines A call retrieved from hold may wind up + with no audio. If the retrieved call is natively bridged then the + call may not have any audio path. The following warning message + is given: "Failed to add to conference /: + Invalid argument". * Open the media on a B channel when + pri_fixup_principle() moves the call from a no_b_channel channel + to a real channel. * Added lock protection while + pri_fixup_principle() moves a call from one private structure to + another. * Made some pri_fixup_principle() messages more + meaningful. .......... ................ + +2011-01-05 18:57 +0000 [r300624] Tilghman Lesher + + * /, res/res_odbc.c: Merged revisions 300623 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300623 | tilghman | 2011-01-05 12:56:12 -0600 + (Wed, 05 Jan 2011) | 24 lines Merged revisions 300622 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r300622 | tilghman | 2011-01-05 12:54:58 -0600 + (Wed, 05 Jan 2011) | 17 lines Merged revisions 300621 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r300621 | tilghman | 2011-01-05 12:47:46 -0600 (Wed, 05 Jan 2011) + | 10 lines Use the sanity check in place of the + disconnect/connect cycle. The disconnect/connect cycle has the + potential to cause random crashes. (closes issue #18243) Reported + by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147) + Tested by: ks3 ........ ................ ................ + +2011-01-05 16:30 +0000 [r300576] Paul Belanger + + * /, cdr/cdr_sqlite.c: Merged revisions 300575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300575 | pabelanger | 2011-01-05 11:29:19 -0500 + (Wed, 05 Jan 2011) | 13 lines Merged revisions 300574 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r300574 | pabelanger | 2011-01-05 11:28:07 -0500 (Wed, 05 Jan + 2011) | 6 lines Change deprecated message to LOG_WARNING Also + removed latter part of message Discussed on #asterisk-dev + ........ ................ + +2011-01-04 21:54 +0000 [r300434-300522] Leif Madsen + + * /, channels/chan_sip.c, channels/chan_agent.c, + channels/chan_iax2.c, main/xmldoc.c: Merged revisions 300521 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300521 | lmadsen | 2011-01-04 15:53:27 -0600 + (Tue, 04 Jan 2011) | 17 lines Merged revisions 300520 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011) + | 9 lines Fix backwards and broken XML documentation. (closes + issue #18547) Reported by: jcovert Patches: xmldoc.c.patch + uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded + by jcovert (license 551) chan_sip.c.patch uploaded by jcovert + (license 551) chan_agent.c.patch uploaded by jcovert (license + 551) ........ ................ + + * configs/users.conf.sample, /: Merged revisions 300433 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300433 | lmadsen | 2011-01-04 15:00:55 -0600 + (Tue, 04 Jan 2011) | 15 lines Merged revisions 300431 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r300431 | lmadsen | 2011-01-04 15:00:29 -0600 (Tue, 04 Jan 2011) + | 7 lines Add some documentation to users.conf.sample. (closes + issue #18531) Reported by: lathama Patches: + users.conf.sample2.diff uploaded by lathama (license 1028) Tested + by: lathama ........ ................ + +2011-01-04 21:00 +0000 [r300432] Russell Bryant + + * /, contrib/scripts/autosupport.8, contrib/scripts/autosupport: + Merged revisions 300430 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300430 | russell | 2011-01-04 15:00:16 -0600 + (Tue, 04 Jan 2011) | 18 lines Merged revisions 300429 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r300429 | russell | 2011-01-04 14:59:56 -0600 + (Tue, 04 Jan 2011) | 11 lines Merged revisions 300428 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r300428 | russell | 2011-01-04 14:56:04 -0600 (Tue, 04 Jan 2011) + | 4 lines Update the autosupport script from Digium support. + (closes AST-395) ........ ................ ................ + +2011-01-04 19:45 +0000 [r300385] Leif Madsen + + * phoneprov/000000000000.cfg, /: Merged revisions 300384 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r300384 | lmadsen | 2011-01-04 13:45:22 -0600 (Tue, 04 Jan 2011) + | 7 lines Update STAT() to use the comma instead of the pipe. + (closes issue #18503) Reported by: cjacobsen Patches: + old_separator.diff uploaded by cjacobsen (license 1029) Tested + by: lathama ........ + +2011-01-04 18:51 +0000 [r300345] Moises Silva + + * channels/chan_dahdi.c: Update MFC-R2 code to use new DTMF-R2 + functionality in OpenR2 (closes issue #18576) + +2011-01-04 18:06 +0000 [r300302] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 300301 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300301 | twilson | 2011-01-04 11:54:41 -0600 + (Tue, 04 Jan 2011) | 29 lines Merged revisions 300298 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r300298 | twilson | 2011-01-04 11:37:26 -0600 + (Tue, 04 Jan 2011) | 22 lines Merged revisions 300216 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r300216 | twilson | 2011-01-04 11:11:48 -0600 (Tue, 04 Jan 2011) + | 15 lines Don't authenticate SUBSCRIBE re-transmissions This + only skips authentication on retransmissions that are already + authenticated. A similar method is already used for INVITES. This + is the kind of thing we end up having to do when we don't have a + transaction layer... (closes issue #18075) Reported by: mdu113 + Patches: diff.txt uploaded by twilson (license 396) Tested by: + twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ + ........ ................ ................ + +2011-01-04 17:04 +0000 [r300215] Jan Kalab + + * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, /: + Merged revisions 300214 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r300214 | pitel | 2011-01-04 18:01:52 +0100 (Út, 04 led 2011) | 7 + lines Memory leaking in calendars ne_request_destroy() was + missing in icalendar and exchange calendar modules, causing + memory leak. (closes issue #18521) Review: + https://reviewboard.asterisk.org/r/1068/ ........ + +2011-01-04 16:38 +0000 [r300168-300212] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, + configs/chan_dahdi.conf.sample, UPGRADE.txt, CHANGES, + channels/sig_pri.c: Optional HOLD/RETRIEVE signaling for PTMP TE + when the bridge goes on and off hold. Added the moh_signaling + option to specify what to do when the channel's bridged peer puts + the ISDN channel on and off of hold. Implemented as a FSM to + control libpri ISDN signaling when the bridged peer places the + channel on and off of hold with the AST_CONTROL_HOLD and + AST_CONTROL_UNHOLD control frames. JIRA SWP-2687 JIRA ABE-2691 + Review: https://reviewboard.asterisk.org/r/1063/ + + * /, main/features.c: Merged revisions 300166 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r300166 | rmudgett | 2011-01-03 17:14:55 -0600 + (Mon, 03 Jan 2011) | 11 lines Merged revisions 300165 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r300165 | rmudgett | 2011-01-03 17:02:13 -0600 (Mon, 03 Jan 2011) + | 4 lines Use correct variable for atxfercallbackretries config + option. * Misc formatting changes. ........ ................ + +2011-01-03 14:09 +0000 [r300121] David Ruggles + + * apps/app_externalivr.c: initialize playing_silence in struct + initialization playing_silence was not initialized with the + struct was initialized, it was being set after the fact which + caused problems if something that relied on playing_silence being + set was called too quickly (closes issue #18430) Reported by: + stevebrandli Patches: externalivr.patch uploaded by + thedavidfactor (license 903) Tested by: thedavidfactor, + stevebrandli + +2011-01-03 13:15 +0000 [r300083] Leif Madsen + + * /, pbx/pbx_dundi.c: Merged revisions 300082 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r300082 | lmadsen | 2011-01-03 07:14:25 -0600 (Mon, 03 Jan 2011) + | 11 lines Increase side of mapping response field. I've + increased the size of the response field in a DUNDi mapping + because of some documentation I'm writing. Previously it was set + to AST_MAX_EXTENSION which is only 80 characters, which is far + too small when you're using some dialplan functions to craft a + response. The example I'm using is: extensions => + RegisteredDevices,0,SIP,dundi:very_awesome_password/${IF($[${DB_EXISTS(phones/${NUMBER}/device)}]?${DB(phones/${NUMBER}/device)}:None)},nopartial + ........ + +2010-12-31 09:29 +0000 [r300044-300045] Tilghman Lesher + + * cdr/cdr_adaptive_odbc.c, CHANGES, + configs/cdr_adaptive_odbc.conf.sample: Support negative filters. + (closes issue #17979) Reported by: tilghman Patches: + 20100911__for_blitzrage.diff.txt uploaded by tilghman (license + 14) Tested by: lmadsen + + * main/logger.c, CHANGES: Support an alternate configuration file + for the 'logger reload' command. (closes issue #17668) Reported + by: tilghman Patches: 20100718__logger_reload_altconf__2.diff.txt + uploaded by tilghman (license 14) Review: (by lmadsen, russell + within comments on issue tracker) + +2010-12-29 22:19 +0000 [r300000] Sean Bright + + * main/asterisk.c: Remove some trailing whitespace and steal + revision 300000. + +2010-12-29 22:03 +0000 [r299990] Tilghman Lesher + + * /, main/file.c, apps/app_voicemail.c: Merged revisions 299989 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299989 | tilghman | 2010-12-29 16:02:59 -0600 (Wed, 29 Dec 2010) + | 4 lines Quote arguments, just in case there's a space in a + pathname. (Diagnosed by pabelanger on #asterisk-dev, fixed by + me.) ........ + +2010-12-29 19:29 +0000 [r299866-299949] Paul Belanger + + * /, sounds/Makefile: Merged revisions 299948 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299948 | pabelanger | 2010-12-29 14:28:36 -0500 (Wed, 29 Dec + 2010) | 2 lines Only remove /tmp/astdatadir, not + /var/lib/asterisk ........ + + * Makefile, /, build_tools/make_sample_voicemail, sounds/Makefile: + Merged revisions 299907 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299907 | pabelanger | 2010-12-29 13:22:23 -0500 (Wed, 29 Dec + 2010) | 2 lines Properly quote varibles for MAC OS X ........ + + * /, apps/app_chanspy.c: Merged revisions 299865 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299865 | pabelanger | 2010-12-28 13:53:37 -0500 + (Tue, 28 Dec 2010) | 9 lines Merged revisions 299864 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue, + 28 Dec 2010) | 2 lines Documentation typo ........ + ................ + +2010-12-27 21:23 +0000 [r299754-299824] Tilghman Lesher + + * /, sounds/Makefile: Merged revisions 299820 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299820 | tilghman | 2010-12-27 15:23:10 -0600 (Mon, 27 Dec 2010) + | 2 lines More space-in-pathname issues. ........ + + * Makefile, /, Makefile.moddir_rules, sounds/Makefile: Merged + revisions 299794 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299794 | tilghman | 2010-12-27 14:41:04 -0600 (Mon, 27 Dec 2010) + | 2 lines Mac OS X spaces-in-pathnames fix. ........ + + * /, configure, configure.ac: Merged revisions 299752 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r299752 | tilghman | 2010-12-26 15:15:58 -0600 (Sun, 26 + Dec 2010) | 2 lines Properly quote path on Darwin. ........ + +2010-12-25 16:35 +0000 [r299715] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/ooh323c/src/ooh245.c, + addons/ooh323c/src/oochannels.c, addons/ooh323c/src/ooq931.c: + Change order of sending TCS and MSD packets Change order of + sending Terminal Capability Set and MasterSlave Determination + packets, MSD send when TCS exchange procedure is done (we send + tcs ack to remote and we have remote tcs ack already or we + receive tcs ack from remote and we have send our tcs ack to + remote already). Some endpoints can work in this sequence only, i + suggest they can't work with both (tcs and msd) exchange + procedures simultaneously. Also changed StartH245 facility + message sending. It send on incoming calls only due to some + endpoints can't proccess properly this facility messages on their + incoming calls. (closes issue #18433) Reported by: MrHanMan + Patches: tcs-msd-h245-3.patch uploaded by may213 (license 454) + Tested by: MrHanMan, may213 + +2010-12-25 10:08 +0000 [r299584-299627] Tilghman Lesher + + * channels/chan_local.c, /: Merged revisions 299626 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299626 | tilghman | 2010-12-25 04:07:15 -0600 + (Sat, 25 Dec 2010) | 19 lines Merged revisions 299625 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r299625 | tilghman | 2010-12-25 04:05:00 -0600 + (Sat, 25 Dec 2010) | 12 lines Merged revisions 299624 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r299624 | tilghman | 2010-12-25 04:04:06 -0600 (Sat, 25 Dec 2010) + | 5 lines Move check for extension existence below variable + inheritance, due to the possible use of an eswitch. (closes issue + #16228) Reported by: jlaguilar ........ ................ + ................ + + * /, addons/res_config_mysql.c: Merged revisions 299583 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299583 | tilghman | 2010-12-24 11:58:30 -0600 (Fri, 24 Dec 2010) + | 7 lines Reset 'first' variable after usage. (closes issue + #18525) Reported by: makoto Patches: + res-config-mysql-update2.patch uploaded by makoto (license 38) + ........ + +2010-12-23 01:46 +0000 [r299493] Moises Silva + + * channels/chan_dahdi.c: Enqueue AST_CONTROL_PROGRESS after + AST_CONTROL_RINGING when MFC-R2 calls are accepted (closes issue + #18438) Reported by: mariner7 Tested by: moy + +2010-12-22 20:10 +0000 [r299450] Tilghman Lesher + + * res/ael/pval.c, pbx/ael/ael-test/ref.ael-vtest25, + pbx/ael/ael-test/ref.ael-vtest17, /, + pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test19, + pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 299449 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299449 | tilghman | 2010-12-22 14:05:02 -0600 + (Wed, 22 Dec 2010) | 15 lines Merged revisions 299448 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r299448 | tilghman | 2010-12-22 14:03:30 -0600 (Wed, 22 Dec 2010) + | 8 lines Resolve warnings by disambiguating the "s" extension as + used by chan_dahdi from the "s" extension as used by the AEL + macros. (closes issue #18480) Reported by: nivek Patches: + 20101215__issue18480__2.diff.txt uploaded by tilghman (license + 14) Tested by: nivek ........ ................ + +2010-12-22 02:12 +0000 [r299406] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 299405 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299405 | rmudgett | 2010-12-21 20:10:39 -0600 (Tue, 21 Dec 2010) + | 17 lines Chan_dahdi sends an empty COLP on the bridged channel. + Chan_dahdi always inserts a connected party IE when you call from + one dahdi channel to another dahdi channel, even if no such + information was received on the 2nd channel. This clears the + display of many phones. * Removed leftover artifact from before + the valid flag was added. * Updated all of the channel's caller + id information with the new connected line information instead of + just the string parts. (closes issue #18508) Reported by: wimpy + Patches: issue18508_trunk.patch uploaded by rmudgett (license + 664) Tested by: wimpy, rmudgett ........ + +2010-12-21 16:02 +0000 [r299355] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 299353 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299353 | mnicholson | 2010-12-21 09:25:03 -0600 + (Tue, 21 Dec 2010) | 30 lines Merged revisions 299242 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r299242 | mnicholson | 2010-12-20 15:25:35 -0600 + (Mon, 20 Dec 2010) | 23 lines Merged revisions + 299194,299198,299220 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r299194 | mnicholson | 2010-12-20 14:45:38 -0600 (Mon, 20 Dec + 2010) | 6 lines Respond as soon as possible with a 202 Accepted + to refer requests. This change also plugs a few memory leaks that + can occur when parking sip calls. ABE-2656 ........ r299198 | + mnicholson | 2010-12-20 15:00:44 -0600 (Mon, 20 Dec 2010) | 2 + lines Remove changes to via processing that were not supposed to + go into the last commit. ........ r299220 | mnicholson | + 2010-12-20 15:21:39 -0600 (Mon, 20 Dec 2010) | 4 lines Use + ast_free() instead of free() ABE-2656 ........ ................ + ................ + +2010-12-21 00:45 +0000 [r299313] Paul Belanger + + * configs/cel.conf.sample, /: Merged revisions 299312 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r299312 | pabelanger | 2010-12-20 19:44:08 -0500 (Mon, + 20 Dec 2010) | 8 lines Correct typo with USER_DEFINED event. + (closes issue #18461) Reported by: joscas Patches: + cel.conf.sample.diff uploaded by lathama (license 1028) Tested + by: lathama, joscas ........ + +2010-12-20 21:40 +0000 [r299249] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 299248 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r299248 | mmichelson | 2010-12-20 15:38:30 -0600 (Mon, 20 Dec + 2010) | 20 lines Fix a couple of CCSS issues. * Make sure to + allocate a cc_params structure when creating autopeers. * Use + sip_uri_cmp when retrieving SIP CC agents and monitors in case + parameters appear in the URI. (closes issue #18504) Reported by: + kkm (closes issue #18338) Reported by: GeorgeKonopacki Patches: + 18338.diff uploaded by mmichelson (license 60) Tested by: + GeorgeKonopacki ........ + +2010-12-20 18:18 +0000 [r299142] Tilghman Lesher + + * /, sample.call: Merged revisions 299138 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299138 | tilghman | 2010-12-20 12:17:28 -0600 + (Mon, 20 Dec 2010) | 9 lines Merged revisions 299136 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r299136 | tilghman | 2010-12-20 12:16:37 -0600 (Mon, 20 + Dec 2010) | 2 lines Documentation fix ........ ................ + +2010-12-20 18:03 +0000 [r299135] David Vossel + + * include/asterisk/astobj2.h, main/astobj2.c: New astobj2 flag for + issuing a callback without locking the container. + +2010-12-20 17:59 +0000 [r299133-299134] Russell Bryant + + * channels/chan_misdn.c: Fix chan_misdn build after sched API + changes. + + * addons/chan_ooh323.c, addons/chan_mobile.c: Fix some build errors + in addons due to sched API changes. + +2010-12-20 17:48 +0000 [r299132] Tilghman Lesher + + * /, cdr/cdr_pgsql.c: Merged revisions 299131 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299131 | tilghman | 2010-12-20 11:47:10 -0600 + (Mon, 20 Dec 2010) | 18 lines Merged revisions 299130 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r299130 | tilghman | 2010-12-20 11:41:24 -0600 (Mon, 20 Dec 2010) + | 11 lines If a call was not answered, then the billsec was + calculated unusually large. Also, due to a copy and paste error, + a request for the answer field would have given the start value, + instead. (closes issue #18460) Reported by: joscas Patches: + 20101215__issue18460.diff.txt uploaded by tilghman (license 14) + Tested by: joscas ........ ................ + +2010-12-20 17:15 +0000 [r299091] Russell Bryant + + * channels/chan_unistim.c, main/udptl.c, res/res_rtp_asterisk.c, + include/asterisk.h, main/rtp_engine.c, main/dnsmgr.c, + channels/chan_sip.c, main/ccss.c, include/asterisk/channel.h, + channels/chan_gtalk.c, tests/test_sched.c, channels/chan_iax2.c, + res/res_rtp_multicast.c, main/channel.c, main/cdr.c, + channels/chan_jingle.c, channels/chan_skinny.c, + channels/sip/include/globals.h, res/res_stun_monitor.c, + channels/sip/dialplan_functions.c, channels/chan_h323.c, + include/asterisk/sched.h, pbx/pbx_dundi.c, + include/asterisk/udptl.h, include/asterisk/rtp_engine.h, + main/sched.c, channels/chan_mgcp.c, res/res_calendar.c: Some + scheduler API cleanup and improvements. Previously, I had added + the ast_sched_thread stuff that was a generic scheduler thread + implementation. However, if you used it, it required using + different functions for modifying scheduler contents. This patch + reworks how this is done and just allows you to optionally start + a thread on the original scheduler context structure that has + always been there. This makes it trivial to switch to the generic + scheduler thread implementation without having to touch any of + the other code that adds or removes scheduler entries. In + passing, I made some naming tweaks to add ast_ prefixes where + they were not there before. Review: + https://reviewboard.asterisk.org/r/1007/ + +2010-12-20 16:19 +0000 [r299089] Leif Madsen + + * /, main/features.c: Merged revisions 299088 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r299088 | lmadsen | 2010-12-20 10:18:26 -0600 + (Mon, 20 Dec 2010) | 13 lines Merged revisions 299087 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r299087 | lmadsen | 2010-12-20 10:18:03 -0600 (Mon, 20 Dec 2010) + | 5 lines Note that Park() timeout is milliseconds. (closes issue + #15758) Reported by: mmurdock Tested by: mmurdock, seanbright + ........ ................ + +2010-12-20 09:14 +0000 [r299005] Tzafrir Cohen + + * channels/sig_pri.h, channels/chan_sip.c, main/aoc.c: Typos: + recieved => received + +2010-12-18 00:08 +0000 [r298819-298961] Tilghman Lesher + + * utils/refcounter.c, include/asterisk/utils.h, + build_tools/cflags-devmode.xml, /, main/Makefile, + include/asterisk/autoconfig.h.in, configure.ac, utils/hashtest.c, + main/utils.c, main/astobj2.c, utils/conf2ael.c, + include/asterisk/logger.h, configure, + build_tools/menuselect-deps.in, main/logger.c, utils/hashtest2.c, + utils/ael_main.c, makeopts.in, utils/check_expr.c: Merged + revisions 298960 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298960 | tilghman | 2010-12-17 17:52:04 -0600 + (Fri, 17 Dec 2010) | 20 lines Merged revisions 298957 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298957 | tilghman | 2010-12-17 17:30:55 -0600 + (Fri, 17 Dec 2010) | 13 lines Merged revisions 298905 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) + | 6 lines Let Asterisk find better backtrace information with + libbfd. The menuselect option BETTER_BACKTRACES, if enabled, will + use libbfd to search for better symbol information within both + the Asterisk binary, as well as loaded modules, to assist when + using inline backtraces to track down problems. Review: + https://reviewboard.asterisk.org/r/1055/ ........ + ................ ................ + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 298827 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r298827 | tilghman | 2010-12-17 15:18:18 -0600 (Fri, 17 Dec 2010) + | 8 lines -v implies -f, so override with -F. (closes issue + #18446) Reported by: lathama Patches: rc.debian.asterisk.diff + uploaded by lathama (license 1028) Tested by: lathama ........ + + * /, configure, configure.ac: Merged revisions 298818 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298818 | tilghman | 2010-12-17 15:04:21 -0600 + (Fri, 17 Dec 2010) | 15 lines Merged revisions 298817 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r298817 | tilghman | 2010-12-17 15:03:06 -0600 (Fri, 17 Dec 2010) + | 8 lines Also include PTHREAD_LIBS and PTHREAD_CFLAGS for SQLite + 3, as it's needed on some platforms. (closes issue #18493) + Reported by: pprindeville Patches: asterisk-1.8-sqlite3.patch + uploaded by pprindeville (license 347) Tested by: pprindeville + ........ ................ + +2010-12-17 17:29 +0000 [r298774] Brad Watkins + + * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions + 298773 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r298773 | marquis | 2010-12-17 12:26:31 -0500 (Fri, 17 Dec 2010) + | 10 lines Fix parsing of mwi => lines in sip.conf Reworking + parsing of mwi => lines to resolve a segfault. Also add a set of + unit tests for the function that does the parsing. (closes issue + #18350) Reported by: gbour Tested by: Marquis, gbour Review: + https://reviewboard.asterisk.org/r/1053/ ........ + +2010-12-16 23:33 +0000 [r298599-298686] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 298685 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298685 | jpeeler | 2010-12-16 17:31:50 -0600 + (Thu, 16 Dec 2010) | 16 lines Merged revisions 298684 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298684 | jpeeler | 2010-12-16 17:30:59 -0600 + (Thu, 16 Dec 2010) | 9 lines Merged revisions 298683 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16 + Dec 2010) | 2 lines After recording only silence for a voicemail + prepending, restore backup files. ........ ................ + ................ + + * /, apps/app_queue.c: Merged revisions 298598 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298598 | jpeeler | 2010-12-16 14:51:44 -0600 + (Thu, 16 Dec 2010) | 21 lines Merged revisions 298597 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298597 | jpeeler | 2010-12-16 14:49:33 -0600 + (Thu, 16 Dec 2010) | 14 lines Merged revisions 298596 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) + | 7 lines Fix improper hangup when doing an attended transfer to + queue. Had to indicate ringing in wait_for_answer so the attended + transfer code would not try and hang up the local channel it + created, which would kill the call. ABE-2624 ........ + ................ ................ + +2010-12-16 09:29 +0000 [r298441-298545] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 298539 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r298539 | tilghman | 2010-12-16 03:28:17 -0600 (Thu, 16 Dec 2010) + | 8 lines Ensure the ipaddr field in realtime is large enough to + handle IPv6 addresses. (closes issue #18464) Reported by: IgorG + Patches: realtime_ipv6store.diff uploaded by IgorG (license 20) + (plus a few additional lines by tilghman) ........ + + * res/res_config_odbc.c, /: Merged revisions 298482 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298482 | tilghman | 2010-12-16 03:05:28 -0600 + (Thu, 16 Dec 2010) | 28 lines Merged revisions 298481 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298481 | tilghman | 2010-12-16 03:04:38 -0600 + (Thu, 16 Dec 2010) | 21 lines Merged revisions 298480 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r298480 | tilghman | 2010-12-16 03:03:40 -0600 (Thu, 16 Dec 2010) + | 14 lines Only increment the pointer once per loop, otherwise we + corrupt the value. (closes issue #18251) Reported by: bcnit + Patches: 20101110__issue18251.diff.txt uploaded by tilghman + (license 14) Tested by: trev, jthurman, elguero (closes issue + #18279) Reported by: zerohalo Patches: + 20101109__issue18279.diff.txt uploaded by tilghman (license 14) + Tested by: zerohalo ........ ................ ................ + + * /, funcs/func_dialgroup.c: Merged revisions 298478 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298478 | tilghman | 2010-12-16 02:56:13 -0600 + (Thu, 16 Dec 2010) | 15 lines Merged revisions 298477 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r298477 | tilghman | 2010-12-16 02:54:23 -0600 (Thu, 16 Dec 2010) + | 8 lines Eliminate duplicates from container. (closes issue + #18091) Reported by: bunny Patches: 20101006__issue18091.diff.txt + uploaded by tilghman (license 14) Tested by: bunny ........ + ................ + + * /, cdr/cdr_sqlite.c: Merged revisions 298394 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298394 | tilghman | 2010-12-15 18:30:04 -0600 + (Wed, 15 Dec 2010) | 22 lines Merged revisions 298393 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298393 | tilghman | 2010-12-15 18:29:10 -0600 + (Wed, 15 Dec 2010) | 15 lines Merged revisions 298392 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r298392 | tilghman | 2010-12-15 18:28:04 -0600 (Wed, 15 Dec 2010) + | 8 lines Unregister before shutting down the connection, to + avoid a race. (closes issue #18481) Reported by: pabelanger + Patches: 20101215__issue18481.diff.txt uploaded by tilghman + (license 14) Tested by: pabelanger ........ ................ + ................ + +2010-12-13 22:10 +0000 [r298201-298288] Richard Mudgett + + * channels/sig_pri.c: Post AMI hold events on PRI spans when the + remote party HOLD/RETRIEVEs the call. Part of JIRA + SWP-2687/ABE-2691. + + * channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions + 298195 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298195 | rmudgett | 2010-12-13 11:11:43 -0600 + (Mon, 13 Dec 2010) | 33 lines Merged revisions 298194 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r298194 | rmudgett | 2010-12-13 11:04:41 -0600 + (Mon, 13 Dec 2010) | 26 lines Merged revisions 298193 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010) + | 19 lines Outgoing PRI/BRI calls cannot do DTMF triggered + transfers. Outgoing PRI/BRI calls cannot do DTMF triggered + transfers if a PROCEEDING message is not received. The debug + output shows that the DTMF begin event is seen, but the DTMF end + event is missing. When the DTMF begin happens, the call is muted + so we now have one way audio (until a DTMF end event is somehow + seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER + event is received. * Made absorb the DTMF begin and DTMF end + events if we are overlap dialing and have not seen a PROCEEDING + message. * Added a debug message when absorbing a DTMF event. + JIRA SWP-2690 JIRA ABE-2697 ........ ................ + ................ + +2010-12-12 03:58 +0000 [r298137] Jeff Peeler + + * include/asterisk/utils.h, configure, + include/asterisk/autoconfig.h.in, configure.ac, main/logger.c, + main/utils.c, main/asterisk.c: Add support for several platforms + to obtain the real thread ID. Already had the pthread ID which is + not the same. The most obvious enhancement is in the "core show + threads" output. As stated in the utils header, if the platform + isn't supported -1 is reported (instead of the process ID + previously). + +2010-12-11 21:47 +0000 [r298100] Alexandr Anikin + + * addons/ooh323c/src/ooGkClient.c: Correction to work with + gatekeeper which don't send GK ID Don't use GK ID if it's not + presented in GK replies Extract GK ID not only in GK confirm but + in GK register confirm also (closes issue #18401) Reported by: + MrHanMan Patches: no-gkid-2.patch uploaded by may213 (license + 454) Tested by: may213, MrHanMan + +2010-12-10 16:53 +0000 [r298055] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 298054 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r298054 | mnicholson | 2010-12-10 10:52:11 -0600 (Fri, 10 Dec + 2010) | 2 lines Prevent a memcpy overlap in + GENERIC_FAX_EXEC_SET_VARS ........ + +2010-12-10 16:28 +0000 [r298052] Tilghman Lesher + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + main/netsock.c: Merged revisions 298051 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r298051 | tilghman | 2010-12-10 10:26:46 -0600 + (Fri, 10 Dec 2010) | 18 lines Merged revisions 298050 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r298050 | tilghman | 2010-12-10 10:24:13 -0600 (Fri, 10 Dec 2010) + | 11 lines Portability issue on OpenSolaris. Also detect the + required structure element, because OpenSolaris defines + SIOCGIFHWADDR, but without support for IP sockets. (closes issue + #18442) Reported by: ranjtech Patches: + 20101209__issue18442.diff.txt uploaded by tilghman (license 14) + Tested by: ranjtech ........ ................ + +2010-12-09 22:19 +0000 [r297972] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 297965 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297965 | twilson | 2010-12-09 16:18:19 -0600 + (Thu, 09 Dec 2010) | 28 lines Merged revisions 297960 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297960 | twilson | 2010-12-09 16:10:31 -0600 + (Thu, 09 Dec 2010) | 21 lines Merged revisions 297959 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297959 | twilson | 2010-12-09 16:00:30 -0600 (Thu, 09 Dec 2010) + | 14 lines Ignore spurious REGISTER requests If a REGISTER + request with a Call-ID matching an existing transaction is + received it was possible that the REGISTER request would + overwrite the initreq of the private structure. This info is used + to generate messages for other responses in the transaction. This + patch ignores REGISTER requests that match non-REGISTER + transactions. (closes issue #18051) Reported by: eeman Tested by: + twilson Review: https://reviewboard.asterisk.org/r/1050/ ........ + ................ ................ + +2010-12-09 21:33 +0000 [r297958] David Vossel + + * /, channels/chan_gtalk.c: Merged revisions 297957 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r297957 | dvossel | 2010-12-09 15:32:20 -0600 (Thu, 09 + Dec 2010) | 11 lines Fixes issue with outbound google voice calls + not working. Thanks to az1234 and nevermind_quack for their input + in helping debug the issue. (closes issue #18412) Reported by: + nevermind_quack Patches: fix uploaded by dvossel (license 671) + ........ + +2010-12-09 21:26 +0000 [r297956] Terry Wilson + + * /, main/features.c: Merged revisions 297952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r297952 | twilson | 2010-12-09 14:48:44 -0600 (Thu, 09 Dec 2010) + | 10 lines Don't crash after Set(CDR(userfield)=...) in + ast_bridge_call Instead of setting peer->cdr = NULL, set it to + not post. (closes issue #18415) Reported by: macbrody Patches: + patch-18415 uploaded by jsolares (license 1167) Tested by: + jsolares, twilson ........ + +2010-12-08 18:08 +0000 [r297910] Tilghman Lesher + + * /, configs/extensions.conf.sample: Merged revisions 297909 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297909 | tilghman | 2010-12-08 12:06:04 -0600 + (Wed, 08 Dec 2010) | 11 lines Merged revisions 297908 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r297908 | tilghman | 2010-12-08 12:04:38 -0600 (Wed, 08 Dec 2010) + | 4 lines Use inheritance to get correct results for + SIPFROMDOMAIN. (from an internal Digium discussion) ........ + ................ + +2010-12-07 23:00 +0000 [r297826] Jeff Peeler + + * main/channel.c, /: Merged revisions 297825 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297825 | jpeeler | 2010-12-07 16:59:30 -0600 + (Tue, 07 Dec 2010) | 26 lines Merged revisions 297824 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297824 | jpeeler | 2010-12-07 16:58:54 -0600 + (Tue, 07 Dec 2010) | 19 lines Merged revisions 297823 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297823 | jpeeler | 2010-12-07 16:57:48 -0600 (Tue, 07 Dec 2010) + | 12 lines Revert code that changed SSRC for DTMF. Some previous + behavior was attempted to be restored, but mistakingly I did not + realize that the previous behavior was incorrect. This fixes DTMF + not being detected since DTMF shouldn't cause the SSRC to change. + (related to issue #17404) (closes issue #18189) (closes issue + #18352) Reported by: marcbou Tested by: cmbaker82 ........ + ................ ................ + +2010-12-07 22:54 +0000 [r297734-297822] Tilghman Lesher + + * Makefile, contrib/init.d/org.asterisk.asterisk.plist, + utils/muted.c, /, contrib/init.d/org.asterisk.muted.plist + (added): Merged revisions 297821 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297821 | tilghman | 2010-12-07 16:51:05 -0600 + (Tue, 07 Dec 2010) | 18 lines Merged revisions 297819 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297819 | tilghman | 2010-12-07 16:40:45 -0600 + (Tue, 07 Dec 2010) | 11 lines Merged revisions 297818 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297818 | tilghman | 2010-12-07 16:35:50 -0600 (Tue, 07 Dec 2010) + | 4 lines Use non-deprecated APIs for CoreAudio Review: + https://reviewboard.asterisk.org/r/1040/ ........ + ................ ................ + + * /, apps/app_followme.c: Merged revisions 297733 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297733 | tilghman | 2010-12-06 18:29:26 -0600 + (Mon, 06 Dec 2010) | 22 lines Merged revisions 297713 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297713 | tilghman | 2010-12-06 18:21:50 -0600 + (Mon, 06 Dec 2010) | 15 lines Merged revisions 297689 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) + | 8 lines Don't create a Local channel if the target extension + does not exist. (closes issue #18126) Reported by: junky Patches: + followme.diff uploaded by junky (license 177) (partially + restructured by me to avoid a possible memory leak) ........ + ................ ................ + +2010-12-06 22:10 +0000 [r297608] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 297607 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297607 | jpeeler | 2010-12-06 16:06:37 -0600 + (Mon, 06 Dec 2010) | 25 lines Merged revisions 297605 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297605 | jpeeler | 2010-12-06 16:03:04 -0600 + (Mon, 06 Dec 2010) | 18 lines Merged revisions 297603 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297603 | jpeeler | 2010-12-06 15:57:15 -0600 (Mon, 06 Dec 2010) + | 12 lines Improve handling of REGISTER requests with multiple + contact headers. The changes here attempt to more strictly follow + RFC 3261 section 10.3. Basically the following will now cause a + 400 Bad Response to be returned, if: - multiple Contact headers + are present with one set to expire all bindings ("*") - wildcard + parameter is specified for Contact without Expires header or + Expires header is not set to zero. ABE-2442 ABE-2443 ........ + ................ ................ + +2010-12-03 17:42 +0000 [r297536] Sean Bright + + * /, channels/chan_console.c: Merged revisions 297535 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297535 | seanbright | 2010-12-03 12:41:30 -0500 + (Fri, 03 Dec 2010) | 9 lines Merged revisions 297534 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r297534 | seanbright | 2010-12-03 12:40:52 -0500 (Fri, + 03 Dec 2010) | 3 lines The CLI command should not contain + s, these are for descriptions. ........ + ................ + +2010-12-03 15:32 +0000 [r297496] Matthew Nicholson + + * /, res/res_fax.c, include/asterisk/res_fax.h: Merged revisions + 297157,297486,297495 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r297157 | mnicholson | 2010-12-01 13:47:33 -0600 (Wed, 01 Dec + 2010) | 2 lines Changed some NOTICE and WARNING messages to DEBUG + messages. ........ r297486 | mnicholson | 2010-12-02 15:30:47 + -0600 (Thu, 02 Dec 2010) | 6 lines Add support for reserving a + fax session before answering the channel. Note: this change + breaks ABI compatibility. FAX-217 ........ r297495 | mnicholson | + 2010-12-03 09:21:52 -0600 (Fri, 03 Dec 2010) | 4 lines Print a + DEBUG message instead of a WARNING message when the selected fax + tech does not support reserving sessions. Answer the channel + before quering it for t.38 support. This is necessary for the + query to work properly over local channels. ........ + +2010-12-02 20:11 +0000 [r297407] Paul Belanger + + * Makefile, /: Merged revisions 297406 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297406 | pabelanger | 2010-12-02 15:09:29 -0500 + (Thu, 02 Dec 2010) | 21 lines Merged revisions 297405 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297405 | pabelanger | 2010-12-02 15:06:43 -0500 + (Thu, 02 Dec 2010) | 14 lines Merged revisions 297404 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297404 | pabelanger | 2010-12-02 15:01:08 -0500 (Thu, 02 Dec + 2010) | 7 lines Resolve compile error under FreeBSD We now set + _ASTCFLAGS+=-march=i686 for i386 processors, still allowing + ASTCFLAGS to override the setting. Review: + https://reviewboard.asterisk.org/r/1043/ ........ + ................ ................ + +2010-12-02 18:28 +0000 [r297356] Terry Wilson + + * /, main/abstract_jb.c: Merged revisions 297312 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297312 | twilson | 2010-12-02 12:13:49 -0600 + (Thu, 02 Dec 2010) | 28 lines Merged revisions 297311 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297311 | twilson | 2010-12-02 12:07:39 -0600 + (Thu, 02 Dec 2010) | 21 lines Merged revisions 297310 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297310 | twilson | 2010-12-02 12:00:27 -0600 (Thu, 02 Dec 2010) + | 12 lines Initialize offset for adaptive jitter buffer When the + adaptive jitter buffer is enabled in sip.conf, the first frame + placed in the jitter buffer fails with something like: + jb_warning_output: Resyncing the jb. last_delay 0, this delay + -215886466, threshold 1000, new offset 215886466 This happens + because the offset is not initialized before calling jb_put(). + This patch modifies jb_put_first_adaptive() to set the offset to + the frame's timestamp. Review: + https://reviewboard.asterisk.org/r/1041/ ........ + ................ ................ + +2010-12-02 13:20 +0000 [r297248] Russell Bryant + + * /, apps/app_meetme.c: Merged revisions 297245 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297245 | russell | 2010-12-02 07:20:19 -0600 + (Thu, 02 Dec 2010) | 20 lines Merged revisions 297229 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297229 | russell | 2010-12-02 07:16:47 -0600 + (Thu, 02 Dec 2010) | 13 lines Merged revisions 297228 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) + | 6 lines Add "DAHDI" to a couple of app_meetme error messages. + This is in response to some questions on IRC. To the user, there + was nothing that made it obvious that this error had anything to + do with DAHDI not being loaded. ........ ................ + ................ + +2010-12-01 17:53 +0000 [r297076] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 297075 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r297075 | jpeeler | 2010-12-01 11:53:13 -0600 + (Wed, 01 Dec 2010) | 37 lines Merged revisions 297073 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r297073 | jpeeler | 2010-12-01 11:52:46 -0600 + (Wed, 01 Dec 2010) | 30 lines Merged revisions 297072 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r297072 | jpeeler | 2010-12-01 11:50:09 -0600 (Wed, 01 Dec 2010) + | 23 lines Fix not stopping MOH when transfered local channel + queue member is answered. The problem here is only present when + local channels are used with the MOH passthru option as well as + no optimization (/nm). I will describe the slightly bizarre + scenario that was used to test, where phones B and C are queue + members: Phone A dials into a queue with two members using local + channels and the above options. Phone B answers. Phone A blind + transfers phone B into the same queue. Phone A hangs up. Phone C + answers, but phone B didn't stop playing MOH. In this scenario, + the unhold frame that should have gotten to phone B never arrived + due to the masquerade from the blind transfer. This is usually + fine since app_queue manages the starting and stopping of MOH. + However, with the passthrough option enabled when app_queue + attempts to stop MOH it tries to do so on the local channel + rather than the real channel. The easiest solution was to just + make sure to send an unhold frame during the transfer since it + wouldn't make sense to have MOH playing after a transfer anyway. + This only modifies SIP transfers, but the other transfers did not + seem to be a problem. If DTMF based transfers were a problem it + might be okay to add ast_moh_stop to finishup, but I didn't want + to have to add that unless required. ABE-2624 ........ + ................ ................ + +2010-12-01 17:03 +0000 [r296952-296993] Tilghman Lesher + + * /, include/asterisk/frame.h: Merged revisions 296992 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296992 | tilghman | 2010-12-01 11:01:56 -0600 + (Wed, 01 Dec 2010) | 19 lines Merged revisions 296991 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296991 | tilghman | 2010-12-01 11:01:00 -0600 + (Wed, 01 Dec 2010) | 12 lines Merged revisions 296990 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296990 | tilghman | 2010-12-01 10:59:26 -0600 (Wed, 01 Dec 2010) + | 5 lines Clarify documentation on how we store codec preference + lists. (closes issue #18397) Reported by: birgita ........ + ................ ................ + + * /, channels/chan_iax2.c: Merged revisions 296951 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296951 | tilghman | 2010-11-30 19:46:32 -0600 + (Tue, 30 Nov 2010) | 9 lines Merged revisions 296950 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r296950 | tilghman | 2010-11-30 19:38:19 -0600 (Tue, 30 + Nov 2010) | 2 lines Missed initializations caused startup errors + on Mac OS X (and possibly others, too). ........ ................ + +2010-12-01 00:28 +0000 [r296871] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 296870 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296870 | jpeeler | 2010-11-30 18:28:16 -0600 + (Tue, 30 Nov 2010) | 18 lines Merged revisions 296869 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296869 | jpeeler | 2010-11-30 18:24:58 -0600 + (Tue, 30 Nov 2010) | 11 lines Merged revisions 296868 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) + | 4 lines Properly restore backup information file when hanging + up during message prepending. ABE-2654 ........ ................ + ................ + +2010-11-30 22:32 +0000 [r296788-296826] Tilghman Lesher + + * include/asterisk/frame.h: Add a comment on why the reserved bit + is reserved. Came up when reviewing discussion on the CODEC PREFS + IE in IAX2. + + * /, apps/app_meetme.c: Merged revisions 296787 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r296787 | tilghman | 2010-11-30 13:12:48 -0600 (Tue, 30 Nov 2010) + | 2 lines DOC: Conference number can be omitted; if omitted, all + users in a meetme are listed. ........ + +2010-11-30 09:49 +0000 [r296752] Stefan Schmidt + + * include/asterisk.h, main/pbx.c, main/asterisk.c: move devices + from hints into an ao2_container by splitting up devices from + hints into an own ao2_container the callback to get these devices + for statechange handling is faster. with this changes the length + of a device used in a hint isnt longer restricted to 80 + characters. Tests showed that calling handle_statechange is 40 + times faster if no hints are used and 25 times faster if there + are any hints. (closes issue #17928) Reported by: mdu113 Tested + by: schmidts Review: https://reviewboard.asterisk.org/r/1003/ + +2010-11-29 23:07 +0000 [r296674] Paul Belanger + + * /, channels/chan_iax2.c: Merged revisions 296673 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296673 | pabelanger | 2010-11-29 18:05:45 -0500 + (Mon, 29 Nov 2010) | 19 lines Merged revisions 296671 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296671 | pabelanger | 2010-11-29 17:54:14 -0500 + (Mon, 29 Nov 2010) | 12 lines Merged revisions 296670 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296670 | pabelanger | 2010-11-29 17:49:39 -0500 (Mon, 29 Nov + 2010) | 5 lines Make sure nothing else is needed before + destroying the scheduler. (closes issue #18398) Reported by: + pabelanger ........ ................ ................ + +2010-11-29 21:31 +0000 [r296630] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 296628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r296628 | russell | 2010-11-29 15:26:44 -0600 (Mon, 29 Nov 2010) + | 6 lines Complete some error handling in transmit_publish() in + chan_sip.c. This error handling block caught my eye. It was + missing a couple of things, but it should be safe now. Thanks to + mmichelson for the quick peer review on IRC. ........ + +2010-11-29 20:54 +0000 [r296585] Richard Mudgett + + * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: + Merged revisions 296582 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296582 | rmudgett | 2010-11-29 14:46:03 -0600 + (Mon, 29 Nov 2010) | 24 lines Merged revision 296575 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r296575 | rmudgett | 2010-11-29 14:27:37 -0600 (Mon, + 29 Nov 2010) | 13 lines Invalid mISDN PTMP redirecting signaling + as TE towards NT. The mISDN PTMP redirection signaling (NOTIFY + redirecting number and notification code, SETUP redirecting + number) is also sent in PTMP/TE mode. It should only apply in + PTMP/NT mode. The call setup proceeds but the network (Deutsche + Telekom) reacts with ugly ISDN STATUS messages. Also don't send + the redirecting number ie when PTP is also sending the + DivertingLegInformation2 facility. The redirecting number ie is + redundant and the network (Deutsche Telekom) complains about it. + Patches: abe_2651_v4.patch uploaded by rmudgett (license 664) + JIRA ABE-2651 JIRA SWP-2537 .......... ................ + +2010-11-29 07:30 +0000 [r296535] Tilghman Lesher + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + main/asterisk.c: Merged revisions 296534 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296534 | tilghman | 2010-11-29 01:28:44 -0600 + (Mon, 29 Nov 2010) | 20 lines Merged revisions 296533 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010) + | 13 lines I love standards. There are so many to choose from. + Except when there isn't one. Linux and *BSD disagree on the + elements within the ucred structure. Detect which one is in use + on the system. (closes issue #18384) Reported by: bjm Patches: + cred-diffs uploaded by bjm (license 473) + 20101127__issue18384__1.6.2.diff.txt uploaded by tilghman + (license 14) 20101127__issue18384__1.8.diff.txt uploaded by + tilghman (license 14) Tested by: tilghman, bjm ........ + ................ + +2010-11-27 10:41 +0000 [r296430-296468] Tilghman Lesher + + * /, apps/app_meetme.c: Merged revisions 296467 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296467 | tilghman | 2010-11-27 04:40:22 -0600 + (Sat, 27 Nov 2010) | 12 lines Merged revisions 296466 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010) + | 5 lines 18 characters is too short for most date/times (20 is + the usual, but we add more in case of greater precision). (closes + issue #18369) Reported by: tnakonz ........ ................ + + * include/asterisk.h, /: Merged revisions 296429 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r296429 | tilghman | 2010-11-27 03:58:57 -0600 (Sat, 27 Nov 2010) + | 5 lines Also don't build DEBUG_FD_LEAKS when STANDALONE2 is + defined. (closes issue #18385) Reported by: cmaj ........ + +2010-11-26 22:02 +0000 [r296393] Olle Johansson + + * /, main/say.c: Merged revisions 296391 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296391 | oej | 2010-11-26 22:37:21 +0100 (Fre, + 26 Nov 2010) | 24 lines Merged revisions 296351 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296351 | oej | 2010-11-26 13:23:03 +0100 (Fre, + 26 Nov 2010) | 17 lines Merged revisions 296309 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296309 | oej | 2010-11-26 10:53:31 +0100 (Fre, 26 Nov 2010) | 11 + lines Fix bugs in saying numbers using the Swedish language + syntax (closes issue #18355) Reported by: oej Patch by: oej Much + help from Peter Lindahl. Testing by the ClearIT team during a + coffee break. Review: https://reviewboard.asterisk.org/r/1033/ + ........ ................ ................ + +2010-11-26 18:31 +0000 [r296353-296355] Brad Watkins + + * /, res/res_jabber.c: Merged revisions 296354 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r296354 | marquis | 2010-11-26 13:31:17 -0500 (Fri, 26 Nov 2010) + | 12 lines Fix XMPP PubSub-based distributed device state. + Initialize pubsubflags to 0 so res_jabber doesn't think there is + already an XMPP connection sending device state. Also clean up + CLI commands a bit. (closes issue #18272) Reported by: klaus3000 + Patches: res_jabber_fix_pubsubflags_and_CLI-patch.txt uploaded by + klaus3000 (license 65) Tested by: klaus3000, Marquis Review: + https://reviewboard.asterisk.org/r/1030/ ........ + + * /, channels/chan_sip.c: Merged revisions 296352 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r296352 | marquis | 2010-11-26 13:19:02 -0500 (Fri, 26 Nov 2010) + | 12 lines Fix reloading of peer when a user is requested. + Prevent peer reloading from causing multiple MWI subscriptions to + be created when using realtime. This had the effect of sending + one NOTIFY for every time a sip peer made a call, in one case + eventually overwhelming the phone and causing it to reboot. + (closes issue #18342) Reported by: nivek Patches: + issue0018342p1.patch uploaded by nivek (license 636) Tested by: + nivek Review: https://reviewboard.asterisk.org/r/1029/ ........ + +2010-11-24 23:46 +0000 [r296249] Andrew Parisio + + * apps/app_meetme.c, CHANGES: Meetme use voicemail greet for + join/leave announce Added option v(mailbox@[context]) which tells + MeetMe where to look for a users greet file. If one does not + exist it clears the v option and defers to the functionality of + i/I as/if set by the MeetMe() command. Review: + https://reviewboard.asterisk.org/r/1009/ (closes issue #18297) + Reported by: parisioa Patches: meetme_final_patch_v.diff uploaded + by parisioa (license 1153) + +2010-11-24 23:30 +0000 [r296235] Russell Bryant + + * main/channel.c, /: Merged revisions 296230 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296230 | russell | 2010-11-24 17:29:44 -0600 + (Wed, 24 Nov 2010) | 20 lines Merged revisions 296221 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296221 | russell | 2010-11-24 17:28:19 -0600 + (Wed, 24 Nov 2010) | 13 lines Merged revisions 296213 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296213 | russell | 2010-11-24 17:26:43 -0600 (Wed, 24 Nov 2010) + | 6 lines Make Asterisk less crashy. Since we might not put a new + translation path on the channel, go ahead and set it to NULL + right after destroying the old one to ensure we don't try to free + an invalid translation path later on. ........ ................ + ................ + +2010-11-24 22:52 +0000 [r296168] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, channels/sig_analog.c, + /, channels/sig_analog.h: Merged revisions 296167 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296167 | rmudgett | 2010-11-24 16:49:48 -0600 + (Wed, 24 Nov 2010) | 57 lines Merged revisions 296166 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296166 | rmudgett | 2010-11-24 16:42:45 -0600 + (Wed, 24 Nov 2010) | 50 lines Merged revisions 296165 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) + | 43 lines Oneway audio to SIP phone from FXS port after FXS port + gets a CallWaiting pip. The FXS connected phone has to have + CW/CID support to fail, as it will send back a DTMF 'A' or 'D' + when it's ready to receive CallerID. A normal phone with no CID + never fails. Also the SIP phone does not hear MOH when the CW + call is answered. The DTMF end frame is suppressed when the phone + acknowledges the CW signal for CID. The problem is the DTMF begin + frame needs to be suppressed as well. The DTMF begin frame is + causing SIP to start sending the DTMF RTP frames. Since the DTMF + end frame is suppressed, SIP will not stop sending those DTMF RTP + packets. * Suppress the DTMF begin and end frames when the + channel driver is looking for DTMF digits. * Fixed a couple + issues caused by not cleaning up the CID spill if you answer the + CW call while it is sending the CID spill. * Fixed not sending + CW/CID spill to the phone when the call is natively bridged. + (Fixed by not using native bridge if CW/CID is possible.) * + Suppress received audio when sending CW/CID spills. The other + parties involved do not need to hear the CW/CID spills and may be + confused if the CW call is for them. (closes issue #18129) + Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch + uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett + NOTE: * v1.4 does not have the main problem fixed by suppressing + the DTMF start frames. The other three items fixed are relevant. + * If you really must restore native bridging between analog + ports, you need to disable CW/CID either by configuring + chan_dahdi.conf callwaitingcallerid=no or dialing *70 before + dialing the number to temporarily disable CW. ........ + ................ ................ + +2010-11-24 20:24 +0000 [r296034-296085] Russell Bryant + + * main/channel.c, /: Merged revisions 296084 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296084 | russell | 2010-11-24 14:23:46 -0600 + (Wed, 24 Nov 2010) | 26 lines Merged revisions 296083 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296083 | russell | 2010-11-24 14:23:11 -0600 + (Wed, 24 Nov 2010) | 19 lines Merged revisions 296082 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296082 | russell | 2010-11-24 14:22:32 -0600 (Wed, 24 Nov 2010) + | 12 lines Fix false reporting of an error by set_format(). In + the case that the native format was able to be changed to match + the new requested format, the code proceeded to attempt to build + a translation path, anyway. The result would be NULL, since no + translation path is necessary and resulted in this function + thinking an error has occurred. This case is now specifically + caught and no attempt to build a translation path is attempted. + Thanks to our automated tests and bamboo.asterisk.org for + catching this problem and making a whole lot of noise when things + started failing. :-) ........ ................ ................ + + * apps/app_dial.c, main/channel.c, /: Merged revisions 296002 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r296002 | russell | 2010-11-24 11:13:08 -0600 + (Wed, 24 Nov 2010) | 52 lines Merged revisions 296001 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r296001 | russell | 2010-11-24 11:03:16 -0600 + (Wed, 24 Nov 2010) | 45 lines Merged revisions 296000 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r296000 | russell | 2010-11-24 10:48:39 -0600 (Wed, 24 Nov 2010) + | 38 lines Handle failures building translation paths more + effectively. The problem scenario occurred on a heavily loaded + system that was using the codec_dahdi module and exceeded the + hardware transcoding capacity. The failure mode at that point was + not good. The report came in to us as an Asterisk lock-up. The + "core show locks" shows a ton of threads locked up (but no + obvious deadlock). Upon deeper investigation, when the system is + in this state, the CPU was maxed out. The CPU was being consumed + by the Asterisk logger spewing messages on every audio frame for + calls set up after transcoder capacity was reached. The purpose + of this patch is to make Asterisk handle failures to create a + translation path in a more graceful manner. If we can't + translate, then the call just needs to be dropped, as it's not + going to work. These are the changes: 1) In set_format() of + channel.c (which is called by set_read_format() and + set_write_format()), it was ignoring if + ast_translator_build_path() failed and returned NULL. It now pays + attention to that case and returns a result reflecting failure. + With this change in place, the bridging code will immediately + detect a failure and end the bridge instead of proceeding to try + to bridge frames that can't be translated and making channel + drivers freak out by sending them frames in a format they weren't + expecting. 2) In ast_indicate_data() of channel.c, failure of + ast_playtones_start() was ignored. It is now reflected in the + return value of the function. This didn't turn out to have any + affect on the bug, but seemed like a good change to leave in. 3) + In app_dial(), when only sending a call to a single endpoint, it + will attempt to do some bridging of its own of early audio. It + uses make_compatible() when it's going to do this. However, it + ignored failure from make compatible. So, even with the fix from + #1, if there was early audio going through app_dial, there would + still be a period of invalid frames passing through. After + detecting failure here, Dial() exits. ABE-2658 ........ + ................ ................ + +2010-11-23 10:34 +0000 [r295950] Olle Johansson + + * /, main/say.c: Merged revisions 295949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295949 | oej | 2010-11-23 11:30:05 +0100 (Tis, + 23 Nov 2010) | 21 lines Merged revisions 295907 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r295907 | oej | 2010-11-23 10:36:38 +0100 (Tis, + 23 Nov 2010) | 14 lines Merged revisions 295906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r295906 | oej | 2010-11-23 10:28:14 +0100 (Tis, 23 Nov 2010) | 8 + lines Fix support of saynumber(1,n) in the Swedish language + (closes issue #18353) Reported by: oej Review: + https://reviewboard.asterisk.org/r/1031/ ........ + ................ ................ + +2010-11-22 20:05 +0000 [r295870] Sean Bright + + * configs/chan_dahdi.conf.sample, /: Merged revisions 295869 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295869 | seanbright | 2010-11-22 15:03:49 -0500 + (Mon, 22 Nov 2010) | 9 lines Merged revisions 295868 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r295868 | seanbright | 2010-11-22 15:02:37 -0500 (Mon, + 22 Nov 2010) | 2 lines Change some documentation to suggest + dahdi_monitor instead of ztmonitor. ........ ................ + +2010-11-22 19:42 +0000 [r295867] Richard Mudgett + + * main/channel.c, main/pbx.c, /, apps/app_macro.c, + include/asterisk/channel.h, include/asterisk/frame.h: Merged + revisions 295866 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295866 | rmudgett | 2010-11-22 13:36:10 -0600 + (Mon, 22 Nov 2010) | 60 lines Merged revisions 295843 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r295843 | rmudgett | 2010-11-22 13:28:23 -0600 + (Mon, 22 Nov 2010) | 53 lines Merged revisions 295790 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) + | 46 lines The channel redirect function (CLI or AMI) hangs up + the call instead of redirecting the call. To recreate the + problem: 1) Party A calls Party B 2) Invoke CLI "channel + redirect" command to redirect channel call leg associated with A. + 3) All associated channels are hung up. Note that if the CLI + command were done on the channel call leg associated with B it + works. This regression was a result of the fix for issue #16946 + (https://reviewboard.asterisk.org/r/740/). The regression affects + all features that use an async goto to execute the dialplan + because of an external event: Channel redirect, AMI redirect, SIP + REFER, and FAX detection. The struct ast_channel._softhangup code + is a mess. The variable is used for several purposes that do not + necessarily result in the call being hung up. I have added + doxygen comments to describe how the various _softhangup bits are + used. I have corrected all the places where the variable was + tested in a non-bit oriented manner. The primary fix is the new + AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so + the soft hangup requests that do not normally result in a hangup + do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) + Reported by: SantaFox (closes issue #18185) Reported by: + kwemheuer (closes issue #18211) Reported by: zahir_koradia + (closes issue #18230) Reported by: vmarrone (closes issue #18299) + Reported by: mbrevda (closes issue #18322) Reported by: nerbos + Review: https://reviewboard.asterisk.org/r/1013/ ........ + ................ ................ + +2010-11-22 18:43 +0000 [r295789] Erin Spiceland + + * res/res_agi.c: Revert to the previous behavior of AGI command + WAIT FOR DIGIT, since the behavior of the command with this patch + is almost exactly like that of GET DATA. + +2010-11-20 03:13 +0000 [r295748] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Merged revisions 295747 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295747 | rmudgett | 2010-11-19 21:11:15 -0600 (Fri, 19 Nov 2010) + | 13 lines One way audio before answering call waiting call on + analog port. * Analog call waiting Caller ID spills could get + stuck resulting in one way audio until the waiting call is + answered. This only happens on the second (and later) call + waiting call if the active call is not the first call. * The + CLI/AMI "dahdi show channel" command could report the wrong + channel information. Must keep the struct analog_pvt.owner and + struct dahdi_pvt.owner pointer in sync. ........ + +2010-11-20 00:52 +0000 [r295712] Russell Bryant + + * include/asterisk/event.h, /, main/event.c: Merged revisions + 295711 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295711 | russell | 2010-11-19 18:50:00 -0600 + (Fri, 19 Nov 2010) | 36 lines Merged revisions 295710 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) + | 29 lines Fix cache of device state changes for multiple + servers. This patch addresses a regression where device states + across multiple servers were not being processing completely + correctly. The code works to determine the overall state by + looking at the last known state of a device on each server. + However, there was a regression due to some invasive rewrites of + how the cache works that led to the cache only storing the last + device state change for a device, regardless of which server it + was on. The code is set up to cache device state change events by + ensuring that each event in the cache has a unique device name + + entity ID (server ID). The code that was responsible for + comparing raw information elements (which EID is) always returned + a match due to a memcmp() with a length of 0. There isn't much + code to fix the actual bug. This patch also introduces a new CLI + command that was very useful for debugging this problem. The + command allows you to dump the contents of the event cache. + (closes issue #18284) Reported by: klaus3000 Patches: + issue18284.rev1.txt uploaded by russell (license 2) Tested by: + russell, klaus3000 (closes issue #18280) Reported by: klaus3000 + Review: https://reviewboard.asterisk.org/r/1012/ ........ + ................ + +2010-11-19 22:15 +0000 [r295674] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 295673 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295673 | twilson | 2010-11-19 14:06:10 -0800 + (Fri, 19 Nov 2010) | 22 lines Merged revisions 295672 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r295672 | twilson | 2010-11-19 13:55:48 -0800 + (Fri, 19 Nov 2010) | 15 lines Merged revisions 295628 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r295628 | twilson | 2010-11-19 12:53:36 -0800 (Fri, 19 Nov 2010) + | 8 lines Discard responses with more than one Via This is not a + perfect solution as headers that are joined via commas are not + detected. This is a parsing issue that to fix "correctly" would + necessitate a new SIP parser. Review: + https://reviewboard.asterisk.org/r/1019/ ........ + ................ ................ + +2010-11-19 21:42 +0000 [r295671] Brett Bryant + + * /, apps/app_queue.c: Merged revisions 295670 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295670 | bbryant | 2010-11-19 16:40:21 -0500 (Fri, 19 Nov 2010) + | 8 lines Patch for deadlock from ordering issue between + channel/queue locks in app_queue (set_queue_variables). (closes + issue #18031) Reported by: rain Review: + https://reviewboard.asterisk.org/r/1018/ ........ + +2010-11-19 19:32 +0000 [r295554] Erin Spiceland + + * res/res_agi.c: Add extra functionality to AGI command WAIT FOR + DIGIT. Add the ability to play a sound file, listen for more than + just one digit, specify escape characters. Backwards compatible + (to work with only timeout specified). (closes issue #15531) + Reported by: diLLec Patches: + asterisk-res_agi-203638-patched.patch uploaded by diLLec (license + 839) Tested by: diLLec, espiceland + +2010-11-19 16:49 +0000 [r295517] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Merged revisions 295516 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295516 | rmudgett | 2010-11-19 10:47:11 -0600 (Fri, 19 Nov 2010) + | 13 lines Bring sig_analog extraction more into alignment with + orig-trunk/v1.6.2 chan_dahdi. * Restore SMDI support. * Fixed + initial value of struct analog_pvt.use_callerid. It may get + forced on depending upon other config options. * Call + analog_dnd() instead of manual inlined code. * Removed unused + struct analog_pvt.usedistinctiveringdetection. * Removed the + struct analog_pvt.unknown_alarm flag. It was really the struct + analog_pvt.inalarm flag. * Use ast_debug() instead of + ast_log(LOG_DEBUG). * Rename several function's index variable to + idx. * Some formatting tweaks. ........ + +2010-11-18 20:31 +0000 [r295478] Leif Madsen + + * configs/sip_notify.conf.sample, /: Merged revisions 295477 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295477 | lmadsen | 2010-11-18 14:30:35 -0600 (Thu, 18 Nov 2010) + | 6 lines 'sip notify clear-mwi' needs terminating CRLF. (closes + issue #18275) Reported by: klaus3000 Patches: + fix_body_CRLF_patch.txt uploaded by klaus3000 (license 65) + ........ + +2010-11-18 18:08 +0000 [r295364-295442] Paul Belanger + + * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions + 295441 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295441 | pabelanger | 2010-11-18 13:02:12 -0500 + (Thu, 18 Nov 2010) | 11 lines Merged revisions 295440 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r295440 | pabelanger | 2010-11-18 12:51:34 -0500 (Thu, 18 Nov + 2010) | 4 lines Fix compiler warnings when using openssl-dev + 1.0.0+ Review: https://reviewboard.asterisk.org/r/1016/ ........ + ................ + + * /, contrib/scripts/install_prereq: Merged revisions 295404 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295404 | pabelanger | 2010-11-18 00:12:05 -0500 (Thu, 18 Nov + 2010) | 2 lines Add RedHat specific dependencies ........ + + * /, configs/res_curl.conf.sample: Merged revisions 295361 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r295361 | pabelanger | 2010-11-17 09:09:38 -0500 (Wed, 17 Nov + 2010) | 2 lines Uncomment settings under [global], to surpress + warning when loading Asterisk. ........ + +2010-11-16 23:04 +0000 [r295283] Richard Mudgett + + * main/channel.c, /: Merged revisions 295282 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295282 | rmudgett | 2010-11-16 17:02:36 -0600 + (Tue, 16 Nov 2010) | 16 lines Merged revisions 295281 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r295281 | rmudgett | 2010-11-16 16:57:07 -0600 + (Tue, 16 Nov 2010) | 9 lines Merged revisions 295280 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r295280 | rmudgett | 2010-11-16 16:52:06 -0600 (Tue, 16 + Nov 2010) | 1 line Dead code elimination in + channel.c:ast_channel_bridge() variable who. ........ + ................ ................ + +2010-11-16 22:41 +0000 [r295125-295279] Russell Bryant + + * /, build_tools/prep_tarball: Merged revisions 295278 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r295278 | russell | 2010-11-16 16:41:11 -0600 (Tue, 16 + Nov 2010) | 2 lines Check for pdftotext and give a useful error + if not found. ........ + + * /, build_tools/prep_tarball: Merged revisions 295201 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r295201 | russell | 2010-11-16 15:46:18 -0600 (Tue, 16 + Nov 2010) | 2 lines Remove intentional typo I had added when + testing the check. oops. ........ + + * /, build_tools/prep_tarball: Merged revisions 295164 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r295164 | russell | 2010-11-16 14:50:03 -0600 (Tue, 16 + Nov 2010) | 2 lines Check for wikiexport.py in PATH and give a + useful error message if not found. ........ + + * main/app.c: Remove a trailing space. (testing something with + bamboo ...) + +2010-11-15 19:11 +0000 [r294990-295079] Tilghman Lesher + + * tests/test_expr.c (added), /: Merged revisions 295078 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r295078 | tilghman | 2010-11-15 12:30:13 -0600 + (Mon, 15 Nov 2010) | 16 lines Merged revisions 295062 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r295062 | tilghman | 2010-11-15 12:24:02 -0600 + (Mon, 15 Nov 2010) | 9 lines Merged revisions 295026 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r295026 | tilghman | 2010-11-15 11:58:37 -0600 (Mon, 15 + Nov 2010) | 2 lines Create test verifying results of expression + parser ........ ................ ................ + + * funcs/func_curl.c, /: Merged revisions 294989 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294989 | tilghman | 2010-11-15 01:44:38 -0600 + (Mon, 15 Nov 2010) | 15 lines Merged revisions 294988 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010) + | 8 lines It is possible to crash Asterisk by feeding the curl + engine invalid data. (closes issue #18161) Reported by: wdoekes + Patches: 20101029__issue18161.diff.txt uploaded by tilghman + (license 14) Tested by: tilghman ........ ................ + +2010-11-12 21:15 +0000 [r294907-294912] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 294911 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294911 | jpeeler | 2010-11-12 15:14:43 -0600 + (Fri, 12 Nov 2010) | 11 lines Merged revisions 294910 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294910 | jpeeler | 2010-11-12 15:14:23 -0600 (Fri, 12 Nov 2010) + | 4 lines Return correct error code if lock path fails. The + recent changes to open_mailbox actually caused it to be fixed, + but let's be consistent. Reported by alecdavis in asterisk-dev. + ........ ................ + + * /, apps/app_voicemail.c: Merged revisions 294905 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294905 | jpeeler | 2010-11-12 14:52:06 -0600 + (Fri, 12 Nov 2010) | 30 lines Merged revisions 294904 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r294904 | jpeeler | 2010-11-12 14:51:15 -0600 + (Fri, 12 Nov 2010) | 23 lines Merged revisions 294903 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) + | 16 lines Fix regression causing abort in voicemail after + opening a mailbox with no mesgs. In order to be more safe, some + error handling code was changed to respect more error conditions + including the potential memory allocation failure for deleted and + heard message tracking introduced in 293004. However, + last_message_index returns -1 for zero messages (perhaps as + expected) and was triggering the stricter error checking. Because + last_message_index is only called directly in one place, just + return 0 from open_mailbox (for file based storage) when no + messages are detected unless a real error has occurred. (closes + issue #18240) Reported by: leobrown Patches: + bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) + Tested by: pabelanger ........ ................ ................ + +2010-11-12 02:46 +0000 [r294824] Richard Mudgett + + * channels/sig_pri.h, /, channels/sig_pri.c: Merged revisions + 294823 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294823 | rmudgett | 2010-11-11 20:45:22 -0600 + (Thu, 11 Nov 2010) | 25 lines Merged revisions 294822 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r294822 | rmudgett | 2010-11-11 20:44:12 -0600 + (Thu, 11 Nov 2010) | 18 lines Merged revisions 294821 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) + | 11 lines Asterisk is getting a "No D-channels available!" + warning message every 4 seconds. Asterisk is just whining too + much with this message: "No D-channels available! Using Primary + channel XXX as D-channel anyway!". Filtered the message so it + only comes out once if there is no D channel available without an + intervening D channel available period. (closes issue #17270) + Reported by: jmls ........ ................ ................ + +2010-11-11 22:18 +0000 [r294741-294749] Russell Bryant + + * /, doc/CCSS_architecture.pdf (removed): Merged revisions 294745 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294745 | russell | 2010-11-11 16:17:57 -0600 (Thu, 11 Nov 2010) + | 6 lines Remove CCSS architecture PDF. It has been moved to: + https://wiki.asterisk.org/wiki/display/AST/CCSS+Architecture + ........ + + * doc/CODING-GUIDELINES (removed), doc/ss7.txt (removed), /, + doc/backtrace.txt (removed), doc/India-CID.txt (removed), + doc/digium-mib.txt (removed), doc/followme.txt (removed), + doc/building_queues.txt (removed), doc/timing.txt (removed), + doc/advice_of_charge.txt (removed), doc/unistim.txt (removed), + doc/video_console.txt (removed), doc/macroexclusive.txt + (removed), doc/google-soc2009-ideas.txt (removed), doc/README.txt + (added), doc/callfiles.txt (removed), build_tools/prep_tarball, + doc/codec-64bit.txt (removed), doc/externalivr.txt (removed), + doc/video.txt (removed), doc/jingle.txt (removed), + doc/modules.txt (removed), doc/manager_1_1.txt (removed), + doc/PEERING (removed), doc/snmp.txt (removed), doc/siptls.txt + (removed), doc/HOWTO_collect_debug_information.txt (removed), + doc/ldap.txt (removed), doc/sip-retransmit.txt (removed), + doc/distributed_devstate.txt (removed), + doc/voicemail_odbc_postgresql.txt (removed), doc/tex (removed), + doc/queue.txt (removed), doc/jabber.txt (removed), + doc/chan_sip-perf-testing.txt (removed), doc/asterisk-mib.txt + (removed), Makefile, doc/database_transactions.txt (removed), + doc/smdi.txt (removed), doc/janitor-projects.txt (removed), + doc/hoard.txt (removed), doc/res_config_sqlite.txt (removed), + doc/osp.txt (removed), doc/speechrec.txt (removed), doc/sms.txt + (removed), doc/distributed_devstate-XMPP.txt (removed), + doc/valgrind.txt (removed), doc/realtimetext.txt (removed), + doc/cli.txt (removed), doc/rtp-packetization.txt (removed), + doc/datastores.txt (removed): Merged revisions 294740 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294740 | russell | 2010-11-11 16:13:38 -0600 (Thu, 11 Nov 2010) + | 11 lines Remove most of the contents of the doc dir in favor of + the wiki content. This merge does the following things: * Removes + most of the contents from the doc/ directory in favor of the wiki + - http://wiki.asterisk.org/ * Updates the + build_tools/prep_tarball script to know how to export the + contents of the wiki in both PDF and plain text formats so that + the documentation is still included in Asterisk release tarballs. + ........ + +2010-11-11 22:01 +0000 [r294735] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 294734 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294734 | jpeeler | 2010-11-11 15:58:25 -0600 + (Thu, 11 Nov 2010) | 32 lines Merged revisions 294733 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r294733 | jpeeler | 2010-11-11 15:57:22 -0600 + (Thu, 11 Nov 2010) | 25 lines Merged revisions 294688 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) + | 18 lines Fix problem with qualify option packets for realtime + peers never stopping. The option packets not only never stopped, + but if a realtime peer was not in the peer list multiple options + dialogs could accumulate over time. This scenario has the + potential to progress to the point of saturating a link just from + options packets. The fix was to ensure that the poke scheduler + checks to see if a peer is in the peer list before continuing to + poke. The reason a peer must be in the peer list to be able to + properly manage an options dialog is because otherwise the call + pointer is lost when the peer is regenerated from the database, + which is how existing qualify dialogs are detected. (closes issue + #16382) (closes issue #17779) Reported by: lftsy Patches: + bug16382-3.patch uploaded by jpeeler (license 325) Tested by: + zerohalo ........ ................ ................ + +2010-11-10 23:27 +0000 [r294570-294606] Tilghman Lesher + + * pbx/pbx_spool.c, /: Merged revisions 294605 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294605 | tilghman | 2010-11-10 17:26:39 -0600 (Wed, 10 Nov 2010) + | 2 lines Fixing the Mac OS X build (bamboo warning) ........ + + * pbx/pbx_spool.c, /: Merged revisions 294569 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294569 | tilghman | 2010-11-10 17:13:37 -0600 (Wed, 10 Nov 2010) + | 8 lines Properly queue files with inotify(7). (closes issue + #18089) Reported by: abelbeck Patches: + 20101021__issue18089.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman ........ + +2010-11-10 14:15 +0000 [r294502-294536] Russell Bryant + + * /, res/ais/clm.c, res/ais/evt.c, UPGRADE-1.8.txt: Merged + revisions 294535 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294535 | russell | 2010-11-10 08:14:51 -0600 (Wed, 10 Nov 2010) + | 5 lines Tweak a couple of CLI commands back to their original + form. The "module" in this case is two parts, so there are two + words before the verb of the CLI command. ........ + + * /, main/devicestate.c: Merged revisions 294501 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294501 | russell | 2010-11-10 06:46:27 -0600 + (Wed, 10 Nov 2010) | 14 lines Merged revisions 294500 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294500 | russell | 2010-11-10 06:41:41 -0600 (Wed, 10 Nov 2010) + | 7 lines Improve a debug message to be more readable and + consistent. (closes issue #18282) Reported by: klaus3000 Patches: + ast_devstate2str-patch.txt uploaded by klaus3000 (license 65) + ........ ................ + +2010-11-09 22:52 +0000 [r294467] Richard Mudgett + + * main/channel.c, /: Merged revisions 294466 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294466 | rmudgett | 2010-11-09 16:46:45 -0600 (Tue, 09 Nov 2010) + | 1 line Allow ast_do_masquerade() failure to be reported again. + ........ + +2010-11-09 20:35 +0000 [r294431] Tilghman Lesher + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 294430 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294430 | tilghman | 2010-11-09 14:33:05 -0600 + (Tue, 09 Nov 2010) | 15 lines Merged revisions 294429 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294429 | tilghman | 2010-11-09 14:27:23 -0600 (Tue, 09 Nov 2010) + | 8 lines Detect GMime properly on systems where gmime flags and + libs are configured with pkg-config. (closes issue #16155) + Reported by: jcollie Patches: 20100917__issue16155.diff.txt + uploaded by tilghman (license 14) Tested by: tilghman ........ + ................ + +2010-11-09 17:00 +0000 [r294351] Richard Mudgett + + * main/channel.c, channels/chan_misdn.c, channels/sig_analog.c, /, + include/asterisk/channel.h, channels/sig_pri.c: Merged revisions + 294349 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294349 | rmudgett | 2010-11-09 10:55:32 -0600 (Tue, 09 Nov 2010) + | 17 lines Analog lines do not transfer CONNECTED LINE or execute + the interception macros. Add connected line update for sig_analog + transfers and simplify the corresponding sig_pri and chan_misdn + transfer code. Note that if you create a three-way call in + sig_analog before transferring the call, the distinction of the + caller/callee interception macros make little sense. The + interception macro writer needs to be prepared for either + caller/callee macro to be executed. The current implementation + swaps which caller/callee interception macro is executed after a + three-way call is created. Review: + https://reviewboard.asterisk.org/r/996/ JIRA ABE-2589 JIRA + SWP-2372 ........ + +2010-11-08 22:33 +0000 [r294279-294314] Jeff Peeler + + * /, res/res_timing_timerfd.c: Merged revisions 294313 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294313 | jpeeler | 2010-11-08 16:32:13 -0600 + (Mon, 08 Nov 2010) | 9 lines Merged revisions 294312 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r294312 | jpeeler | 2010-11-08 16:30:49 -0600 (Mon, 08 + Nov 2010) | 1 line add missing unlock not present in 294277 + ........ ................ + + * main/channel.c, /, res/res_timing_timerfd.c, + include/asterisk/timing.h, main/timing.c: Merged revisions 294278 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294278 | jpeeler | 2010-11-08 15:59:45 -0600 + (Mon, 08 Nov 2010) | 23 lines Merged revisions 294277 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) + | 16 lines Fix playback failure when using IAX with the timerfd + module. To fix this issue the alert pipe will now be used when + the timerfd module is in use. There appeared to be a race that + was not solved by adding locking in the timerfd module, but + needed to be there anyway. The race was between the timer being + put in non-continuous mode in ast_read on the channel thread and + the IAX frame scheduler queuing a frame which would enable + continuous mode before the non-continuous mode event was read. + This race for now is simply avoided. (closes issue #18110) + Reported by: tpanton Tested by: tpanton I put tested by tpanton + because it was tested on his hardware. Thanks for the remote + access to debug this issue! ........ ................ + +2010-11-08 21:04 +0000 [r294244] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 294243 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r294243 | mnicholson | 2010-11-08 14:56:30 -0600 + (Mon, 08 Nov 2010) | 15 lines Merged revisions 294242 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r294242 | mnicholson | 2010-11-08 14:50:21 -0600 (Mon, 08 Nov + 2010) | 8 lines Go off hold when we get an empty reinvite telling + us to. (closes issue 0014448) Reported by: frawd (closes issue + #17878) Reported by: frawd ........ ................ + +2010-11-08 19:59 +0000 [r294208] Terry Wilson + + * /, configs/calendar.conf.sample, res/res_calendar.c: Merged + revisions 294207 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294207 | twilson | 2010-11-08 13:56:10 -0600 (Mon, 08 Nov 2010) + | 2 lines Set a default waittime, and make sure to convert it to + milliseconds ........ + +2010-11-08 17:19 +0000 [r294127] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 294125 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r294125 | rmudgett | 2010-11-08 11:16:01 -0600 (Mon, 08 + Nov 2010) | 33 lines valgrind reported references to freed memory + during a mISDN hangup collision. Bad things have been happening + in chan_misdn because the chan_misdn channel private struct + chan_list is not protected from reentrancy. Hangup collisions + have be causing read and write accesses to freed memory. + Converted chan_misdn struct chan_list to an ao2 object for its + reference counting feature. ********** Removed an impediment to + converting chan_list to an ao2 object. The use of the other_ch + member in chan_list is shaky at best. It is set if the incoming + and outgoing call legs are mISDN. The use of the other_ch member + goes against the Asterisk architecture and can even cause + problems. 1) It is used to disable echo cancellation. This could + be bad if the call is forked and the winning call leg is not + mISDN or the winning call leg is not the last mISDN channel + called by the fork. The other_ch would become a dangling pointer. + 2) It is used when the far end is alerting to hear the far end's + inband audio instead of Asterisk's generated ringback tone. This + is bad if the call is forked. You would only hear the last forked + mISDN channel and it may not be ringing yet. The other_ch would + become a dangling pointer if the call is later transferred. + ********** JIRA SWP-2423 JIRA ABE-2614 ........ + +2010-11-05 22:17 +0000 [r294086] Brett Bryant + + * /, channels/chan_sip.c: Merged revisions 294084 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294084 | bbryant | 2010-11-05 18:03:11 -0400 (Fri, 05 Nov 2010) + | 9 lines Fixed deadlock avoidance issues while locking channel + when adding the Max-Forwards header to a request. (closes issue + #17949) (closes issue #18200) Reported by: bwg Review: + https://reviewboard.asterisk.org/r/997/ ........ + +2010-11-05 21:56 +0000 [r294083] David Vossel + + * channels/chan_sip.c: Perform proper handling of forked outbound + INVITE requests. RFC3261 section 12 about dialog creation says an + INVITE transaction results in an established dialog once it + receives the 200 OK response. It is possible to receive multiple + differing 200 OK responses for a single outbound INVITE Request, + and this should result in establishing multiple dialogs. This + patch allows for all differing 200 OK responses to an INVITE + request to establish a separate dialog, but only the first dialog + is kept. All other resulting dialogs from the initial request are + immediately ACKed and then immediately terminated with a BYE + request. Review: https://reviewboard.asterisk.org/r/946/ + +2010-11-05 16:07 +0000 [r294048-294050] Terry Wilson + + * contrib/scripts/ast_tls_cert, /: Merged revisions 294049 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294049 | twilson | 2010-11-05 09:05:50 -0700 (Fri, 05 Nov 2010) + | 2 lines Corret spelling and example ........ + + * contrib/scripts/ast_tls_cert, /: Merged revisions 294047 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r294047 | twilson | 2010-11-05 08:36:20 -0700 (Fri, 05 Nov 2010) + | 2 lines Tell people to use the correct common name for clients + as well ........ + +2010-11-05 15:26 +0000 [r294046] David Vossel + + * /, channels/chan_sip.c: Merged revisions 293924 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293924 | dvossel | 2010-11-04 16:39:51 -0500 (Thu, 04 Nov 2010) + | 4 lines Fixes ringback tone on sip semi-attended transfer. + ABE-2168 ........ + +2010-11-05 00:08 +0000 [r293971] Shaun Ruffell + + * /, codecs/codec_dahdi.c: Merged revisions 293970 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293970 | sruffell | 2010-11-04 19:07:11 -0500 + (Thu, 04 Nov 2010) | 32 lines Merged revisions 293969 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293969 | sruffell | 2010-11-04 19:06:02 -0500 + (Thu, 04 Nov 2010) | 25 lines Merged revisions 293968 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010) + | 17 lines codecs/codec_dahdi: Prevent "choppy" audio when + receiving unexpected frame sizes. dahdi-linux 2.4.0 (specifically + commit 9034) added the capability for the wctc4xxp to return more + than a single packet of data in response to a read. However, when + decoding packets, codec_dahdi was still assuming that the default + number of samples was in each read. In other words, each packet + your provider sent you, regardless of size, would result in 20 ms + of decoded data (30 ms if decoding G723). If your provider was + sending 60 ms packets then codec_dahdi would end up stripping 40 + ms of data from each transcoded frame resulting in "choppy" + audio. This would only affect systems where G729 packets are + arriving in sizes greater than 20ms or G723 packets arriving in + sizes greater than 30ms. DAHDI-744. ........ ................ + ................ + +2010-11-04 13:29 +0000 [r293888] Paul Belanger + + * /, channels/chan_sip.c: Merged revisions 293887 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293887 | pabelanger | 2010-11-04 09:27:54 -0400 (Thu, 04 Nov + 2010) | 8 lines Do not output port in IPaddress for AMI sippeers. + (closes issue #18248) Reported by: orn Patches: + ami_sippeers.patch uploaded by pabelanger (license 224) Tested + by: orn ........ + +2010-11-03 18:43 +0000 [r293809] Terry Wilson + + * main/rtp_engine.c, /, channels/chan_sip.c, + include/asterisk/rtp_engine.h: Merged revisions 293803 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293803 | twilson | 2010-11-03 11:05:14 -0700 (Wed, 03 Nov 2010) + | 25 lines Avoid valgrind warnings for + ast_rtp_instance_get_xxx_address The documentation for + ast_rtp_instance_get_(local/remote)_address stated that they + returned 0 for success and -1 on failure. Instead, they returned + 0 if the address structure passed in was already equivalent to + the address instance local/remote address or 1 otherwise. 90% of + the calls to these functions completely ignored the return + address and passed in an uninitialized struct, which would make + valgrind complain even though the operation was technically safe. + This patch fixes the documentation and converts the + get_xxx_address functions to void since all they really do is + copy the address and cannot fail. Additionally two new functions + (ast_rtp_instance_get_and_cmp_(local/remote)_address) are created + for the 3 times where the return value was actually checked. The + get_and_cmp_local_address function is currently unused, but + exists for the sake of symmetry. The only functional change as a + result of this change is that we will not do an + ast_sockaddr_cmp() on (mostly uninitialized) addresses before + doing the ast_sockaddr_copy() in the get_*_address functions. So, + even though it is an API change, it shouldn't have a noticeable + change in behavior. Review: + https://reviewboard.asterisk.org/r/995/ ........ + +2010-11-03 18:38 +0000 [r293808] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 293807 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293807 | rmudgett | 2010-11-03 13:35:19 -0500 + (Wed, 03 Nov 2010) | 34 lines Merged revisions 293806 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293806 | rmudgett | 2010-11-03 13:31:57 -0500 + (Wed, 03 Nov 2010) | 27 lines Merged revisions 293805 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) + | 20 lines Party A in an analog 3-way call would continue to hear + ringback after party C answers. All parties are analog FXS ports. + 1) A calls B. 2) A flash hooks to call C. 3) A flash hooks to + bring C into 3-way call before C answers. (A and B hear ringback) + 4) C answers 5) A continues to hear ringback during the 3-way + call. (All parties can hear each other.) * Fixed use of wrong + variable in dahdi_bridge() that stopped ringback on the wrong + subchannel. * Made several debug messages have more information. + A similar issue happens if B and C are SIP channels. B continues + to hear ringback. For some reason this only affects v1.8 and + trunk. * Don't start ringback on the real and 3-way subchannels + when creating the 3-way conference. Removing this code is benign + on v1.6.2 and earlier. ........ ................ ................ + +2010-11-02 23:10 +0000 [r293725] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 293724 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293724 | jpeeler | 2010-11-02 18:09:06 -0500 + (Tue, 02 Nov 2010) | 22 lines Merged revisions 293723 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293723 | jpeeler | 2010-11-02 18:07:13 -0500 + (Tue, 02 Nov 2010) | 15 lines Merged revisions 293722 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) + | 8 lines Add enabled/disabled information for rtautoclear sip + show settings output. When setting to zero/"no", the numeric + default was shown making it not obvious the disabled setting was + respected. (closes issue #18123) Reported by: zerohalo ........ + ................ ................ + +2010-11-02 21:31 +0000 [r293649] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 293648 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293648 | rmudgett | 2010-11-02 16:29:25 -0500 + (Tue, 02 Nov 2010) | 20 lines Merged revisions 293647 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293647 | rmudgett | 2010-11-02 16:26:30 -0500 + (Tue, 02 Nov 2010) | 13 lines Merged revisions 293639 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) + | 6 lines Make warning message have more useful information in + it. Change "Unable to get index, and nullok is not asserted" to + "Unable to get index for '' on channel + ((), line )". ........ ................ + ................ + +2010-11-02 20:47 +0000 [r293578-293612] Paul Belanger + + * main/manager.c, /: Merged revisions 293611 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293611 | pabelanger | 2010-11-02 16:45:09 -0400 (Tue, 02 Nov + 2010) | 2 lines If manager and tls are disabled, do not display + TCP/TLS Bindaddress. ........ + + * configs/gtalk.conf.sample, UPGRADE.txt, channels/chan_gtalk.c, + CHANGES: New CLI command 'gtalk show settings'. Review: + https://reviewboard.asterisk.org/r/984/ + +2010-11-02 14:43 +0000 [r293577] Mark Michelson + + * CHANGES: Add to the CHANGES file that the HTTP server supports + IPv6 addressing. + +2010-11-01 17:32 +0000 [r293531] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Merged revisions 293530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293530 | rmudgett | 2010-11-01 12:29:30 -0500 (Mon, 01 Nov 2010) + | 10 lines Analog 3-way call would not connect all parties if one + was using sig_pri. Also the "dahdi show channel" would not show + the correct 3-way call status. * Synchronized the inthreeway flag + between chan_dahdi and sig_analog. * Fixed a my_set_linear_mode() + sign error and made take an analog sub channel enum. ........ + +2010-11-01 16:11 +0000 [r293497] Paul Belanger + + * /, channels/chan_iax2.c: Merged revisions 293496 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r293496 | pabelanger | 2010-11-01 12:09:05 -0400 (Mon, + 01 Nov 2010) | 13 lines Use ast_sockaddr_from_sin function not + memcpy This resolves some IAX2 registration issue report on the + asterisk-users mailing list. (closes issue #18202) Reported by: + pabelanger Patches: update_registry.patch.v2 uploaded by + pabelanger (license 224) Tested by: pabelanger, Nic Colledge + (mailing list) Review: https://reviewboard.asterisk.org/r/993 + ........ + +2010-10-30 01:55 +0000 [r293342-293419] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 293418 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293418 | rmudgett | 2010-10-29 20:53:29 -0500 + (Fri, 29 Oct 2010) | 16 lines Merged revisions 293417 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293417 | rmudgett | 2010-10-29 20:49:15 -0500 + (Fri, 29 Oct 2010) | 9 lines Merged revisions 293416 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r293416 | rmudgett | 2010-10-29 20:45:49 -0500 (Fri, 29 + Oct 2010) | 1 line Remove some more code that serves no purpose. + ........ ................ ................ + + * channels/chan_dahdi.c, channels/sig_analog.c, /: Merged revisions + 293341 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293341 | rmudgett | 2010-10-29 19:46:41 -0500 + (Fri, 29 Oct 2010) | 16 lines Merged revisions 293340 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293340 | rmudgett | 2010-10-29 19:40:10 -0500 + (Fri, 29 Oct 2010) | 9 lines Merged revisions 293339 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r293339 | rmudgett | 2010-10-29 19:34:12 -0500 (Fri, 29 + Oct 2010) | 1 line Remove some code that serves no purpose. + ........ ................ ................ + +2010-10-29 21:50 +0000 [r293306] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 293305 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293305 | jpeeler | 2010-10-29 16:48:38 -0500 (Fri, 29 Oct 2010) + | 9 lines Modify sip_setoption to not complain about unknown + options. This now behaves just like the other setoption + callbacks. For the curious the offending option for the reporter + was AST_OPTION_CHANNEL_WRITE which was getting passed due to a + fix for chan_local in 286189. (closes issue #17985) Reported by: + globalnetinc ........ + +2010-10-29 20:46 +0000 [r293273] Mark Michelson + + * main/http.c, UPGRADE.txt, configs/http.conf.sample: Enable IPv6 + for the built-in HTTP server. Review: + https://reviewboard.asterisk.org/r/986 + +2010-10-28 20:01 +0000 [r293198] Tilghman Lesher + + * /, main/ast_expr2.h, res/ael/ael.tab.c, main/ast_expr2.y, + res/ael/ael_lex.c, res/ael/ael.tab.h, main/ast_expr2.c: Merged + revisions 293197 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293197 | tilghman | 2010-10-28 15:00:06 -0500 + (Thu, 28 Oct 2010) | 33 lines Merged revisions 293195-293196 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293195 | tilghman | 2010-10-28 14:52:52 -0500 + (Thu, 28 Oct 2010) | 12 lines Merged revisions 293194 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) + | 5 lines "!00" evaluated as false, which is incorrect. Fixing. + Reported (though the reporter did not understand he was reporting + a bug) on the asterisk-users list: + http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html + ........ ................ r293196 | tilghman | 2010-10-28 + 14:54:34 -0500 (Thu, 28 Oct 2010) | 12 lines Merged revisions + 293194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293194 | tilghman | 2010-10-28 14:44:37 -0500 (Thu, 28 Oct 2010) + | 5 lines "!00" evaluated as false, which is incorrect. Fixing. + Reported (though the reporter did not understand he was reporting + a bug) on the asterisk-users list: + http://lists.digium.com/pipermail/asterisk-users/2010-October/255505.html + ........ ................ ................ + +2010-10-28 16:11 +0000 [r293160] Jeff Peeler + + * /, funcs/func_strings.c: Merged revisions 293159 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293159 | jpeeler | 2010-10-28 11:11:08 -0500 + (Thu, 28 Oct 2010) | 18 lines Merged revisions 293158 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28 Oct 2010) + | 11 lines Fix infinite loop in FILTER(). Specifically when + you're using characters above \x7f or invalid character escapes + (e.g. \xgg). (closes issue #18060) Reported by: wdoekes Patches: + issue18060_func_strings_filter_infinite_loop.patch uploaded by + wdoekes (license 717) Tested by: wdoekes ........ + ................ + +2010-10-26 18:54 +0000 [r293120] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 293119 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r293119 | jpeeler | 2010-10-26 13:49:08 -0500 + (Tue, 26 Oct 2010) | 43 lines Merged revisions 293118 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r293118 | jpeeler | 2010-10-26 13:33:24 -0500 + (Tue, 26 Oct 2010) | 36 lines Merged revisions 293004 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r293004 | jpeeler | 2010-10-25 17:55:28 -0500 (Mon, 25 Oct 2010) + | 29 lines Fix inprocess_container in voicemail to correctly + restrict max messages. The comparison function logic was off, so + the number of sessions for a given mailbox were not being + incremented properly. This problem caused the maximum number of + messages per folder to not be respected when simultaneously + leaving multiple voicemails just below the threshold. These + problems should be fixed by the above, but just in case: Fixed + resequence_mailbox to rely on the actual number of detected + number of files in a directory rather than just assuming only 10 + messages more than the maximum had been left. Also if more + messages than the maximum are deleted they are actually removed + now. The second purpose of this commit should have been separated + out probably, but is related to the above. Again, if the number + of messages in a given voicemail folder exceeds the maximum set + limit make sure to allocate enough space for the deleted and + heard index tracking array. A few random fixes: There was a + forgotten decrement of the inprocess count in imap_store_file. + When using IMAP storage, do not look in the directory where file + based storage messages may still reside and influence the message + count. Ensure to use only the first format in sendmail. ABE-2516 + ........ ................ ................ + +2010-10-26 16:33 +0000 [r293047-293082] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 293081 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293081 | rmudgett | 2010-10-26 11:32:59 -0500 (Tue, 26 Oct 2010) + | 1 line No need to define the struct if there are no users. + ........ + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + channels/sig_pri.c: Merged revisions 293046 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r293046 | rmudgett | 2010-10-26 10:53:58 -0500 (Tue, 26 Oct 2010) + | 4 lines Allow the DAHDI driver to compile, even with a + sufficiently older version of libpri. Fixes our Bamboo builds. + ........ + +2010-10-25 21:16 +0000 [r292915-292970] Tilghman Lesher + + * /, channels/sig_pri.c: Merged revisions 292969 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292969 | tilghman | 2010-10-25 16:15:19 -0500 (Mon, 25 Oct 2010) + | 2 lines Several more defines that need to be altered for + compiling against an older version of libpri ........ + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + channels/sig_pri.c: Merged revisions 292906 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292906 | tilghman | 2010-10-25 14:28:35 -0500 (Mon, 25 Oct 2010) + | 4 lines Allow the DAHDI driver to compile, even with a + sufficiently older version of libpri. Fixes our Bamboo builds. + ........ + +2010-10-25 19:11 +0000 [r292869] David Vossel + + * channels/chan_local.c, /: Merged revisions 292868 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292868 | dvossel | 2010-10-25 14:07:50 -0500 + (Mon, 25 Oct 2010) | 39 lines Merged revisions 292867 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r292867 | dvossel | 2010-10-25 14:06:21 -0500 + (Mon, 25 Oct 2010) | 32 lines Merged revisions 292866 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r292866 | dvossel | 2010-10-25 14:05:07 -0500 (Mon, 25 Oct 2010) + | 27 lines This patch turns chan_local pvts into astobj2 objects. + chan_local does some dangerous things involving deadlock + avoidance. tech_pvt functions like hangup and queue_frame are + provided with a locked channel upon entry. Those functions are + completely safe as long as you don't attempt to give up that + channel lock, but that is impossible to guarantee due to the + required deadlock avoidance necessary to lock both the tech_pvt + and both channels involved. In the past, we have tried to account + for this by doing things like setting a "glare" flag that + indicates what function should destroy the pvt. This was used in + local_hangup and local_queue_frame to decided who should destroy + the pvt if they collided in separate threads. I have removed the + need to do this by converting all chan_local tech_pvts to + astobj2. This means we can ref a pvt before deadlock avoidance + and not have to worry about that pvt possibly getting destroyed + under us. It also cleans up where we destroy the tech_pvt. The + only unlink from the tech_pvt container occurs in local_hangup + now, which is where it should occur. Since there still may be + thread collisions on some functions like local_hangup after + deadlock avoidance, I have added some checks to detect those + collisions and exit appropriately. I think this patch is going to + solve quite a bit of weirdness we have had with local channels in + the past. ........ ................ ................ + +2010-10-22 22:40 +0000 [r292808-292826] Terry Wilson + + * contrib/scripts/ast_tls_cert, /: Merged revisions 292825 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292825 | twilson | 2010-10-22 15:35:29 -0700 (Fri, 22 Oct 2010) + | 4 lines Don't create directories without at least o+x Also, + making files that you are going to modify read-only is dumb. + ........ + + * contrib/scripts/ast_tls_cert, /: Merged revisions 292794 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292794 | twilson | 2010-10-22 15:18:36 -0700 (Fri, 22 Oct 2010) + | 2 lines Make files readable only by the owner ........ + +2010-10-22 21:29 +0000 [r292788] Leif Madsen + + * /, channels/chan_sip.c, configs/res_ldap.conf.sample, + contrib/scripts/asterisk.ldif: Merged revisions 292787 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292787 | lmadsen | 2010-10-22 16:28:43 -0500 + (Fri, 22 Oct 2010) | 21 lines Merged revisions 292786 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r292786 | lmadsen | 2010-10-22 16:16:12 -0500 (Fri, 22 Oct 2010) + | 13 lines Update the LDIF file for LDAP. The LDIF file + asterisk.ldif was quite a bit out of date from the + asterisk.ldap-schema file, so I've now updated that to be in + sync. The asterisk.ldif file being out of sync was a problem on + my systems where I was doing an ldapadd to import the schema into + the LDAP database, and the existing file would cause problems and + ERROR messages when registering. Additional documention has been + added based on feedback in the issue I'm closing. (closes issue + #13861) Reported by: scramatte Patches: ldap-update.txt uploaded + by lmadsen (license 10) Tested by: lmadsen, jcovert, suretec, + rgenthner ........ ................ + +2010-10-22 17:16 +0000 [r292743] Terry Wilson + + * contrib/scripts/ast_tls_cert (added), /: Merged revisions 292740 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292740 | twilson | 2010-10-22 09:49:34 -0700 (Fri, 22 Oct 2010) + | 45 lines Add TLS cert helper script This script is useful for + quickly generating self-signed CA, server, and client + certificates for use with Asterisk. It is still recommended to + obtain certificates from a recognized Certificate Authority and + to develop an understanding how SSL certificates work. Real + security is hard work. OPTIONS: -h Show this message -m Type of + cert "client" or "server". Defaults to server. -f Config filename + (openssl config file format) -c CA cert filename (creates new CA + cert/key as ca.crt/ca.key if not passed) -k CA key filename -C + Common name (cert field) For a server cert, this should be the + same address that clients attempt to connect to. Usually this + will be the Fully Qualified Domain Name, but might be the IP of + the server. For a CA or client cert, it is merely informational. + Make sure your certs have unique common names. -O Org name (cert + field) An informational string (company name) -o Output filename + base (defaults to asterisk) -d Output directory (defaults to the + current directory) Example: To create a CA and a server + (pbx.mycompany.com) cert with output in /tmp: ast_tls_cert -C + pbx.mycompany.com -O "My Company" -d /tmp This will create a CA + cert and key as well as asterisk.pem and the the two files that + it is made from: asterisk.crt and asterisk.key. Copy asterisk.pem + and ca.crt somewhere (like /etc/asterisk) and set + tlscertfile=/etc/asterisk.pem and tlscafile=/etc/ca.crt. Since + this is a self-signed key, many devices will require you to + import the ca.crt file as a trusted cert. To create a client cert + using the CA cert created by the example above: ast_tls_cert -m + client -c /tmp/ca.crt -k /tmp/ca.key -C "Joe User" -O \ "My + Company" -d /tmp -o joe_user This will create client.crt/key/pem + in /tmp. Use this if your device supports a client certificate. + Make sure that you have the ca.crt file set up as a tlscafile in + the necessary Asterisk configs. Make backups of all .key files in + case you need them later. ........ + +2010-10-22 17:10 +0000 [r292742] Mark Michelson + + * /, tests/test_event.c: Merged revisions 292741 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292741 | mmichelson | 2010-10-22 12:09:52 -0500 (Fri, 22 Oct + 2010) | 12 lines Prevent multiple runs of event_sub_test from + producing false failure results. The array of test subscriptions + was declared "static," meaning that the data.count field would + retain its value between runs of the test. After the first test + run, this would result in false reports of test failures. I chose + to just remove the "static" keyword from the structure since it's + not a huge deal to construct this structure during each run of + the test. Another alternative would have been to zero out the + data.count fields of each test subscription instead. ........ + +2010-10-22 15:47 +0000 [r292705] Richard Mudgett + + * main/channel.c, channels/chan_misdn.c, /, channels/sig_pri.c: + Merged revisions 292704 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292704 | rmudgett | 2010-10-22 10:47:08 -0500 (Fri, 22 Oct 2010) + | 19 lines Connected line is not updated when chan_dahdi/sig_pri + or chan_misdn transfers a call. When a call is transfered by ECT + or implicitly by disconnect in sig_pri or implicitly by + disconnect in chan_misdn, the connected line information is not + exchanged. The connected line interception macros also need to be + executed if defined. The CALLER interception macro is executed + for the held call. The CALLEE interception macro is executed for + the active/ringing call. JIRA ABE-2589 JIRA SWP-2296 Patches: + abe_2589_c3bier.patch uploaded by rmudgett (license 664) + abe_2589_v1.8_v2.patch uploaded by rmudgett (license 664) Review: + https://reviewboard.asterisk.org/r/958/ ........ + +2010-10-21 22:11 +0000 [r292668] Tilghman Lesher + + * /, channels/misdn/ie.c: Merged revisions 292667 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292667 | tilghman | 2010-10-21 17:09:25 -0500 (Thu, 21 Oct 2010) + | 2 lines Compile correctly on Linux (asterisk/localtime.h + depends upon asterisk/autoconfig.h loading first). ........ + +2010-10-21 18:23 +0000 [r292630] Paul Belanger + + * /, contrib/init.d/rc.suse.asterisk: Merged revisions 292628 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292628 | pabelanger | 2010-10-21 14:13:18 -0400 (Thu, 21 Oct + 2010) | 5 lines Fix typo in SUSE init script. Reported by: Dave + Cotton on asterisk-users list. ........ + +2010-10-21 16:46 +0000 [r292597] David Vossel + + * main/manager.c, /: Merged revisions 292595 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292595 | dvossel | 2010-10-21 11:14:33 -0500 (Thu, 21 Oct 2010) + | 14 lines Fixes recursive lock problem in manager.c It is + possible for a AMI session to freeze because of invalid use of + recursive locks during the EVENT processing. This patch removes + the unnecessary locks. (closes issue #18167) Reported by: sustav + Patches: manager_locking_v1.diff uploaded by dvossel (license + 671) Tested by: sustav ........ + +2010-10-21 13:17 +0000 [r292559] Leif Madsen + + * /, configs/res_ldap.conf.sample: Merged revisions 292557 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292557 | lmadsen | 2010-10-21 08:12:19 -0500 + (Thu, 21 Oct 2010) | 14 lines Merged revisions 292556 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r292556 | lmadsen | 2010-10-21 08:11:52 -0500 (Thu, 21 Oct 2010) + | 6 lines Change res_ldap.sample.conf to match the schema. + (closes issue #17376) Reported by: jcovert Patches: + res_ldap.conf.sample.patch uploaded by jcovert (license 551) + ........ ................ + +2010-10-21 11:38 +0000 [r292524] Russell Bryant + + * /, res/res_config_ldap.c: Merged revisions 292523 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r292523 | russell | 2010-10-21 06:36:47 -0500 (Thu, 21 + Oct 2010) | 4 lines Add var=value to log message on update + failure, and add newline. ... just for you, Leif. ........ + +2010-10-21 01:03 +0000 [r292490] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 292489 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292489 | rmudgett | 2010-10-20 20:02:50 -0500 (Wed, 20 Oct 2010) + | 7 lines Send CONNECT_ACKNOWLEDGE for CIS calls too. The + originator of the Q.SIG call completion signaling link was not + changed to the active state when the CONNECT message came in. The + T309 processing would immediately kill the signaling link because + it was not in the active state. ........ + +2010-10-21 00:23 +0000 [r292414-292443] Paul Belanger + + * /, apps/app_voicemail.c: Merged revisions 292436 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r292436 | pabelanger | 2010-10-20 20:21:59 -0400 (Wed, + 20 Oct 2010) | 8 lines Application not properly unregister in + voicemail (closes issue #18128) Reported by: junky Patches: + vm_unregister.diff uploaded by junky (license 177) Tested by: + pabelanger, lmadsen ........ + + * apps/app_dial.c, /: Merged revisions 292413 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292413 | pabelanger | 2010-10-20 20:07:17 -0400 + (Wed, 20 Oct 2010) | 24 lines Merged revisions 292412 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r292412 | pabelanger | 2010-10-20 20:05:45 -0400 + (Wed, 20 Oct 2010) | 17 lines Merged revisions 292411 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct + 2010) | 10 lines Record priv-recordintro as sln, not gsm This + removes the gsm->sln step when transcoding priv-recordintro. + (closes issue #18176) Reported by: pabelanger Patches: + chan_sip.diff uploaded by pabelanger (license 224) ........ + ................ ................ + +2010-10-20 00:41 +0000 [r292377] Tilghman Lesher + + * /, res/res_musiconhold.c: Merged revisions 292376 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r292376 | tilghman | 2010-10-19 19:40:29 -0500 (Tue, 19 + Oct 2010) | 5 lines Oops. This module uses the generic timer and + no longer uses DAHDI. This causes a problem with the Solaris and + other system builds that have gcc 4.1 (where optional_api is + non-optional). ........ + +2010-10-19 22:19 +0000 [r292345] Paul Belanger + + * /, contrib/scripts/install_prereq: Merged revisions 292343 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292343 | pabelanger | 2010-10-19 18:14:23 -0400 (Tue, 19 Oct + 2010) | 2 lines Add resample and imap_tk dependencies. ........ + +2010-10-19 19:35 +0000 [r292310] Terry Wilson + + * /, channels/chan_sip.c, res/res_srtp.c: Merged revisions 292309 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292309 | twilson | 2010-10-19 12:27:32 -0700 (Tue, 19 Oct 2010) + | 10 lines Add sip show peer info about crypto and remove dated + comment This patch adds information about the encryption setting + to 'sip show peers' and removes an out-of-date comment from + res_srtp.c and instead directs users to the proper documentation. + (closes issue #18140) Reported by: chodorenko ........ + +2010-10-18 22:14 +0000 [r292231] Leif Madsen + + * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 292225 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292225 | lmadsen | 2010-10-18 16:51:23 -0500 + (Mon, 18 Oct 2010) | 24 lines Merged revisions 292224 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r292224 | lmadsen | 2010-10-18 16:50:47 -0500 + (Mon, 18 Oct 2010) | 17 lines Merged revisions 292222 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r292222 | lmadsen | 2010-10-18 16:47:25 -0500 (Mon, 18 Oct 2010) + | 9 lines Add support for the new English (Australian Accent) + sound files. (closes issue #17426) Reported by: camsown Patches: + core-sounds-en_AU.txt uploaded by camsown (license 1050) + add_AU_sounds.patch.txt uploaded by lmadsen (license 10) Tested + by: camsown, lmadsen, jtodd, qwell ........ ................ + ................ + +2010-10-18 21:56 +0000 [r292228] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 292227 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292227 | jpeeler | 2010-10-18 16:55:46 -0500 + (Mon, 18 Oct 2010) | 25 lines Merged revisions 292226 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r292226 | jpeeler | 2010-10-18 16:54:38 -0500 + (Mon, 18 Oct 2010) | 18 lines Merged revisions 292223 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) + | 11 lines Fix improper operator key acceptance and clean up temp + recording files. This is a fix for when pressing the operator key + after recording an unavailable, busy, name, or temporary message + in mailbox options. The operator key should not be accepted here, + but should be allowed during the message recording. If the + operator key is pressed during ensure the file is saved or + deleted as apporopriate. Also, ensure removal of temporary + recorded files after an early hang up or when message acceptance + confirmation times out. ABE-2518 ........ ................ + ................ + +2010-10-18 19:52 +0000 [r292189] Russell Bryant + + * main/netsock2.c, /: Merged revisions 292188 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292188 | russell | 2010-10-18 14:50:04 -0500 (Mon, 18 Oct 2010) + | 9 lines Resolve some compiler errors in ast_sockaddr_is_any(). + These errors came up once this function was used from within + netsock2.c. The errors were like the following: netsock2.c:393: + error: dereferencing pointer ‘({anonymous})’ does break + strict-aliasing rules The usage of a union here avoids this + problem. ........ + +2010-10-18 19:16 +0000 [r292156] David Vossel + + * main/netsock2.c, /: Merged revisions 292155 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292155 | dvossel | 2010-10-18 14:16:00 -0500 (Mon, 18 Oct 2010) + | 2 lines Fixes build error for systems not supporting + IPV6_TCLASS. ........ + +2010-10-18 17:18 +0000 [r292124] Matthew Nicholson + + * /, addons/chan_mobile.c: Merged revisions 292122 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r292122 | mnicholson | 2010-10-18 12:15:24 -0500 (Mon, + 18 Oct 2010) | 5 lines Fix the cmgr parser. (closes issue + 0018152) Reported by: menschentier ........ + +2010-10-18 16:03 +0000 [r292086] David Vossel + + * main/netsock2.c, /: Merged revisions 292085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292085 | dvossel | 2010-10-18 11:02:17 -0500 (Mon, 18 Oct 2010) + | 7 lines Fixes qos settings for sockets bound to any IPv6 or + IPv4 address. (closes issue #18099) Reported by: jamesnet + Patches: issues_18099_v3.diff uploaded by dvossel (license 671 + ........ + +2010-10-18 15:33 +0000 [r292084] Jeff Peeler + + * pbx/pbx_spool.c, /: Merged revisions 292083 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292083 | jpeeler | 2010-10-18 10:32:40 -0500 (Mon, 18 Oct 2010) + | 4 lines Disable use of inotify for call file handling as it is + not working properly. (related to #18089) ........ + +2010-10-16 11:51 +0000 [r292052] Tzafrir Cohen + + * /, configs/musiconhold.conf.sample, res/res_musiconhold.c: Merged + revisions 292050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r292050 | tzafrir | 2010-10-16 12:47:00 +0200 + (ש', 16 אוק 2010) | 22 lines Merged revisions 292049 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r292049 | tzafrir | 2010-10-16 12:03:04 +0200 (ש', 16 + אוק 2010) | 15 lines Base directory for MOH should be ASTDATADIR + If the directive 'directory' is relative, make it relative to the + datadir, rather than to the varlibdir. In the sample + configuration it is relative ('moh'). This has no effect unless + you have actively set the datadir explicitly (at build time or at + run time). (closes issue #16906) Patches: moh_datadir uploaded by + tzafrir (license 46) Review: + https://reviewboard.asterisk.org/r/974/ ........ ................ + +2010-10-15 21:49 +0000 [r292017] Terry Wilson + + * /, res/res_srtp.c: Merged revisions 292016 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r292016 | twilson | 2010-10-15 16:40:56 -0500 (Fri, 15 Oct 2010) + | 5 lines Ref/unref res_srtp when we create/destroy a session + This avoids unhappy crashing when we try to 'core stop + gracefully' and res_srtp tries to unload before chan_sip does. + Thanks, Russell! ........ + +2010-10-15 20:12 +0000 [r291943] David Vossel + + * /, channels/chan_sip.c: Merged revisions 291942 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291942 | dvossel | 2010-10-15 15:12:04 -0500 (Fri, 15 Oct 2010) + | 8 lines Fixes peer's host port information being lost on sip + reload. (closes issue #18135) Reported by: lmadsen Patches: + crazy_ports_v2.diff uploaded by dvossel (license 671) Tested by: + lmadsen ........ + +2010-10-15 19:53 +0000 [r291941] Paul Belanger + + * configs/gtalk.conf.sample, /: Merged revisions 291940 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291940 | pabelanger | 2010-10-15 15:50:22 -0400 + (Fri, 15 Oct 2010) | 16 lines Merged revisions 291939 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291939 | pabelanger | 2010-10-15 15:35:20 -0400 + (Fri, 15 Oct 2010) | 9 lines Merged revisions 291938 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r291938 | pabelanger | 2010-10-15 15:30:41 -0400 (Fri, + 15 Oct 2010) | 2 lines Clean up formatting. ........ + ................ ................ + +2010-10-15 16:54 +0000 [r291906] Terry Wilson + + * /, res/res_jabber.c: Merged revisions 291905 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291905 | twilson | 2010-10-15 09:39:58 -0700 + (Fri, 15 Oct 2010) | 14 lines Merged revisions 291904 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r291904 | twilson | 2010-10-15 09:16:57 -0700 (Fri, 15 Oct 2010) + | 7 lines Don't crash or deadlock on module unload We can't hold + the lock while pthread_join is called since aji_log_hook will + attempt to lock from the other therad. We reorder the + pthread_join and ast_aji_disconnect so that we don't do an + SSL_read() while SSL_shutdown is running, causing a crash. + ........ ................ + +2010-10-14 22:10 +0000 [r291828-291830] David Vossel + + * main/netsock2.c, /: Merged revisions 291829 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291829 | dvossel | 2010-10-14 17:09:32 -0500 (Thu, 14 Oct 2010) + | 8 lines Set TCLASS field of IPv6 header when sip qos options + are set. (closes issue #18099) Reported by: jamesnet Patches: + issues_18099_v2.diff uploaded by dvossel (license 671) Tested by: + dvossel, jamesnet ........ + + * /, channels/chan_gtalk.c: Merged revisions 291827 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r291827 | dvossel | 2010-10-14 16:27:42 -0500 (Thu, 14 + Oct 2010) | 18 lines Safer xml parsing, treat all clients the + same, and better local candidate selection. The gtalk channel + driver was doing several unsafe operations in regards to how it + parsed incoming XML messages. I have cleaned that code up so it + should be much safer now. We now treat all clients types the + same. We have no reason to distinguish between GMAIL and GOOGLE + VOICE clients anymore because they all work the same way. I also + modified how the local ip is found. If no bindaddress is provided + in the config file, we attempt to determine the local ip we would + use to connect to google.com. If that fails, then we fall back to + the ast_find_ourip() function as a last resort. Using the new + method makes it much less likely that we would ever advertise a + local RTP candidate as a loopback address. ........ + +2010-10-14 18:46 +0000 [r291792] Jeff Peeler + + * /, main/stdtime/localtime.c: Merged revisions 291791 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r291791 | jpeeler | 2010-10-14 13:45:02 -0500 (Thu, 14 + Oct 2010) | 10 lines Add missing ifdefs for test framework and + new locale code. (closes issue #18137) Reported by: ovi Patches: + 18137_test_framework_ifdef.patch uploaded by wdoekes (license + 717) 18137_localelist_warning.patch uploaded by wdoekes (license + 717) Tested by: ovi ........ + +2010-10-14 15:21 +0000 [r291760] Paul Belanger + + * channels/chan_jingle.c, include/asterisk/acl.h, /, + channels/chan_sip.c, channels/chan_h323.c, main/acl.c, + channels/chan_gtalk.c: Merged revisions 291758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291758 | pabelanger | 2010-10-14 11:15:12 -0400 (Thu, 14 Oct + 2010) | 11 lines Add the ability for ast_find_ourip to return + IPv4, IPv6 or both. While testing chan_gtalk I noticed jabber was + using my IPv6 address and not IPv4. When using bindaddr=0.0.0.0 + it is possible for ast_find_ourip() to return both IPv6 and IPv4 + results. Adding a family parameter gives you the ablility to + choose. Since jabber/gtalk/h323 do not support IPv6, we should + only return IPv4 results. Review: + https://reviewboard.asterisk.org/r/973/ ........ + +2010-10-14 12:10 +0000 [r291726] Russell Bryant + + * /, doc/tex/secure-calls.tex: Merged revisions 291725 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r291725 | russell | 2010-10-14 07:08:43 -0500 (Thu, 14 + Oct 2010) | 2 lines Fix a typo - s/seucre/secure/ ........ + +2010-10-13 23:52 +0000 [r291658] Richard Mudgett + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/sig_analog.h: Merged revisions 291656 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291656 | rmudgett | 2010-10-13 18:45:11 -0500 + (Wed, 13 Oct 2010) | 34 lines Merged revisions 291655 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291655 | rmudgett | 2010-10-13 18:36:50 -0500 + (Wed, 13 Oct 2010) | 27 lines Merged revisions 291643 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r291643 | rmudgett | 2010-10-13 18:29:58 -0500 (Wed, 13 Oct 2010) + | 20 lines Deadlock between dahdi_exception() and + dahdi_indicate(). There is a deadlock between dahdi_exception() + and dahdi_indicate() for analog ports. The call-waiting and + three-way-calling feature can experience deadlock if these + features are trying to do something and an event from the bridged + channel happens at the same time. Deadlock avoidance code added + to obtain necessary channel locks before attemting an operation + with call-waiting and three-way-calling. (closes issue #16847) + Reported by: shin-shoryuken Patches: issue_16847_v1.4.patch + uploaded by rmudgett (license 664) issue_16847_v1.6.2.patch + uploaded by rmudgett (license 664) issue_16847_v1.8_v2.patch + uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett + Review: https://reviewboard.asterisk.org/r/971/ ........ + ................ ................ + +2010-10-13 23:47 +0000 [r291657] Terry Wilson + + * main/channel.c, /: Merged revisions 291581 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291581 | twilson | 2010-10-13 16:01:56 -0700 + (Wed, 13 Oct 2010) | 35 lines Merged revisions 291580 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291580 | twilson | 2010-10-13 15:58:43 -0700 + (Wed, 13 Oct 2010) | 28 lines Merged revisions 291577 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r291577 | twilson | 2010-10-13 15:45:15 -0700 (Wed, 13 Oct 2010) + | 21 lines Don't ignore frames that have been queued when + softhangup'd When an outgoing call is answered and hung up by the + far end *very* quickly, we may not read any frames and therefor + end up with a call that displays the wrong + disposition/DIALSTATUS. The reason is because ast_queue_hangup() + immediately sets the _softhangup flag on the channel and then + queues the HANGUP control frame, but __ast_read refuses to read + any frames if ast_check_hangup() indicates that a hangup request + has been made (which it will if _softhangup is set). So, we end + up losing control frames. This change makes __ast_read continue + to read frames even if a soft hangup has been requested. It + queues a hangup frame to make sure that __ast_read() will still + eventually return NULL. Much thanks to David Vossel for all of + the reviews, discussion, and help! (closes issue #16946) Reported + by: davidw Review: https://reviewboard.asterisk.org/r/740/ + ........ ................ ................ + +2010-10-13 22:47 +0000 [r291579] David Vossel + + * /, channels/chan_gtalk.c: Merged revisions 291578 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r291578 | dvossel | 2010-10-13 17:46:34 -0500 (Wed, 13 + Oct 2010) | 4 lines More fixup for chan_gtalk. This patch makes + the xml parsing safer. ........ + +2010-10-13 22:34 +0000 [r291576] Terry Wilson + + * Makefile, /, static-http/mantest.html (added): Merged revisions + 291575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291575 | twilson | 2010-10-13 15:24:44 -0700 (Wed, 13 Oct 2010) + | 8 lines Add a simple AMI client web page This patch uses the + XML docs to parse all of the available AMI commands and allows + you to enter the command name and be presented with a form with + the available fields. You can then rapidly tab through the fields + and submit the command and view the response. It is much + faster/easier than having to use telnet for testing purposes. + ........ + +2010-10-13 20:24 +0000 [r291470-291542] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 291541 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r291541 | rmudgett | 2010-10-13 15:21:02 -0500 (Wed, 13 + Oct 2010) | 26 lines The chan_dahdi faxdetect option only works + for the first FAX call. The chan_dahdi faxdetect option only + works for the first call. After that the option no longer works. + The struct dahdi_pvt.callprogress member is the encoded user + config setting for the callprogress and faxdetect config options. + Changing this value alters the configuration for all following + calls until the chan_dahdi.conf file is reloaded. * Fixed the + chan_dahdi ast_channel_setoption callback to not change the users + faxdetect config setting except for the current call. * Fixed the + chan_dahdi ast_channel_queryoption callback to read the active + DSP setting of the faxdetect option. * Made actually disable the + active faxdetect DSP setting for the current call on the analog + port. my_handle_dtmfup() is used for normal analog ports. + dahdi_handle_dtmfup() is the legacy code and is no longer used + unless in a radio mode. (closes issue #18116) Reported by: + seandarcy Patches: issue18116_v1.8.patch uploaded by rmudgett + (license 664) Review: https://reviewboard.asterisk.org/r/972/ + ........ + + * channels/chan_misdn.c, /: Merged revisions 291507 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291507 | rmudgett | 2010-10-13 14:01:48 -0500 + (Wed, 13 Oct 2010) | 18 lines Merged revision 291504 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r291504 | rmudgett | 2010-10-13 13:30:21 -0500 (Wed, + 13 Oct 2010) | 11 lines Hold off ast_hangup() from destroying the + ast_channel. Must get the ast_channel lock before proceeding with + release_chan() and release_chan_early() to hold off ast_hangup() + from destroying the ast_channel. Missed this change for -r291468. + JIRA ABE-2598 JIRA SWP-2317 .......... ................ + + * channels/chan_misdn.c, /: Merged revisions 291469 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291469 | rmudgett | 2010-10-13 13:10:21 -0500 + (Wed, 13 Oct 2010) | 23 lines Merge revision 291468 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r291468 | rmudgett | 2010-10-13 12:39:02 -0500 (Wed, + 13 Oct 2010) | 16 lines Memory overwrites when releasing mISDN + call. Phone <--> Asterisk <-- ALERTING --> DISCONNECT <-- RELEASE + --> RELEASE_COMPLETE * Add lock protection around channel list + for find/add/delete operations. * Protect misdn_hangup() from + release_chan() and vise versa using the release_lock. JIRA + ABE-2598 JIRA SWP-2317 .......... ................ + +2010-10-13 15:51 +0000 [r291395] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 291394 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291394 | russell | 2010-10-13 10:46:39 -0500 + (Wed, 13 Oct 2010) | 20 lines Merged revisions 291393 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291393 | russell | 2010-10-13 10:29:21 -0500 + (Wed, 13 Oct 2010) | 13 lines Merged revisions 291392 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r291392 | russell | 2010-10-13 10:23:19 -0500 (Wed, 13 Oct 2010) + | 6 lines Lock pvt so pvt->owner can't disappear when queueing up + a frame. This fixes a crash due to a hangup race condition. + ABE-2601 ........ ................ ................ + +2010-10-13 08:58 +0000 [r291361] Stefan Schmidt + + * apps/app_macro.c: Report what extension called a failed macro Add + the extension and context of the calling channel to the log + output if a macro could not be found. (closes issue #18112) + Reported by: prado Patches: app_macro-info.diff uploaded by prado + (license 510) Tested by: schmidts + +2010-10-12 17:21 +0000 [r291287] Leif Madsen + + * /, configs/phoneprov.conf.sample: Merged revisions 291284 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291284 | lmadsen | 2010-10-12 12:20:43 -0500 + (Tue, 12 Oct 2010) | 15 lines Merged revisions 291280 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r291280 | lmadsen | 2010-10-12 12:20:02 -0500 (Tue, 12 Oct 2010) + | 7 lines Add undocumented variables to phoneprov.conf.sample + (closes issue #18107) Reported by: lathama Patches: + phoneprov.conf.sample.diff uploaded by lathama (license 1028) + ........ ................ + +2010-10-12 17:07 +0000 [r291266] Tilghman Lesher + + * /, main/acl.c: Merged revisions 291265 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291265 | tilghman | 2010-10-12 12:06:23 -0500 + (Tue, 12 Oct 2010) | 16 lines Merged revisions 291264 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291264 | tilghman | 2010-10-12 12:05:31 -0500 + (Tue, 12 Oct 2010) | 9 lines Merged revisions 291263 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r291263 | tilghman | 2010-10-12 11:55:30 -0500 (Tue, 12 + Oct 2010) | 2 lines Oops, incorrect range (although unallocated + at ARIN) ........ ................ ................ + +2010-10-12 16:08 +0000 [r291231] Leif Madsen + + * /, configs/manager.conf.sample: Merged revisions 291230 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291230 | lmadsen | 2010-10-12 11:08:04 -0500 + (Tue, 12 Oct 2010) | 10 lines Merged revisions 291229 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r291229 | lmadsen | 2010-10-12 11:07:28 -0500 (Tue, 12 Oct 2010) + | 2 lines Add documention that mentions options are defined but + not used. (Issue #18101) ........ ................ + +2010-10-12 16:00 +0000 [r291193-291228] David Vossel + + * main/manager.c, /: Merged revisions 291227 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291227 | dvossel | 2010-10-12 10:58:56 -0500 (Tue, 12 Oct 2010) + | 16 lines Fixes manager.c crash. This issue was caused by + improper use of the mansession lock and manession_session lock. + These two structures are confusing to begin with so I'm not + surprised this occurred. I fixed this by consistently making sure + we use each of these locks only to protect the data in the + corresponding structure. We had mismatched usage of these locks + which resulted in no mutual exclusivity occurring at all. (closes + issue #17994) Reported by: vrban Patches: + mansession_locking_fix.diff uploaded by dvossel (license 671) + Tested by: vrban ........ + + * /, CHANGES: Merged revisions 291194 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291194 | dvossel | 2010-10-11 16:44:04 -0500 (Mon, 11 Oct 2010) + | 2 lines Update CHANGES to reflect new gtalk.conf options. + ........ + + * configs/gtalk.conf.sample, /, res/res_stun_monitor.c, + channels/chan_gtalk.c, include/asterisk/stun.h: Merged revisions + 291192 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291192 | dvossel | 2010-10-11 16:38:39 -0500 (Mon, 11 Oct 2010) + | 19 lines Gtalk enhancements and general code cleanup. This + patch includes several chan_gtalk enhancements. Two new + gtalk.conf options have been added, externip and stunadd. Setting + externip allows us to manually specify what the external IP + address is outside of a NAT environment. Setting the stunaddr + option to a valid stun server allows for that external ip to be + retrieved via a STUN server automatically. This external IP is + then advertised during call setup as a possible candidate. I have + also attempted to clean up chan_gtalk's code so it meets our + coding guidelines. During this cleanup I noticed several things + that need to be done in the code and made a TODO section at the + top of the file. ........ + +2010-10-11 19:07 +0000 [r291076-291115] Richard Mudgett + + * channels/chan_sip.c: Add todo comment about handle_incoming() + calling assumption. + + * /, channels/chan_sip.c: Merged revisions 291112-291113 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291112 | rmudgett | 2010-10-11 13:48:15 -0500 + (Mon, 11 Oct 2010) | 20 lines Merged revisions 291110-291111 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r291110 | rmudgett | 2010-10-11 13:34:22 -0500 + (Mon, 11 Oct 2010) | 9 lines Merged revisions 291109 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r291109 | rmudgett | 2010-10-11 13:29:43 -0500 (Mon, 11 + Oct 2010) | 1 line Add missing unlock to an exception condition + in reload_config(). ........ ................ r291111 | rmudgett + | 2010-10-11 13:39:06 -0500 (Mon, 11 Oct 2010) | 1 line Make exit + from handle_request_do() consistent. ................ + ................ r291113 | rmudgett | 2010-10-11 13:51:13 -0500 + (Mon, 11 Oct 2010) | 1 line Move declaration closer to where now + used. ................ + + * /, main/cli.c: Merged revisions 291075 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r291075 | rmudgett | 2010-10-11 11:42:54 -0500 + (Mon, 11 Oct 2010) | 22 lines Merged revisions 291073 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r291073 | rmudgett | 2010-10-11 11:39:17 -0500 (Mon, 11 Oct 2010) + | 15 lines Fixed infinite loop in verbose/debug message output. + Setting the module/filename specific message level and then + changing it resulted in the linked list being looped on itself. + Traversing this linked list is an infinite loop if what you are + looking for is not in the list. Also plugged some CLI parsing + holes in the associated CLI command: * Removing a nonexistent + module from the list actually added it with a level of zero. * + Setting the non-module specific level to zero is now equivalent + to setting it to "off" as documented. ........ ................ + +2010-10-11 03:20 +0000 [r291039] Tilghman Lesher + + * /, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Merged + revisions 291038 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r291038 | tilghman | 2010-10-09 18:25:37 -0500 (Sat, 09 Oct 2010) + | 2 lines Add missing option to set calls to be logged in + GMT/UTC. ........ + +2010-10-09 14:04 +0000 [r291006] Alexandr Anikin + + * addons/ooh323c/src/ooh323.c, addons/chan_ooh323.c, + addons/ooh323c/src/ooh245.c: Added fast start and h.245 tunneling + options per user and peer. Added options for faststart/h.245 + tunneling per user/peer, properly handle these and global + options, correction of handling fs/tunneling fields in signalling + responses (closes issue #17972) Reported by: salecha Patches: + fs-tunnel-per-point-3.patch uploaded by may213 (license 454) + Tested by: may213, salecha + +2010-10-08 20:45 +0000 [r290974] David Vossel + + * /, channels/chan_gtalk.c: Merged revisions 290973 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290973 | dvossel | 2010-10-08 15:44:59 -0500 (Fri, 08 + Oct 2010) | 12 lines Make outbound Google Voice calls. This patch + allows for outbound Google Voice calls to be dialed from Asterisk + using chan_gtalk. Below is an example dialstring. exten -> + blah,1,Dial(Gtalk/asterisk/+15552225555@voice.google.com,,) In + this example, 'asterisk' is the jabber.conf profile configured to + connect to your gmail account. In order to receive Google Voice + calls make sure to enable 'allowguest=yes' in gtalk.conf. + ........ + +2010-10-08 16:27 +0000 [r290939] Erin Spiceland + + * addons/app_mysql.c, configs/res_config_mysql.conf.sample, /, + addons/res_config_mysql.c: Add option to res_config_mysql and + app_mysql to specify a character set that MySQL should use. + (closes issue 17948) Reported by qmax. + +2010-10-08 03:00 +0000 [r290865] Jeff Peeler + + * /, main/asterisk.c: Merged revisions 290864 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290864 | jpeeler | 2010-10-07 21:56:24 -0500 + (Thu, 07 Oct 2010) | 23 lines Merged revisions 290863 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r290863 | jpeeler | 2010-10-07 21:45:44 -0500 + (Thu, 07 Oct 2010) | 16 lines Merged revisions 290862 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r290862 | jpeeler | 2010-10-07 21:35:29 -0500 (Thu, 07 Oct 2010) + | 9 lines Ensure editline cleanup occurs when Ctrl-C is pressed + at control console. A recent change was made to avoid a race + condition on shutdown which only called the end functions from + the console thread. However, when pressing Ctrl-C the quit + handler is called from the signal handler thread. (closes issue + #17698) Reported by: jmls ........ ................ + ................ + +2010-10-07 22:39 +0000 [r290830-290831] David Vossel + + * /, channels/chan_gtalk.c: Merged revisions 290829 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290829 | dvossel | 2010-10-07 17:38:05 -0500 (Thu, 07 + Oct 2010) | 6 lines Add Philippe Sultan to chan_gtalk author + list. Philippe has made some notable contributions to the gtalk + channel driver. His name deserves to be listed amoung the authors + of that file. Thanks Philippe! ........ + + * /, channels/chan_gtalk.c: Merged revisions 290828 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290828 | dvossel | 2010-10-07 16:44:58 -0500 (Thu, 07 + Oct 2010) | 5 lines Outbound gtalk calls now work correctly. + There was a problem with how the candidates were being built on + an outbound call. This patch fixes that. ........ + +2010-10-07 20:59 +0000 [r290753] Jason Parker + + * /, configure, include/asterisk/autoconfig.h.in, + autoconf/ast_ext_lib.m4: Merged revisions 290752 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290752 | qwell | 2010-10-07 15:58:47 -0500 + (Thu, 07 Oct 2010) | 23 lines Merged revisions 290751 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r290751 | qwell | 2010-10-07 15:57:14 -0500 + (Thu, 07 Oct 2010) | 16 lines Merged revisions 290750 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r290750 | qwell | 2010-10-07 15:56:04 -0500 (Thu, 07 Oct 2010) | + 9 lines Allow PRI to build properly when using --with-pri. Use + the directories found for the parent when using lib dependencies. + (closes issue #17314) Reported by: tzafrir Patches: + 17314-withdeps.diff uploaded by qwell (license 4) ........ + ................ ................ + +2010-10-07 11:12 +0000 [r290714] Russell Bryant + + * main/pbx.c, /: Merged revisions 290713 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290713 | russell | 2010-10-07 13:00:52 +0200 + (Thu, 07 Oct 2010) | 11 lines Merged revisions 290712 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r290712 | russell | 2010-10-07 12:53:56 +0200 (Thu, 07 Oct 2010) + | 4 lines Don't crash when Set() is called without a value. + Review: https://reviewboard.asterisk.org/r/949/ ........ + ................ + +2010-10-06 21:23 +0000 [r290649-290677] David Vossel + + * /, channels/chan_gtalk.c: Merged revisions 290674 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290674 | dvossel | 2010-10-06 16:22:51 -0500 (Wed, 06 + Oct 2010) | 4 lines Fixes commented out code to use #if 0 + instead. Thanks to rmudgett for catching this! ........ + + * /, channels/chan_gtalk.c: Merged revisions 290648 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290648 | dvossel | 2010-10-06 16:08:19 -0500 (Wed, 06 + Oct 2010) | 12 lines Fixes gtalk outbound DTMF to work properly. + Outbound DTMF with gtalk needs to be done within the RTP stream. + I discovered this after investigating a packet capture from the + gmail client. Instead of performing jingle signaling DTMF, the + gtalk servers expect all DTMF to arrive on the RTP stream using + RFC2833 way of doing things. Chan_gtalk also had an issue with + negotiating RTP payload type 106 for the telephony-event and then + sending DTMF as payload 101. This has been resolved by always + negotiating 101 as the payload type like we do everywhere else. + With this patch, incoming google voice calls forwarded to + Asterisk via gtalk work. ........ + +2010-10-06 18:56 +0000 [r290615] Richard Mudgett + + * apps/app_dial.c, /: Merged revisions 290614 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290614 | rmudgett | 2010-10-06 13:50:37 -0500 + (Wed, 06 Oct 2010) | 12 lines Merged revision 290613 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed, + 06 Oct 2010) | 5 lines Eliminate a redundant test for + AST_CONTROL_REDIRECTING. Eliminate redundant test for + AST_CONTROL_REDIRECTING that prevents running the redirecting + interception macro if it is defined. .......... ................ + +2010-10-06 13:50 +0000 [r290577] Tilghman Lesher + + * /, main/file.c: Merged revisions 290576 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290576 | tilghman | 2010-10-06 08:49:19 -0500 + (Wed, 06 Oct 2010) | 15 lines Merged revisions 290575 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r290575 | tilghman | 2010-10-06 08:48:27 -0500 (Wed, 06 Oct 2010) + | 8 lines Allow streaming audio from a pipe. (closes issue + #18001) Reported by: jamicque Patches: + 20100926__issue18001.diff.txt uploaded by tilghman (license 14) + Tested by: jamicque ........ ................ + +2010-10-06 04:47 +0000 [r290543] Terry Wilson + + * res/res_rtp_asterisk.c, /: Merged revisions 290542 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290542 | twilson | 2010-10-05 21:35:51 -0700 (Tue, 05 + Oct 2010) | 6 lines Don't try to send RTP when remote_address is + null It is possible for ast_rtp_stop() to be called which will + clear the remote address and cause the sendto to fail and spam + warnings. Don't send in this case. ........ + +2010-10-05 22:23 +0000 [r290480-290509] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 290506 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290506 | dvossel | 2010-10-05 17:23:00 -0500 (Tue, 05 + Oct 2010) | 2 lines Fixes uninitialized memory problem in 'iax2 + set debug peer' option. ........ + + * /, include/asterisk/jabber.h, include/asterisk/jingle.h, + channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 290479 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r290479 | dvossel | 2010-10-05 17:00:43 -0500 (Tue, 05 Oct 2010) + | 6 lines Fixes chan_gtalk to work with gmail client This patch + was written by Philippe Sultan (phsultan). Thanks for keeping + this up to date! ........ + +2010-10-05 20:24 +0000 [r290414] Tilghman Lesher + + * /, res/res_jabber.c: Merged revisions 290408 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290408 | tilghman | 2010-10-05 15:23:33 -0500 + (Tue, 05 Oct 2010) | 22 lines Merged revisions 290396 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r290396 | tilghman | 2010-10-05 15:21:02 -0500 + (Tue, 05 Oct 2010) | 15 lines Merged revisions 290392 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r290392 | tilghman | 2010-10-05 15:20:07 -0500 (Tue, 05 Oct 2010) + | 8 lines Fix a crash by ensuring that we don't alter memory + after it's freed. (closes issue #17387) Reported by: jmls + Patches: 20100726__issue17387.diff.txt uploaded by tilghman + (license 14) Tested by: jmls ........ ................ + ................ + +2010-10-05 20:10 +0000 [r290377-290379] David Vossel + + * /, channels/chan_iax2.c: Merged revisions 290378 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290378 | dvossel | 2010-10-05 15:09:06 -0500 (Tue, 05 + Oct 2010) | 11 lines Resolves dnsmgr memory corruption in + chan_iax2. (closes issue #17902) Reported by: afried Patches: + issue_17902.rev1.txt uploaded by russell (license 2) Tested by: + afried, russell, dvossel Review: + https://reviewboard.asterisk.org/r/965/ ........ + + * /, apps/app_directed_pickup.c: Merged revisions 290376 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290376 | dvossel | 2010-10-05 14:56:29 -0500 + (Tue, 05 Oct 2010) | 16 lines Merged revisions 290375 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010) + | 10 lines Fixes PickupChan() not working with full channel name. + (closes issue #18011) Reported by: schern Patches: + app_directed_pickup.c.2.patch uploaded by schern (license 995) + app_directed_pickup.c.trunk.patch uploaded by schern (license + 995) Tested by: schern, dvossel ........ ................ + +2010-10-05 14:17 +0000 [r290067-290291] Tilghman Lesher + + * /, configure, configure.ac: Merged revisions 290289 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r290289 | tilghman | 2010-10-05 09:15:46 -0500 (Tue, 05 + Oct 2010) | 2 lines Restore run directory for OS X, as well as + standardizing some other paths to Mac OS X. ........ + + * res/ael/pval.c, main/pbx.c, pbx/ael/ael-test/ref.ael-vtest17, /, + pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, + pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, + pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5, + pbx/ael/ael-test/ref.ael-test19, + pbx/ael/ael-test/ref.ael-vtest13: Merged revisions 290255 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290255 | tilghman | 2010-10-04 18:23:11 -0500 + (Mon, 04 Oct 2010) | 18 lines Merged revisions 290254 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010) + | 11 lines Change new pattern matcher to regard dashes the same + as the old pattern matcher -- as visual candy to be ignored. Also + change the AEL parser to not generate dashes within extensions, + as those dashes would be ignored. Update the AEL tests to match + this behavior. (closes issue #17366) Reported by: murf Patches: + 20100727__issue17366.diff.txt uploaded by tilghman (license 14) + Tested by: tilghman ........ ................ + + * /, configure, configure.ac: Merged revisions 290209 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290209 | tilghman | 2010-10-04 15:23:13 -0500 + (Mon, 04 Oct 2010) | 16 lines Merged revisions 290201 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r290201 | tilghman | 2010-10-04 15:22:03 -0500 + (Mon, 04 Oct 2010) | 9 lines Merged revisions 290177 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r290177 | tilghman | 2010-10-04 15:15:26 -0500 (Mon, 04 + Oct 2010) | 2 lines Fixing Mac OS X auto-builder. ........ + ................ ................ + + * /, configure, configure.ac: Merged revisions 290102 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r290102 | tilghman | 2010-10-03 16:08:45 -0500 + (Sun, 03 Oct 2010) | 16 lines Merged revisions 290101 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r290101 | tilghman | 2010-10-03 16:06:58 -0500 + (Sun, 03 Oct 2010) | 9 lines Merged revisions 290100 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r290100 | tilghman | 2010-10-03 16:04:29 -0500 (Sun, 03 + Oct 2010) | 2 lines Automatically re-run configure test for + menuselect, when the relevant makeopts settings change. ........ + ................ ................ + + * pbx/pbx_spool.c, /: Merged revisions 290066 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r290066 | tilghman | 2010-10-03 15:02:29 -0500 (Sun, 03 Oct 2010) + | 8 lines Get notification only when file is closed, not when + created. (closes issue #17924) Reported by: mkeuter Patches: + asterisk-1.8-bugid17924.patch uploaded by abelbeck (license 946) + Tested by: abelbeck ........ + +2010-10-02 18:06 +0000 [r290027] Kevin P. Fleming + + * contrib/scripts/get_mp3_source.sh, /: Merged revisions 290026 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r290026 | kpfleming | 2010-10-02 12:57:13 -0500 (Sat, 02 Oct + 2010) | 6 lines Allow users to pass additional arguments to the + Subversion command that obtains the MP-3 source code. (reported + on IRC by jmls) ........ + +2010-10-02 08:58 +0000 [r289952] Olle Johansson + + * main/manager.c, /: Merged revisions 289951 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289951 | oej | 2010-10-02 10:56:08 +0200 (Lör, + 02 Okt 2010) | 16 lines Merged revisions 289950 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör, + 02 Okt 2010) | 9 lines Merged revisions 289949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2 + lines Add documentation for undocumented option to AMI action + originate ........ ................ ................ + +2010-10-02 04:54 +0000 [r289876] Tilghman Lesher + + * /, apps/app_voicemail.c: Merged revisions 289875 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289875 | tilghman | 2010-10-01 23:46:43 -0500 + (Fri, 01 Oct 2010) | 22 lines Merged revisions 289874 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289874 | tilghman | 2010-10-01 23:45:49 -0500 + (Fri, 01 Oct 2010) | 15 lines Merged revisions 289873 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) + | 8 lines When forwarding a message, a prepend means that the + filesystem will always have a better copy. (closes issue #17803) + Reported by: dpetersen Patches: 20100923__issue17803.diff.txt + uploaded by tilghman (license 14) Tested by: dpetersen ........ + ................ ................ + +2010-10-02 02:46 +0000 [r289841] Jeff Peeler + + * res/res_rtp_asterisk.c, main/rtp_engine.c, /, + channels/chan_sip.c, include/asterisk/rtp_engine.h: Merged + revisions 289840 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289840 | jpeeler | 2010-10-01 21:43:45 -0500 + (Fri, 01 Oct 2010) | 29 lines Merged revisions 289798 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289798 | jpeeler | 2010-10-01 18:01:31 -0500 + (Fri, 01 Oct 2010) | 22 lines Merged revisions 289797 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289797 | jpeeler | 2010-10-01 17:58:38 -0500 (Fri, 01 Oct 2010) + | 15 lines Change RFC2833 DTMF event duration on end to report + actual elapsed time. The scenario here is with a non P2P early + media session. The reported time length of DTMF presses are + coming up short when sending to the remote side. Currently the + event duration is a running total that is incremented when + sending continuation packets. These continuation packets are only + triggered upon incoming media from the remote side, which means + that the running total probably is not going to end up matching + the actual length of time Asterisk received DTMF. This patch + changes the end event duration to be lengthened if it is detected + that the end event is going to come up short. Review: + https://reviewboard.asterisk.org/r/957/ ABE-2476 ........ + ................ ................ + +2010-10-01 17:22 +0000 [r289732] Paul Belanger + + * /, configs/jabber.conf.sample, res/res_jabber.c: Merged revisions + 289718 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289718 | pabelanger | 2010-10-01 13:19:49 -0400 + (Fri, 01 Oct 2010) | 20 lines Merged revisions 289704 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289704 | pabelanger | 2010-10-01 13:09:03 -0400 + (Fri, 01 Oct 2010) | 13 lines Merged revisions 289703 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289703 | pabelanger | 2010-10-01 13:03:11 -0400 (Fri, 01 Oct + 2010) | 6 lines Disable debugging by default and reformat .config + file. Review: https://reviewboard.asterisk.org/r/929/ ........ + ................ ................ + +2010-10-01 16:23 +0000 [r289702] Jeff Peeler + + * /, channels/chan_sip.c: Merged revisions 289701 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289701 | jpeeler | 2010-10-01 11:22:19 -0500 + (Fri, 01 Oct 2010) | 28 lines Merged revisions 289700 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289700 | jpeeler | 2010-10-01 11:21:04 -0500 + (Fri, 01 Oct 2010) | 21 lines Merged revisions 289699 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289699 | jpeeler | 2010-10-01 11:20:00 -0500 (Fri, 01 Oct 2010) + | 14 lines Ensure user portion of SIP URI matches dialplan when + using encoded characters. This commit takes a simliar approach to + 288112 and checks the dialplan to determine the proper action for + an incoming contact header as to whether or not it should be + decoded or not. sip_new was blindly always decoding the + extension, which also caused the outgoing contact header to be + incorrect as well as failing to match the encoded extension in + the dialplan. (closes issue #17892) Reported by: wdoekes Patches: + bug17892-1.patch uploaded by jpeeler (license 325) Tested by: + wdoekes ........ ................ ................ + +2010-10-01 10:04 +0000 [r289623] Stefan Schmidt + + * channels/chan_sip.c: don't iterate through all dialogs to find + and delete old subscribes On every incoming subscribe there is a + iteration through all dialogs to find old subscribes and delete + them. This is slow and not RFC conform. This was only needed in + 1.2 cause a subscribe was not deleted when a dialog was + destroyed, after 1.4 a subscribe get removed when its dialog is + destroyed. Review: https://reviewboard.asterisk.org/r/901/ + +2010-09-30 20:40 +0000 [r289588] Tilghman Lesher + + * /, tests/test_time.c, funcs/func_env.c, tests/test_utils.c, + res/res_agi.c, include/asterisk/localtime.h, + main/stdtime/localtime.c: Merged revisions 289543,289581 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289543 | tilghman | 2010-09-30 12:50:52 -0500 (Thu, 30 Sep 2010) + | 2 lines More Solaris compatibility fixes ........ r289581 | + tilghman | 2010-09-30 15:23:10 -0500 (Thu, 30 Sep 2010) | 2 lines + Solaris fixes. ........ + +2010-09-30 19:54 +0000 [r289555] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 289554 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289554 | mnicholson | 2010-09-30 14:53:10 -0500 + (Thu, 30 Sep 2010) | 11 lines Merged revisions 289553 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r289553 | mnicholson | 2010-09-30 14:51:27 -0500 (Thu, 30 Sep + 2010) | 4 lines Properly handle channel allocation failures duing + invites with replaces. ABE-2588 ........ ................ + +2010-09-30 19:35 +0000 [r289552] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 289549 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289549 | rmudgett | 2010-09-30 14:28:36 -0500 + (Thu, 30 Sep 2010) | 17 lines Merged revision 289547 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r289547 | rmudgett | 2010-09-30 14:16:36 -0500 (Thu, + 30 Sep 2010) | 10 lines In chan_misdn, the + DivertingLegInformation2 DivertingNr is garbage when the number + is restricted. The same thing happens with + DivertingLegInformation1 DivertedTo number. The + misdn_PresentedNumberUnscreened_extract() extracted the + Unscreened PartyNumber field unconditionally. It now checks the + presented number unscreened type to see if the PartyNumber was + even present. JIRA ABE-2595 .......... ................ + +2010-09-30 15:40 +0000 [r289427] Russell Bryant + + * /, apps/app_sms.c: Merged revisions 289426 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289426 | russell | 2010-09-30 10:39:45 -0500 + (Thu, 30 Sep 2010) | 22 lines Merged revisions 289425 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289425 | russell | 2010-09-30 10:37:29 -0500 + (Thu, 30 Sep 2010) | 15 lines Merged revisions 289424 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) + | 8 lines Fix a crash in app_sms. Since the data being passed to + the generator callback is on the stack of the SMS() application, + we must ensure that the generator is stopped before the + application exits. ABE-2587 ........ ................ + ................ + +2010-09-29 21:19 +0000 [r289354] Jason Parker + + * main/channel.c, /, main/features.c: Merged revisions 289340 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289340 | qwell | 2010-09-29 16:12:43 -0500 + (Wed, 29 Sep 2010) | 22 lines Merged revisions 289339 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289339 | qwell | 2010-09-29 16:03:47 -0500 + (Wed, 29 Sep 2010) | 15 lines Merged revisions 289338 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289338 | qwell | 2010-09-29 15:56:26 -0500 (Wed, 29 Sep 2010) | + 8 lines Allow a manager originate to succeed on forwarded + devices. The timeout to wait for an answer was being set to 0 + when a device forwarded to another extension. We don't always + need the timeout set like this, so make it an optional parameter, + and don't use it in this case. ABE-2544 ........ ................ + ................ + +2010-09-29 20:29 +0000 [r289337] Leif Madsen + + * /, configs/res_ldap.conf.sample: Merged revisions 289336 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289336 | lmadsen | 2010-09-29 15:27:25 -0500 + (Wed, 29 Sep 2010) | 9 lines Merged revisions 289334 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r289334 | lmadsen | 2010-09-29 15:24:47 -0500 (Wed, 29 + Sep 2010) | 1 line Update sample documentation to note md5secret + requirements. ........ ................ + +2010-09-29 20:24 +0000 [r289335] Russell Bryant + + * /, res/res_config_ldap.c: Merged revisions 289333 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289333 | russell | 2010-09-29 15:20:23 -0500 + (Wed, 29 Sep 2010) | 11 lines Merged revisions 289332 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r289332 | russell | 2010-09-29 15:15:57 -0500 (Wed, 29 Sep 2010) + | 4 lines Don't completely ignore md5secret from LDAP if the + value does not begin with {md5}. This fixes a problem that + lmadsen ran in to where md5secret was not working for him. + ........ ................ + +2010-09-29 17:54 +0000 [r289269-289301] Matthew Nicholson + + * /, configs/res_fax.conf.sample: Merged revisions 289300 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289300 | mnicholson | 2010-09-29 12:53:54 -0500 (Wed, 29 Sep + 2010) | 2 lines Add 'ecm' to the sample fax config file ........ + + * main/channel.c, /: Merged revisions 289268 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289268 | mnicholson | 2010-09-29 12:08:20 -0500 (Wed, 29 Sep + 2010) | 5 lines Update the CDR record when + ast_channel_set_caller_event() is called (related to issue + #17569) Reported by: tbelder ........ + +2010-09-29 16:17 +0000 [r289254] Richard Mudgett + + * main/channel.c, /: Merged revisions 289253 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289253 | rmudgett | 2010-09-29 11:16:47 -0500 (Wed, 29 Sep 2010) + | 1 line Make development error message indicate which channel. + ........ + +2010-09-29 15:07 +0000 [r289180] Matthew Nicholson + + * main/channel.c, /: Merged revisions 289179 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289179 | mnicholson | 2010-09-29 10:04:56 -0500 + (Wed, 29 Sep 2010) | 22 lines Merged revisions 289178 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289178 | mnicholson | 2010-09-29 10:04:11 -0500 + (Wed, 29 Sep 2010) | 15 lines Merged revisions 289177 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289177 | mnicholson | 2010-09-29 10:03:27 -0500 (Wed, 29 Sep + 2010) | 8 lines Set the caller id on CDRs when it is set on the + parent channel. (closes issue #17569) Reported by: tbelder + Patches: 17569.diff uploaded by tbelder (license 618) Tested by: + tbelder ........ ................ ................ + +2010-09-28 18:24 +0000 [r289131] Brett Bryant + + * main/channel.c, /: Merged revisions 289099 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r289099 | bbryant | 2010-09-28 14:18:02 -0400 + (Tue, 28 Sep 2010) | 28 lines Merged revisions 289095 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r289095 | bbryant | 2010-09-28 14:14:19 -0400 + (Tue, 28 Sep 2010) | 21 lines Merged revisions 289094 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r289094 | bbryant | 2010-09-28 14:10:19 -0400 (Tue, 28 Sep 2010) + | 14 lines Fixes an issue with the Newchannel AMI event during + the Masquerading process. Fixes an issue with the Newchannel AMI + event during the Masquerading process, where no Newchannel AMI + event was generated for the psuedo channel used during the + masquerading process. (closes issue #17987) Reported by: + RadicAlish Patches: newchannel.patch.txt uploaded by RadicAlish + (license 1122) Tested by: RadicAlish Review: + https://reviewboard.asterisk.org/r/937/ ........ ................ + ................ + +2010-09-28 18:20 +0000 [r289112] Tilghman Lesher + + * Makefile, /, tests/test_time.c, configure, + include/asterisk/autoconfig.h.in, include/asterisk/compat.h, + main/strcompat.c, tests/test_utils.c, configure.ac, makeopts.in, + apps/app_voicemail.c: Merged revisions 289104 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289104 | tilghman | 2010-09-28 13:18:43 -0500 (Tue, 28 Sep 2010) + | 4 lines Solaris compatibility fixes Review: + https://reviewboard.asterisk.org/r/942/ ........ + +2010-09-28 01:10 +0000 [r289056-289058] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 289057 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289057 | rmudgett | 2010-09-27 20:04:37 -0500 (Mon, 27 Sep 2010) + | 5 lines Avoid deadlock processing incoming AOC-E messages. + Deadlock avoidance for the owner channel was not done when + processing incoming AOC-E messages. ........ + + * /, channels/chan_sip.c: Merged revisions 289054-289055 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r289054 | rmudgett | 2010-09-27 19:32:18 -0500 (Mon, 27 Sep 2010) + | 1 line Break up long ast_manager_event_multichan() event lines. + ........ r289055 | rmudgett | 2010-09-27 19:35:25 -0500 (Mon, 27 + Sep 2010) | 1 line Revert stuff not ready for commit in -r289054. + ........ + +2010-09-27 22:03 +0000 [r289023] David Vossel + + * channels/chan_sip.c: For an INVITE transaction, treat all 2XX + responses the same as a 200. ABE-2305 + +2010-09-27 19:45 +0000 [r288992-288993] Olle Johansson + + * channels/chan_sip.c: Formatting fixes + + * cdr/cdr_pgsql.c: Formating changes + +2010-09-27 18:39 +0000 [r288962] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 288961 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288961 | tilghman | 2010-09-27 13:37:41 -0500 (Mon, 27 Sep 2010) + | 5 lines Still build SIP, even if res_crypto cannot be built + (use, not depend). (closes issue #18062) Reported by: a user on + the mailing list ........ + +2010-09-27 13:04 +0000 [r288926-288928] Russell Bryant + + * /, res/res_agi.c: Merged revisions 288927 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288927 | russell | 2010-09-27 08:03:43 -0500 (Mon, 27 Sep 2010) + | 2 lines Fix some documentation typos and spelling errors. + ........ + + * /, res/res_agi.c: Merged revisions 288925 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288925 | russell | 2010-09-27 07:42:10 -0500 (Mon, 27 Sep 2010) + | 2 lines Fix a documentation spelling error. ........ + +2010-09-25 07:58 +0000 [r288893] Alexandr Anikin + + * addons/ooh323c/src/oochannels.c: small correction for verbose + print h.323 packets + +2010-09-24 17:59 +0000 [r288822-288853] David Vossel + + * /, channels/chan_sip.c: Merged revisions 288852 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288852 | dvossel | 2010-09-24 12:58:57 -0500 (Fri, 24 Sep 2010) + | 5 lines Append Retry-After header on 500 error response to + Re-INVITE according to RFC3261 section 14.2. ABE-2301 ........ + + * /, channels/chan_sip.c: Merged revisions 288821 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288821 | dvossel | 2010-09-24 12:05:12 -0500 (Fri, 24 Sep 2010) + | 4 lines Inspect Require header on BYE transaction according to + RFC3261 section 8.2.2.3. ABE-2293 ........ + +2010-09-24 16:11 +0000 [r288749] Terry Wilson + + * channels/chan_local.c, /: Merged revisions 288748 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288748 | twilson | 2010-09-24 09:02:27 -0700 + (Fri, 24 Sep 2010) | 19 lines Merged revisions 288747 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288747 | twilson | 2010-09-24 08:37:39 -0700 + (Fri, 24 Sep 2010) | 12 lines Merged revisions 288746 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288746 | twilson | 2010-09-24 08:26:09 -0700 (Fri, 24 Sep 2010) + | 5 lines Don't fail a masquerade if it is already being hung up + This avoids noise on some Local channel situations where we don't + use /n. Thanks to Alec Davis for the suggestion. ........ + ................ ................ + +2010-09-24 13:55 +0000 [r288607-288714] Tilghman Lesher + + * /, funcs/func_strings.c: Merged revisions 288713 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288713 | tilghman | 2010-09-24 08:54:17 -0500 + (Fri, 24 Sep 2010) | 12 lines Merged revisions 288712 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24 Sep 2010) + | 5 lines Solaris won't printf a NULL. (closes issue #18041) + Reported by: asgaroth ........ ................ + + * /, main/asterisk.exports.in: Merged revisions 288640 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r288640 | tilghman | 2010-09-23 22:42:37 -0500 (Thu, 23 + Sep 2010) | 2 lines Export timersub for platforms which do not + have it ........ + + * /, configure, include/asterisk/autoconfig.h.in, + include/asterisk/compat.h, main/strcompat.c, configure.ac, + include/asterisk/channel.h, cdr/cdr_pgsql.c: Merged revisions + 288638 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288638 | tilghman | 2010-09-23 22:39:29 -0500 + (Thu, 23 Sep 2010) | 16 lines Merged revisions 288637 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288637 | tilghman | 2010-09-23 22:36:01 -0500 + (Thu, 23 Sep 2010) | 9 lines Merged revisions 288636 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r288636 | tilghman | 2010-09-23 22:20:24 -0500 (Thu, 23 + Sep 2010) | 2 lines Solaris compatibility fixes ........ + ................ ................ + + * /, CHANGES: Merged revisions 288606 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288606 | tilghman | 2010-09-23 13:44:44 -0500 (Thu, 23 Sep 2010) + | 2 lines Add note about the checkhangup option of ${CHANNEL()} + ........ + +2010-09-23 18:08 +0000 [r288519-288573] Terry Wilson + + * main/manager.c, /: Merged revisions 288572 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288572 | twilson | 2010-09-23 13:05:16 -0500 (Thu, 23 Sep 2010) + | 2 lines Make AMI honor enabled=no ........ + + * channels/chan_local.c, /: Merged revisions 288507 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288507 | twilson | 2010-09-22 16:18:27 -0700 + (Wed, 22 Sep 2010) | 22 lines Merged revisions 288500 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288500 | twilson | 2010-09-22 16:10:09 -0700 + (Wed, 22 Sep 2010) | 15 lines Merged revisions 288499 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288499 | twilson | 2010-09-22 16:00:30 -0700 (Wed, 22 Sep 2010) + | 8 lines Don't let a Local channel get bridged to itself If a + local channel gets bridged to itself, it becomes orphaned with no + devices left to actually tell it to hang up. This patch modifies + local_fixup() to detect this case and deny it. Review: + https://reviewboard.asterisk.org/r/934 ........ ................ + ................ + +2010-09-22 17:50 +0000 [r288346-288419] David Vossel + + * /, channels/chan_sip.c: Merged revisions 288418 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288418 | dvossel | 2010-09-22 12:49:56 -0500 + (Wed, 22 Sep 2010) | 18 lines Merged revisions 288417 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288417 | dvossel | 2010-09-22 12:49:05 -0500 + (Wed, 22 Sep 2010) | 11 lines Merged revisions 288416 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288416 | dvossel | 2010-09-22 12:48:15 -0500 (Wed, 22 Sep 2010) + | 5 lines RFC3261 section 12.2 explicitly says out of order + requests are responded with a 500 Server Internal Error response. + ABE-2458 ........ ................ ................ + + * /, channels/chan_sip.c: Merged revisions 288345 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288345 | dvossel | 2010-09-22 11:59:14 -0500 + (Wed, 22 Sep 2010) | 16 lines Merged revisions 288344 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288344 | dvossel | 2010-09-22 11:53:28 -0500 + (Wed, 22 Sep 2010) | 9 lines Merged revisions 288343 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r288343 | dvossel | 2010-09-22 11:49:56 -0500 (Wed, 22 + Sep 2010) | 2 lines During check_pendings, if the dialog is + terminated with a CANCEL, change the invitestate to INV_CANCEL + like in sip_hangup. ........ ................ ................ + +2010-09-22 16:46 +0000 [r288342] Russell Bryant + + * /, main/asterisk.c: Merged revisions 288341 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288341 | russell | 2010-09-22 11:45:18 -0500 + (Wed, 22 Sep 2010) | 25 lines Merged revisions 288340 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288340 | russell | 2010-09-22 11:44:13 -0500 + (Wed, 22 Sep 2010) | 18 lines Merged revisions 288339 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288339 | russell | 2010-09-22 11:39:16 -0500 (Wed, 22 Sep 2010) + | 11 lines Fix a 100% CPU consumption problem when setting + console=yes in asterisk.conf. The handling of -c and console=yes + should be the same, but they were not. When you specify -c, it + sets both a flag for console module and for asterisk not to + fork() off into the background. The handling of console=yes only + set console mode, so you would end up with a background process() + trying to run the Asterisk console and freaking out since it + didn't have anything to read input from. Thanks to beagles for + reporting and helping debug the problem! ........ + ................ ................ + +2010-09-22 15:18 +0000 [r288278] Tilghman Lesher + + * /, cdr/cdr_pgsql.c, configs/cdr_pgsql.conf.sample: Merged + revisions 288268 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288268 | tilghman | 2010-09-22 10:14:02 -0500 + (Wed, 22 Sep 2010) | 30 lines Merged revisions 288267 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288267 | tilghman | 2010-09-22 10:11:09 -0500 + (Wed, 22 Sep 2010) | 23 lines Merged revisions 288265-288266 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288265 | tilghman | 2010-09-22 09:48:04 -0500 (Wed, 22 Sep 2010) + | 9 lines Allow the encoding to be set, in case local charset + does not agree with database. (closes issue #16940) Reported by: + jamicque Patches: 20100827__issue16940.diff.txt uploaded by + tilghman (license 14) 20100921__issue16940__1.6.2.diff.txt + uploaded by tilghman (license 14) Tested by: jamicque ........ + r288266 | tilghman | 2010-09-22 10:04:52 -0500 (Wed, 22 Sep 2010) + | 5 lines Document addition of encoding parameter. (issue #16940) + Reported by: jamicque ........ ................ ................ + +2010-09-22 00:08 +0000 [r288195] Richard Mudgett + + * /, channels/chan_iax2.c: Merged revisions 288194 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288194 | rmudgett | 2010-09-21 19:06:21 -0500 + (Tue, 21 Sep 2010) | 40 lines Merged revisions 288193 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288193 | rmudgett | 2010-09-21 19:03:37 -0500 + (Tue, 21 Sep 2010) | 33 lines Merged revisions 288192 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288192 | rmudgett | 2010-09-21 18:55:58 -0500 (Tue, 21 Sep 2010) + | 26 lines In chan_iax2.c:schedule_delivery() calls + ast_bridged_channel() on an unlocked channel. Near the beginning + of schedule_delivery(), ast_bridged_channel() is called on + iaxs[fr->callno]->owner. However, the channel is not locked, + which can result in ast_bridged_channel() crashing should + owner->tech change to a technology that doesn't implement + bridged_channel. I also fixed the other calls to + ast_bridged_channel() in chan_iax2.c since the owner lock was not + held there either. Converted the existing channel deadlock + avoidance to use iax2_lock_owner(). Using the new function + simplified some awkward code. In the process of fixing the + locking on ast_bridged_channel(), I also found a memory leak in + socket_process() for v1.6.2 and v1.8. The local struct variable + ies.vars is not freed on early/abnormal function exits. (closes + issue #17919) Reported by: rain Patches: issue17919_v1.4.patch + uploaded by rmudgett (license 664) issue17919_w_leak_v1.6.2.patch + uploaded by rmudgett (license 664) issue17919_w_leak_v1.8.patch + uploaded by rmudgett (license 664) Review: + https://reviewboard.asterisk.org/r/926/ ........ ................ + ................ + +2010-09-21 22:58 +0000 [r288160] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 288159 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288159 | tilghman | 2010-09-21 17:57:22 -0500 + (Tue, 21 Sep 2010) | 29 lines Merged revisions 288113 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288113 | tilghman | 2010-09-21 16:59:46 -0500 + (Tue, 21 Sep 2010) | 22 lines Merged revisions 288112 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288112 | tilghman | 2010-09-21 16:58:13 -0500 (Tue, 21 Sep 2010) + | 15 lines Try both the encoded and unencoded subscription URI + for a match in hints. When a phone sends an encoded URI for a + subscription, the URI is not matched with the actual hint that is + in decoded format. For example, if we have an extension with a + hint that is named: "#5601" or "*5601", the subscription will + work fine if the phone subscribes with an already decoded URI, + but when it's decoded like "%255601" or "%2A5601", Asterisk is + unable to match it with the correct hint. (closes issue #17785) + Reported by: ramonpeek Patches: 20100831__issue17785.diff.txt + uploaded by tilghman (license 14) Tested by: ramonpeek ........ + ................ ................ + +2010-09-21 22:28 +0000 [r288158] Paul Belanger + + * /, channels/chan_iax2.c: Merged revisions 288157 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288157 | pabelanger | 2010-09-21 18:26:15 -0400 + (Tue, 21 Sep 2010) | 15 lines Merged revisions 288147 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r288147 | pabelanger | 2010-09-21 18:22:43 -0400 (Tue, 21 Sep + 2010) | 9 lines Setup timer before set_config(). (closes issue + #18019) Reported by: Netview Patches: issue_0018019.patch + uploaded by pabelanger (license 224) Tested by: Netview ........ + ................ + +2010-09-21 21:04 +0000 [r288081-288083] Richard Mudgett + + * /, doc/tex/partymanip.tex: Merged revisions 288082 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r288082 | rmudgett | 2010-09-21 16:03:28 -0500 (Tue, 21 + Sep 2010) | 1 line Add note in party manipulation chapter on + interception macros. ........ + + * apps/app_dial.c, main/channel.c, /, apps/app_queue.c: Merged + revisions 288079-288080 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) + | 2 lines Protect channel access in CONNECTED_LINE and + REDIRECTING interception macro launch code. ........ r288080 | + rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines + Simplify locking code for REDIRECTING interception macro when + forwarding a call. Simplified the locking code by using a local + copy of the redirecting party information in + app_dial.c:do_forward() and app_queue.c:wait_for_answer() for + launching the REDIRECTING interception macro when a call is + forwarded. Reduced the lock time of the 'o->chan' and 'in' + channels. ........ + +2010-09-21 20:27 +0000 [r288063] Stefan Schmidt + + * channels/chan_sip.c: Instead of iterate through all dialogs, add + two separte container for needdestroy and rtptimeout adding two + dialog container, one for dialogs which need destroy, another for + rtptimeout checks. both container will be checked on every loop + of do_monitor instead of iterate through all dialogs. (closes + issue #17912) Reported by: schmidts Tested by: schmidts Review: + https://reviewboard.asterisk.org/r/917/ + +2010-09-21 19:50 +0000 [r288008] Brett Bryant + + * main/channel.c, /: Merged revisions 288007 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r288007 | bbryant | 2010-09-21 15:48:53 -0400 + (Tue, 21 Sep 2010) | 21 lines Merged revisions 288006 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r288006 | bbryant | 2010-09-21 15:46:20 -0400 + (Tue, 21 Sep 2010) | 14 lines Merged revisions 288005 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r288005 | bbryant | 2010-09-21 15:43:46 -0400 (Tue, 21 Sep 2010) + | 8 lines Add a check to fix a rare segmentation fault you'd get + if ast_frdup couldn't allocate memory on the first frame being + queued in ast_queue_frame. (closes issue #17882) Reported by: + seanbright Tested by: seanbright ........ ................ + ................ + +2010-09-21 19:09 +0000 [r287936] Tilghman Lesher + + * /, main/asterisk.c: Merged revisions 287935 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287935 | tilghman | 2010-09-21 14:08:36 -0500 + (Tue, 21 Sep 2010) | 16 lines Merged revisions 287934 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287934 | tilghman | 2010-09-21 14:07:53 -0500 + (Tue, 21 Sep 2010) | 9 lines Merged revisions 287933 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r287933 | tilghman | 2010-09-21 14:07:07 -0500 (Tue, 21 + Sep 2010) | 2 lines Less than zero is an error, not any non-zero + value. ........ ................ ................ + +2010-09-21 19:04 +0000 [r287932] Terry Wilson + + * main/channel.c, /: Merged revisions 287931 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287931 | twilson | 2010-09-21 14:02:40 -0500 (Tue, 21 Sep 2010) + | 2 lines Revert change in favor of a more targeted fix ........ + +2010-09-21 18:33 +0000 [r287930] David Vossel + + * /, channels/chan_sip.c: Merged revisions 287929 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287929 | dvossel | 2010-09-21 13:32:12 -0500 (Tue, 21 Sep 2010) + | 4 lines Send a "415 Unsupported Media Type" after failure to + process sdp due to unknown Content-Encoding header. ABE-2258 + ........ + +2010-09-21 15:54 +0000 [r287898] Richard Mudgett + + * /, main/features.c: Merged revisions 287897 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287897 | rmudgett | 2010-09-21 10:53:19 -0500 (Tue, 21 Sep 2010) + | 1 line Cut-n-paste error in builtin_blindtransfer(). ........ + +2010-09-21 15:45 +0000 [r287896] Russell Bryant + + * res/res_rtp_asterisk.c, main/dnsmgr.c, /, channels/chan_sip.c, + main/acl.c: Merged revisions 287895 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287895 | russell | 2010-09-21 10:43:33 -0500 (Tue, 21 Sep 2010) + | 10 lines Don't use ast_strdupa() from within the arguments to a + function. (closes issue #17902) Reported by: afried Patches: + issue_17902.rev1.txt uploaded by russell (license 2) Tested by: + russell Review: https://reviewboard.asterisk.org/r/927/ ........ + +2010-09-21 15:27 +0000 [r287894] Tilghman Lesher + + * /, channels/chan_sip.c: Merged revisions 287893 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287893 | tilghman | 2010-09-21 10:24:47 -0500 (Tue, 21 Sep 2010) + | 9 lines Anonymous callerid needs a "sip:" uri prefix. (closes + issue #17981) Reported by: avalentin Patches: + sip-anonymous-aastra.patch uploaded by avalentin (license 1107) + (plus an additional fix by me) Tested by: avalentin ........ + +2010-09-21 13:45 +0000 [r287864] Russell Bryant + + * /, main/logger.c: Merged revisions 287863 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287863 | russell | 2010-09-21 08:41:41 -0500 (Tue, 21 Sep 2010) + | 2 lines Fix a regression in verbose logger processing. ........ + +2010-09-21 04:39 +0000 [r287764-287834] Terry Wilson + + * main/channel.c, /: Merged revisions 287833 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287833 | twilson | 2010-09-20 23:37:44 -0500 (Mon, 20 Sep 2010) + | 3 lines Don't generate connected line buffer twice for + comparison ........ + + * main/channel.c, /: Merged revisions 287757 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287757 | twilson | 2010-09-20 18:51:38 -0500 (Mon, 20 Sep 2010) + | 7 lines Avoid infinite loop with certain local channel + connected line updates Compare connected line data before sending + a connected line indication to avoid possible loops. Review: + https://reviewboard.asterisk.org/r/932/ ........ + +2010-09-21 00:04 +0000 [r287763] Brett Bryant + + * /, apps/app_meetme.c: Merged revisions 287760 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287760 | bbryant | 2010-09-20 20:00:23 -0400 + (Mon, 20 Sep 2010) | 30 lines Merged revisions 287759 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287759 | bbryant | 2010-09-20 19:58:26 -0400 + (Mon, 20 Sep 2010) | 23 lines Merged revisions 287758 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) + | 16 lines Fix misvalidation of meetme pins in conjunction with + the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a + user and admin pin setup for your conference, using the user pin + would gain you admin priviledges. Also, when no user pin was set, + an admin pin was, the 'a' MeetMe flag wasn't used, and the user + tried to enter a conference then they were still prompted for a + pin and forced to hit #. (closes issue #17908) Reported by: kuj + Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: + kuj Review: [full review board URL with trailing slash] ........ + ................ ................ + +2010-09-21 00:01 +0000 [r287761-287762] Terry Wilson + + * /: Add alecdavis' commit to merged props + + * /: Add merge properties back. + +2010-09-20 23:42 +0000 [r287756] Alec L Davis + + * main/channel.c, /: Merged revisions 287685 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r287685 | alecdavis | 2010-09-21 11:16:45 +1200 (Tue, 21 Sep + 2010) | 18 lines ast_channel_masquerade: Avoid recursive + masquerades. Check all 4 combinations of (original/clonechan) * + (masq/masqr). Initially original->masq and clonechan->masqr were + only checked. It's possible with multiple masq's planned - and + not yet executed, that the 'original' chan could already have + another masq'd into it - thus original->masqr would be set, that + masqr would lost. Likewise for the clonechan->masq. (closes issue + #16057;#17363) Reported by: amorsen;davidw,alecdavis Patches: + based on bug16057.diff4.txt uploaded by alecdavis (license 585) + Tested by: ramonpeek, davidw, alecdavis ........ + +2010-09-20 23:18 +0000 [r287693] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 287683 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r287683 | rmudgett | 2010-09-20 18:14:42 -0500 (Mon, 20 + Sep 2010) | 9 lines The inalarm flag was not set in sig_analog + struct if the port is initially in alarm. Fixed initial inalarm + value for sig_analog ports. Along with -r261007, this gets the + inalarm flag in sync with chan_dahdi for sig_analog ports. + (closes issue #16983) ........ + +2010-09-20 22:24 +0000 [r287671] Alec L Davis + + * main/channel.c, /: Merged revisions 287661 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287661 | alecdavis | 2010-09-21 10:21:50 +1200 (Tue, 21 Sep + 2010) | 14 lines ast_do_masquerade. Keep channels ao2_container + locked while unlink and linking channels. Previously, Masquerade + would unlock 'original' and 'clonechan' and allow another masq + thread to run. End result would be corrupted memory, and the + frequent report 'Bad Magic Number'. (closes issue #17801,#17710) + Reported by: notthematrix Patches: Based on bug17801.diff1.txt + uploaded by alecdavis (license 585) Tested by: alecdavis Review: + https://reviewboard.asterisk.org/r/928 ........ + +2010-09-20 22:16 +0000 [r287646-287648] David Vossel + + * main/channel.c, main/framehook.c (added), /, + funcs/func_frame_trace.c (added), include/asterisk/channel.h, + CHANGES, include/asterisk/framehook.h (added): Merged revisions + 287647 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) + | 21 lines Addition of the FrameHook API (AKA AwesomeHooks) So + far all our tools for viewing and manipulating media streams + within Asterisk have been entirely focused on audio. That made + sense then, but is not scalable now. The FrameHook API lets us + tap into and manipulate _ANY_ type of media or signaling passed + on a channel present today or in the future. This tool is a step + in the direction of expanding Asterisk's boundaries and will help + generate some rather interesting applications in the future. In + addition to the FrameHook API, a simple dialplan function + exercising the api has been included as well. This function is + called FRAME_TRACE(). FRAME_TRACE() allows for the internal + ast_frames read and written to a channel to be output. Filters + can be placed on this function to debug only certain types of + frames. This function could be thought of as an internal way of + doing ast_frame packet captures. Review: + https://reviewboard.asterisk.org/r/925/ ........ + + * /, channels/chan_sip.c: Merged revisions 287645 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287645 | dvossel | 2010-09-20 16:34:15 -0500 (Mon, 20 Sep 2010) + | 9 lines Fixes issue with registrations not working properly + with pedantic=yes. (closes issue #18017) Reported by: schmidts + Patches: issues_18017_v1.diff uploaded by dvossel (license 671) + Tested by: schmidts ........ + +2010-09-20 21:30 +0000 [r287644] Jason Parker + + * /, channels/chan_skinny.c: Merged revisions 287643 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287643 | qwell | 2010-09-20 16:29:46 -0500 + (Mon, 20 Sep 2010) | 15 lines Merged revisions 287642 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r287642 | qwell | 2010-09-20 16:28:32 -0500 (Mon, 20 Sep 2010) | + 8 lines Don't crash when parking a non-bridged call. (closes + issue #17680) Reported by: jmhunter Patches: + chan_skinny-park-v1.txt uploaded by DEA (license 3) Tested by: + jmhunter, DEA ........ ................ + +2010-09-20 21:25 +0000 [r287640] Brett Bryant + + * /, main/logger.c: Merged revisions 287639 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287639 | bbryant | 2010-09-20 17:19:12 -0400 (Mon, 20 Sep 2010) + | 8 lines Fixes an error with the logger that caused verbose + messages to be spammed to the screen if syslog was configured in + logger.conf (closes issue #17974) Reported by: lmadsen Review: + https://reviewboard.asterisk.org/r/915/ ........ + +2010-09-20 15:57 +0000 [r287560] Matthew Nicholson + + * main/pbx.c, /: Merged revisions 287559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287559 | mnicholson | 2010-09-20 10:57:14 -0500 + (Mon, 20 Sep 2010) | 21 lines Merged revisions 287558 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287558 | mnicholson | 2010-09-20 10:56:21 -0500 + (Mon, 20 Sep 2010) | 14 lines Use ast_str when processing hint + state changes Merged revisions 287555 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287555 | mnicholson | 2010-09-20 10:48:14 -0500 (Mon, 20 Sep + 2010) | 5 lines Use ast_dynamic_str when processing hint state + changes (related to issue #17928) Reported by: mdu113 ........ + ................ ................ + +2010-09-19 16:12 +0000 [r287472] Olle Johansson + + * main/manager.c, /: Merged revisions 287471 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287471 | oej | 2010-09-19 18:09:28 +0200 (Sön, + 19 Sep 2010) | 21 lines Merged revisions 287470 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön, + 19 Sep 2010) | 14 lines Merged revisions 287469 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7 + lines Make sure we always free variables properly in manager + originate. (closes issue #17891) reported, solved and tested by + oej Review: https://reviewboard.asterisk.org/r/869/ ........ + ................ ................ + +2010-09-17 21:10 +0000 [r287389] Tilghman Lesher + + * /, apps/app_queue.c: Merged revisions 287388 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287388 | tilghman | 2010-09-17 16:08:54 -0500 + (Fri, 17 Sep 2010) | 21 lines Merged revisions 287387 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287387 | tilghman | 2010-09-17 16:08:00 -0500 + (Fri, 17 Sep 2010) | 14 lines Merged revisions 287386 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) + | 7 lines Blank columns should get set on reload, not ignored. + (closes issue #16893) Reported by: haakon Patches: + 20100818__issue16893.diff.txt uploaded by tilghman (license 14) + ........ ................ ................ + +2010-09-17 13:38 +0000 [r287310] Matthew Nicholson + + * main/pbx.c, /: Merged revisions 287309 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287309 | mnicholson | 2010-09-17 08:37:10 -0500 + (Fri, 17 Sep 2010) | 19 lines Merged revisions 287308 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287308 | mnicholson | 2010-09-17 08:36:07 -0500 + (Fri, 17 Sep 2010) | 12 lines Merged revisions 287307 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287307 | mnicholson | 2010-09-17 08:34:34 -0500 (Fri, 17 Sep + 2010) | 5 lines Use ast_strdup() instead of ast_strdupa() while + processing in ast_hint_state_changed(). (related to issue #17928) + Reported by: mdu113 ........ ................ ................ + +2010-09-17 08:46 +0000 [r287272] Jan Kalab + + * res/res_calendar_exchange.c, res/res_calendar_icalendar.c, /, + res/res_calendar_caldav.c, res/res_calendar_ews.c: Merged + revisions 287269-287271 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287269 | pitel | 2010-09-17 10:37:49 +0200 (Pá, 17 zář 2010) | 8 + lines Support for HTTP redirects in calendar's URL libneon does + not support HTTP redirects (3xx responses) by default. You must + tell it to follow them. Also, another little unsigned int fix. + (closes issue #17776) Review: + https://reviewboard.asterisk.org/r/921/ ........ r287270 | pitel + | 2010-09-17 10:42:37 +0200 (Pá, 17 zář 2010) | 6 lines Asterisk + crashing because of double free when EWS request fails The free + is done later in code. I think ast_free() should have built in + checks for double free. (closes issue #17782) ........ r287271 | + pitel | 2010-09-17 10:44:28 +0200 (Pá, 17 zář 2010) | 6 lines + Events are visible after they were removed from EWS calendar + Because we must merge calendar even when it's empty. (closes + issue #17786) ........ + +2010-09-16 22:05 +0000 [r287196] Jason Parker + + * contrib/init.d/rc.debian.asterisk, /: Merged revisions 287195 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287195 | qwell | 2010-09-16 17:04:38 -0500 (Thu, 16 Sep 2010) | + 7 lines Don't fail when running the Debian init script directly + (as one would normally do). readlink apparently returns 1 when + the arg isn't a symlink, which caused the script to exit. (closes + issue #17910) Reported by: wurstsalat ........ + +2010-09-16 22:00 +0000 [r287194] Russell Bryant + + * /, configs/queues.conf.sample, apps/app_queue.c, UPGRADE-1.8.txt: + Merged revisions 287193 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287193 | russell | 2010-09-16 16:57:51 -0500 (Thu, 16 Sep 2010) + | 4 lines Set the default for "autofill" and "shared_lastcall" to + "yes" in queues.conf. Review: + https://reviewboard.asterisk.org/r/922/ ........ + +2010-09-16 20:08 +0000 [r287117-287121] Matthew Nicholson + + * main/pbx.c, /: Merged revisions 287120 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287120 | mnicholson | 2010-09-16 15:07:38 -0500 + (Thu, 16 Sep 2010) | 22 lines Merged revisions 287119 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287119 | mnicholson | 2010-09-16 15:06:16 -0500 + (Thu, 16 Sep 2010) | 15 lines Merged revisions 287118 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287118 | mnicholson | 2010-09-16 15:04:46 -0500 (Thu, 16 Sep + 2010) | 8 lines Don't limit hint processing in + ast_hint_state_changed() to AST_MAX_EXTENSION length strings. + (closes issue #17928) Reported by: mdu113 Patches: + 20100831__issue17928.diff.txt uploaded by tilghman (license 14) + Tested by: mdu113 ........ ................ ................ + + * main/cdr.c, /: Merged revisions 287116 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287116 | mnicholson | 2010-09-16 14:54:48 -0500 + (Thu, 16 Sep 2010) | 22 lines Merged revisions 287115 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r287115 | mnicholson | 2010-09-16 14:53:41 -0500 + (Thu, 16 Sep 2010) | 15 lines Merged revisions 287114 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r287114 | mnicholson | 2010-09-16 14:52:39 -0500 (Thu, 16 Sep + 2010) | 8 lines Don't stop printing cdr variables if we encounter + one with a blank name or value. (closes issue #17900) Reported + by: under Patches: core-show-channel-cdr-fix1.diff uploaded by + mnicholson (license 96) Tested by: mnicholson ........ + ................ ................ + +2010-09-16 16:49 +0000 [r287086-287087] Olle Johansson + + * channels/chan_sip.c: We do not handle AST_CAUSE_INTERWORKING + which we set on a lot of incoming SIP messages. Adding error + based on RFC 3398 recommendations. + + * main/indications.c: Add doxygen docs for indications.c + +2010-09-15 22:28 +0000 [r287057] Terry Wilson + + * /, res/res_srtp.c: Merged revisions 287056 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287056 | twilson | 2010-09-15 17:17:17 -0500 (Wed, 15 Sep 2010) + | 10 lines Don't hang up a call on an SRTP unprotect failure Also + make it more obvious when there is an issue en/decrypting. + (closes issue #17563) Reported by: Alexcr Patches: + res_srtp.c.patch uploaded by sfritsch (license 1089) Tested by: + twilson ........ + +2010-09-15 21:00 +0000 [r287021] Jeff Peeler + + * /, main/features.c: Merged revisions 287020 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r287020 | jpeeler | 2010-09-15 15:58:39 -0500 (Wed, 15 Sep 2010) + | 1 line fix uninintialized variable ........ + +2010-09-15 20:56 +0000 [r287018] Richard Mudgett + + * channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c: + Merged revisions 287017 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287017 | rmudgett | 2010-09-15 15:53:38 -0500 + (Wed, 15 Sep 2010) | 65 lines Merged revision 287014 from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + .......... r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, + 15 Sep 2010) | 58 lines The handling of call transfer signaling + for mISDN PTMP is not fully implemented. The handling of call + transfer signaling for mISDN PTMP is not fully implemented. The + signaling of number updates with ISDN/DSS1 ECT supplementary + services (ETS 300 369-1) comes along with a notification + indicator IE and redirection number IE for PTMP. The + implementation in the current Asterisk mISDN channel + unfortunately can handle these information elements only in a + NOTIFY message. These information elements are also signaled in a + FACILTY message with a RequestSubaddress facility, when the + subscriber is already in the active state (see 9.2.4 and 9.2.5 of + ETS 300 369-1). ********** abe_2526_ast.patch * Added support to + handle the notification indicator IE and redirection number IE + with the RequestSubaddress facility. * Made + misdn_update_connected_line() send a NOTIFY message if Asterisk + originated the call and it is not connected yet. * Made + misdn_update_connected_line() send a FACILITY message if the call + is already connected. This patch requires the presence of the + associated mISDN patches to compile. I had to enhance mISDN to + allow the notification indicator IE and the redirection number IE + to be used with a FACILITY message. Earlier versions of the + Digium enhanced mISDN are no longer going to work. ********** + abe_2526_misdn.patch * Made an incoming FACILITY message allow + the presence of the notification indicator IE and the redirection + number IE. ********** abe_2526_misdnuser_v3.patch * Added support + to send and receive a FACILITY message with the notification + indicator IE and the redirection number IE. * Added the ability + to send a NOTIFY message in PTMP/NT mode to all responding + subcalls in Q.931 states 6, 7, 8, 9, and 25. ********** Patches: + abe_2526_ast.patch uploaded by rmudgett (license 664) + abe_2526_misdn.patch uploaded by rmudgett (license 664) + abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664) + Tested by: rmudgett and reporter JIRA SWP-2146 JIRA ABE-2526 + .......... ................ + +2010-09-15 20:36 +0000 [r286939-287016] Jeff Peeler + + * /, apps/app_voicemail.c: Merged revisions 287015 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r287015 | jpeeler | 2010-09-15 15:32:52 -0500 + (Wed, 15 Sep 2010) | 21 lines Merged revisions 286998 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286998 | jpeeler | 2010-09-15 15:28:02 -0500 + (Wed, 15 Sep 2010) | 14 lines Merged revisions 286941 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) + | 7 lines Ensure mailbox is not filled to capacity before doing + message forwarding. Specifically, before prompting to record a + prepended message the capacity is checked first. If the mailbox + is full the extension will be reprompted. ABE-2517 ........ + ................ ................ + + * channels/chan_dahdi.c, channels/sig_analog.c, /, + channels/chan_sip.c, main/features.c, CHANGES, + channels/chan_iax2.c, channels/sip/include/sip.h, + configs/features.conf.sample, channels/chan_mgcp.c, + include/asterisk/features.h: Merged revisions 286931 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 + Sep 2010) | 16 lines Add parking extension for non-default + parking lots. This is a new feature that allows for parking to + custom parking lots to be accessed directly, rather than with + channel variables or by changing the default parking lot. The + extension is set with the parkext option just as the default + parking lot is done. Also, the manager action has been updated to + optionally allow a specified parking lot. (closes issue #14882) + Reported by: vmikhnevych Patches: patch_14882.txt uploaded by + mnick (license 874) modified by me Review: + https://reviewboard.asterisk.org/r/884/ ........ + +2010-09-15 18:30 +0000 [r286906] Richard Mudgett + + * channels/sig_analog.c, /: Merged revisions 286904-286905 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286904 | rmudgett | 2010-09-15 13:28:05 -0500 (Wed, 15 Sep 2010) + | 12 lines Unable to originate calls using E&M over T1. When + originating a call from Unit Under Test to Reference Unit using + E&M RBS signaling mode, I get the following warning message: + "Ring/Off-hook in strange state 3 on channel 1". Fixed the + sig_analog outgoing flag. It was never set when sig_analog was + extracted from chan_dahdi. JIRA SWP-2191 JIRA AST-408 ........ + r286905 | rmudgett | 2010-09-15 13:29:21 -0500 (Wed, 15 Sep 2010) + | 1 line Simplify some code in sig_analog. ........ + +2010-09-15 13:10 +0000 [r286869] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 286868 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286868 | mnicholson | 2010-09-15 08:05:52 -0500 (Wed, 15 Sep + 2010) | 16 lines Set tohost to the domain specified in the + configuration file instead of the IP address of the host we are + calling. This fixes a regression introduced in r274783. (closes + issue #17960) Reported by: adriavidal Patches: + sip-tohost-fix1.diff uploaded by mnicholson (license 96) Tested + by: mich, mnicholson, adriavidal (closes issue #17676) Reported + by: outcast Patches: sip-tohost-fix1.diff uploaded by mnicholson + (license 96) Tested by: mnicholson ........ + +2010-09-14 22:02 +0000 [r286835] David Vossel + + * /, channels/chan_sip.c: Merged revisions 286834 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286834 | dvossel | 2010-09-14 16:57:35 -0500 (Tue, 14 Sep 2010) + | 2 lines Sets subscribed type for outgoing MWI subscriptions so + correct Event header is used. ........ + +2010-09-14 19:29 +0000 [r286683-286759] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 286758 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286758 | mnicholson | 2010-09-14 14:28:38 -0500 + (Tue, 14 Sep 2010) | 27 lines Merged revisions 286757 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286757 | mnicholson | 2010-09-14 14:27:28 -0500 + (Tue, 14 Sep 2010) | 20 lines Merged revisions 286756 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286756 | mnicholson | 2010-09-14 14:26:18 -0500 (Tue, 14 Sep + 2010) | 13 lines Don't clear the username from a realtime + database when a registration expires. Non-realtime chan_sip does + not clear the username from memory when a registration expiries + so realtime probably shouldn't either. (closes issue #17551) + Reported by: ricardolandim Patches: + reg-expiry-username-1.4-fix1.diff uploaded by mnicholson (license + 96) reg-expiry-username-1.6.2-fix1.diff uploaded by mnicholson + (license 96) reg-expiry-username-1.8-fix1.diff uploaded by + mnicholson (license 96) reg-expiry-username-trunk-fix1.diff + uploaded by mnicholson (license 96) Tested by: ricardolandim, + mnicholson ........ ................ ................ + + * main/channel.c, /: Merged revisions 286682 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286682 | mnicholson | 2010-09-14 13:04:21 -0500 + (Tue, 14 Sep 2010) | 21 lines Merged revisions 286681 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286681 | mnicholson | 2010-09-14 13:02:24 -0500 + (Tue, 14 Sep 2010) | 14 lines Merged revisions 286679 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286679 | mnicholson | 2010-09-14 13:00:01 -0500 (Tue, 14 Sep + 2010) | 7 lines Only drop duplicate answer frames if the channel + is bridged. Back in r3710 ast_read() was modified to drop answer + frames on channels that were in the UP state. This modification + prevented bridges that were up before the answer from being + broken and reestablished by an ANSWER control frame. That change + also prevents pickup of channels called from the ast_dial + framework from working properly. The ast_dial framework expects + to see an ANSWER frame after dialing and the pickup code queues + one but ast_read() drops it. This new change only drops ANSWER + frames when the channel is bridged, allowing the answer queued by + the pickup code to properly pass through ast_read() on to the + ast_dial framework. ABE-2473 (related to issue #2342) ........ + ................ ................ + +2010-09-14 15:31 +0000 [r286648] Richard Mudgett + + * /, doc/tex/channelvariables.tex, doc/tex/partymanip.tex: Merged + revisions 286647 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286647 | rmudgett | 2010-09-14 10:30:49 -0500 (Tue, 14 Sep 2010) + | 1 line Corrected documented CONNECTED_LINE and REDIRECTING + party manipulation macro names. ........ + +2010-09-14 06:58 +0000 [r286618] Jan Kalab + + * /, res/res_calendar_ews.c: Merged revisions 286617 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r286617 | pitel | 2010-09-14 08:55:44 +0200 (Út, 14 zář + 2010) | 7 lines Merging events for Exchange web service doesn't + work as expected, resulting in only one event in calendar The + solution is to use "global" counter of events, since we do new + requests for every event and calendar sync after every request. + So now we do sync only after last request. (closes issue #17877) + Review: https://reviewboard.asterisk.org/r/916/ ........ + +2010-09-14 05:08 +0000 [r286529-286589] Tilghman Lesher + + * /, contrib/realtime/mysql/voicemail_messages.sql (added), + contrib/realtime/mysql/voicemail_data.sql (added): Merged + revisions 286588 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286588 | tilghman | 2010-09-14 00:07:16 -0500 + (Tue, 14 Sep 2010) | 9 lines Merged revisions 286587 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r286587 | tilghman | 2010-09-14 00:06:05 -0500 (Tue, 14 + Sep 2010) | 2 lines Add documentation on missing backend tables + for Voicemail ........ ................ + + * /, main/features.c: Merged revisions 286558 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286558 | tilghman | 2010-09-13 18:50:34 -0500 + (Mon, 13 Sep 2010) | 9 lines Merged revisions 286557 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r286557 | tilghman | 2010-09-13 18:48:51 -0500 (Mon, 13 + Sep 2010) | 2 lines C precedence got me ........ ................ + + * /, main/features.c: Merged revisions 286528 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286528 | tilghman | 2010-09-13 18:12:21 -0500 + (Mon, 13 Sep 2010) | 9 lines Merged revisions 286527 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r286527 | tilghman | 2010-09-13 18:03:26 -0500 (Mon, 13 + Sep 2010) | 2 lines Refactor conversion to ast_poll() to fix + callparking regression. ........ ................ + +2010-09-13 22:13 +0000 [r286498] Russell Bryant + + * /, main/db.c: Merged revisions 286112 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286112 | russell | 2010-09-10 15:31:58 -0500 (Fri, 10 Sep 2010) + | 9 lines Rate limit calls to fsync() to 1 per second after astdb + updates. Astdb was determined to be one of the most significant + bottlenecks in SIP registration processing. This patch improved + the speed of an astdb load test by 50000% (yes, Fifty-Thousand + Percent). On this particular load test setup, this doubled the + number of SIP registrations the server could handle. Review: + https://reviewboard.asterisk.org/r/825/ ........ + +2010-09-13 19:40 +0000 [r286458] Jason Parker + + * /, channels/chan_sip.c: Merged revisions 286457 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286457 | qwell | 2010-09-13 14:40:05 -0500 + (Mon, 13 Sep 2010) | 12 lines Merged revisions 286456 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r286456 | qwell | 2010-09-13 14:38:35 -0500 (Mon, 13 Sep 2010) | + 5 lines Remove "Internal IP" from sip show settings, as it's not + at all useful to display. (closes issue #17840) Reported by: oej + ........ ................ + +2010-09-13 15:53 +0000 [r286427] Richard Mudgett + + * configs/chan_dahdi.conf.sample, /: Merged revisions 286426 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r286426 | rmudgett | 2010-09-13 10:52:14 -0500 (Mon, 13 Sep 2010) + | 1 line Update chan_dahdi.conf.sample to reflect new libpri T309 + default value. ........ + +2010-09-11 17:35 +0000 [r286271-286342] Olle Johansson + + * main/say.c, main/app.c: Whitespace cleanup + + * main/features.c: Whitespace cleanup and reformatting with { and } + + * /, main/file.c: Merged revisions 286270 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286270 | oej | 2010-09-11 19:09:22 +0200 (Lör, + 11 Sep 2010) | 18 lines Merged revisions 286268 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286268 | oej | 2010-09-11 19:05:16 +0200 (Lör, + 11 Sep 2010) | 11 lines Merged revisions 286267 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286267 | oej | 2010-09-11 18:59:20 +0200 (Lör, 11 Sep 2010) | 4 + lines Handle error response when we can't make file compatible + Review: https://reviewboard.asterisk.org/r/911/ ........ + ................ ................ + + * channels/chan_sip.c: Formatting changes. + +2010-09-10 22:15 +0000 [r286190] Terry Wilson + + * channels/chan_local.c, /, funcs/func_channel.c, + include/asterisk/channel.h, include/asterisk/pbx.h, + include/asterisk/frame.h: Merged revisions 286189 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286189 | twilson | 2010-09-10 17:04:53 -0500 + (Fri, 10 Sep 2010) | 30 lines Merged revisions 286115 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286115 | twilson | 2010-09-10 15:35:25 -0500 + (Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) + | 16 lines Inherit CHANNEL() writes to both sides of a Local + channel Having Local (/n) channels as queue members and setting + the language in the extension with Set(CHANNEL(language)=fr) sets + the language on the Local/...,2 channel. Hold time report + playbacks happen on the Local/...,1 channel and therefor do not + play in the specified language. This patch modifies + func_channel_write to call the setoption callback and pass the + CHANNEL() write info to the callback. chan_local uses this + information to look up the other side of the channel and apply + the same changes to it. (closes issue #17673) Reported by: + Guggemand Review: https://reviewboard.asterisk.org/r/903/ + ........ ................ ................ + +2010-09-10 21:13 +0000 [r286121] Paul Belanger + + * /, channels/chan_iax2.c: Merged revisions 286120 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286120 | pabelanger | 2010-09-10 17:11:08 -0400 + (Fri, 10 Sep 2010) | 18 lines Merged revisions 286117 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286117 | pabelanger | 2010-09-10 16:55:06 -0400 + (Fri, 10 Sep 2010) | 11 lines Merged revisions 286114 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286114 | pabelanger | 2010-09-10 16:35:08 -0400 (Fri, 10 Sep + 2010) | 4 lines Load iax.conf before registering any + functions/applications/actions. Review: + https://reviewboard.asterisk.org/r/914/ ........ ................ + ................ + +2010-09-10 21:03 +0000 [r286119] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 286118 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r286118 | rmudgett | 2010-09-10 15:55:37 -0500 + (Fri, 10 Sep 2010) | 25 lines Merged revisions 286116 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r286116 | rmudgett | 2010-09-10 15:42:44 -0500 + (Fri, 10 Sep 2010) | 18 lines Merged revisions 286113 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r286113 | rmudgett | 2010-09-10 15:33:16 -0500 (Fri, 10 Sep 2010) + | 11 lines An outgoing call may not get hung up if a pre-connect + incoming ISDN call is disconnected. If the ISDN link a + pre-connect incoming call is using fails or is reset, the + outgoing leg may not hang up or be delayed in hanging up. + (Causes: PRI_CAUSE_NETWORK_OUT_OF_ORDER, + PRI_CAUSE_DESTINATION_OUT_OF_ORDER, and + PRI_CAUSE_NORMAL_TEMPORARY_FAILURE.) Just hang up the call if the + incoming call leg hangs up before connecting for any reason. It + makes no sense to send a BUSY or CONGESTION control frame to the + outgoing call leg under these circumstances. ........ + ................ ................ + +2010-09-10 13:20 +0000 [r285993] David Ruggles + + * doc/externalivr.txt, CHANGES: Merged revisions 285992 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285992 | diruggles | 2010-09-10 09:13:16 -0400 (Fri, 10 Sep + 2010) | 1 line Added missing documentation for ExternalIVR + feature added in January 2010 ........ + +2010-09-10 05:33 +0000 [r285932-285963] Tilghman Lesher + + * include/asterisk/select.h, /: Merged revisions 285962 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285962 | tilghman | 2010-09-10 00:32:18 -0500 + (Fri, 10 Sep 2010) | 13 lines Merged revisions 285961 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r285961 | tilghman | 2010-09-10 00:31:31 -0500 (Fri, 10 Sep 2010) + | 6 lines Another fix for Mac OS X. While trying to fix this the + "right" way, I wandered into dependency hell. Two hours later, I + backed out, and just removed the offending code. ast_inline_api + only goes one level deep and then it breaks. Ouch. ........ + ................ + + * include/asterisk/select.h, /, configure, + include/asterisk/autoconfig.h.in, configure.ac, + tests/test_poll.c: Merged revisions 285931 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285931 | tilghman | 2010-09-09 20:25:50 -0500 + (Thu, 09 Sep 2010) | 21 lines Merged revisions 285930 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285930 | tilghman | 2010-09-09 20:16:32 -0500 + (Thu, 09 Sep 2010) | 14 lines Merged revisions 285889 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) + | 7 lines Fix Mac OS X build. This also fixes a rather grievous + calculation error for the offset of ast_fdset, which was masked + on Linux and FreeBSD, because these platforms check the first 256 + FDs regardless of the bitmask setting (due to backwards + compatibility). ........ ................ ................ + +2010-09-09 22:53 +0000 [r285820] Paul Belanger + + * /, codecs/gsm/Makefile: Merged revisions 285819 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285819 | pabelanger | 2010-09-09 18:52:31 -0400 + (Thu, 09 Sep 2010) | 22 lines Merged revisions 285818 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285818 | pabelanger | 2010-09-09 18:49:19 -0400 + (Thu, 09 Sep 2010) | 15 lines Merged revisions 285817 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285817 | pabelanger | 2010-09-09 18:34:35 -0400 (Thu, 09 Sep + 2010) | 8 lines GCC 4.2.x optimizations result in improper + behavior of GSM codec (closes issue #17688) Reported by: + pprindeville Patches: asterisk-trunk-bugid11243.patch uploaded by + pprindeville (license 347) Tested by: mkeuter, pprindeville + ........ ................ ................ + +2010-09-09 20:13 +0000 [r285746] Jason Parker + + * main/channel.c, /: Merged revisions 285745 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285745 | qwell | 2010-09-09 15:11:06 -0500 + (Thu, 09 Sep 2010) | 23 lines Merged revisions 285744 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285744 | qwell | 2010-09-09 15:09:23 -0500 + (Thu, 09 Sep 2010) | 16 lines Merged revisions 285742 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285742 | qwell | 2010-09-09 15:06:31 -0500 (Thu, 09 Sep 2010) | + 9 lines Transmit silence when reading DTMF in ast_readstring. + Otherwise, you could get issues with DTMF timeouts causing + hangups. (closes issue #17370) Reported by: makoto Patches: + channel-readstring-silence-generator.patch uploaded by makoto + (license 38) ........ ................ ................ + +2010-09-09 18:53 +0000 [r285641-285712] Brett Bryant + + * main/pbx.c, /: Merged revisions 285711 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285711 | bbryant | 2010-09-09 14:51:52 -0400 + (Thu, 09 Sep 2010) | 15 lines Merged revisions 285710 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) + | 8 lines Fixes an issue with dialplan pattern matching where the + specificity for pattern ranges and pattern special characters was + inconsistent. (closes issue #16903) Reported by: Nick_Lewis + Patches: pbx.c-specificity.patch uploaded by Nick Lewis (license + 657) Tested by: Nick_Lewis ........ ................ + + * /, res/res_musiconhold.c: Merged revisions 285640 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285640 | bbryant | 2010-09-09 13:23:28 -0400 + (Thu, 09 Sep 2010) | 21 lines Merged revisions 285639 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285639 | bbryant | 2010-09-09 13:22:25 -0400 + (Thu, 09 Sep 2010) | 14 lines Merged revisions 285638 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285638 | bbryant | 2010-09-09 13:20:17 -0400 (Thu, 09 Sep 2010) + | 7 lines Fixes an issue with MOH where it doesn't recover + cleanly when it can't play a file and would just stop, instead of + continuing to find the next playable file in the MOH class. + (closes issue #17807) Reported by: kshumard Review: + https://reviewboard.asterisk.org/r/910/ ........ ................ + ................ + +2010-09-08 22:15 +0000 [r285565-285569] David Vossel + + * /, channels/chan_sip.c: Merged revisions 285568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285568 | dvossel | 2010-09-08 17:14:19 -0500 + (Wed, 08 Sep 2010) | 16 lines Merged revisions 285567 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285567 | dvossel | 2010-09-08 17:11:28 -0500 + (Wed, 08 Sep 2010) | 9 lines Merged revisions 285566 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 + Sep 2010) | 2 lines In retrans_pkt, do not unlock pvt until the + end of the function on a transmit failure. ........ + ................ ................ + + * /, channels/chan_sip.c: Merged revisions 285564 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285564 | dvossel | 2010-09-08 16:48:37 -0500 + (Wed, 08 Sep 2010) | 60 lines Merged revisions 285563 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) + | 54 lines Fixes interoperability problems with session timer + behavior in Asterisk. CHANGES: 1. Never put "timer" in "Require" + header. This is not to our benefit and RFC 4028 section 7.1 even + warns against it. It is possible for one endpoint to perform + session-timer refreshes while the other endpoint does not support + them. If in this case the end point performing the refreshing + puts "timer" in the Require field during a refresh, the dialog + will likely get terminated by the other end. 2. Change the + behavior of 'session-timer=accept' in sip.conf (which is the + default behavior of Asterisk with no session timer configuration + specified) to only run session-timers as result of an incoming + INVITE request if the INVITE contains an "Session-Expires" + header... Asterisk is currently treating having the "timer" + option in the "Supported" header as a request for session timers + by the UAC. I do not agree with this. Session timers should only + be negotiated in "accept" mode when the incoming INVITE supplies + a "Session-Expires" header, otherwise RFC 4028 says we should + treat a request containing no "Session-Expires" header as a + session with no expiration. Below I have outlined some situations + and what Asterisk's behavior is. The table reflects the behavior + changes implemented by this patch. SITUATIONS: -Asterisk as UAS + 1. Incoming INVITE: NO "Session-Expires" 2. Incoming INVITE: HAS + "Session-Expires" -Asterisk as UAC 3. Outgoing INVITE: NO + "Session-Expires". 200 Ok Response HAS "Session-Expires" header + 4. Outgoing INVITE: NO "Session-Expires". 200 Ok Response NO + "Session-Expires" header 5. Outgoing INVITE: HAS + "Session-Expires". Active - Asterisk will have an active refresh + timer regardless if the other endpoint does. Inactive - Asterisk + does not have an active refresh timer regardless if the other + endpoint does. XXXXXXX - Not possible for mode. + ______________________________________ |SITUATIONS | + 'session-timer' MODES | |___________|________________________| | + | originate | accept | |-----------|------------|-----------| |1. + | Active | Inactive | |2. | Active | Active | |3. | XXXXXXXX | + Active | |4. | XXXXXXXX | Inactive | |5. | Active | XXXXXXXX | + -------------------------------------- (closes issue #17005) + Reported by: alexrecarey ........ ................ + +2010-09-08 21:00 +0000 [r285534] Brett Bryant + + * /, apps/app_meetme.c: Merged revisions 285533 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285533 | bbryant | 2010-09-08 16:58:43 -0400 + (Wed, 08 Sep 2010) | 15 lines Merged revisions 285532 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) + | 8 lines Fixes a bug with MeetMe where after announcing the + amount of time left in a conference, if music on hold was + playing, it doesn't restart. (closes issue #17408) Reported by: + sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by + sysreq (license 1009) Tested by: sysreq ........ ................ + +2010-09-08 20:43 +0000 [r285528-285531] Jason Parker + + * /, include/asterisk/astobj2.h, res/res_musiconhold.c: Merged + revisions 285530 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285530 | qwell | 2010-09-08 15:43:10 -0500 + (Wed, 08 Sep 2010) | 9 lines Merged revisions 285529 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r285529 | qwell | 2010-09-08 15:42:44 -0500 (Wed, 08 Sep + 2010) | 1 line Follow coding guidelines in moh rescan fix. Also + fix the documentation that got me in trouble. ........ + ................ + + * /, res/res_musiconhold.c: Merged revisions 285527 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285527 | qwell | 2010-09-08 15:32:13 -0500 + (Wed, 08 Sep 2010) | 15 lines Merged revisions 285526 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r285526 | qwell | 2010-09-08 15:31:43 -0500 (Wed, 08 Sep 2010) | + 8 lines Fixes issue where moh files were no longer rescanned + during a reload. (closes issue #16744) Reported by: pj Patches: + 16744-reload.diff uploaded by qwell (license 4) Tested by: qwell + ........ ................ + +2010-09-08 07:15 +0000 [r285485] Tilghman Lesher + + * /, funcs/func_channel.c: Merged revisions 285484 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r285484 | tilghman | 2010-09-08 02:14:17 -0500 (Wed, 08 + Sep 2010) | 2 lines Documentation only ........ + +2010-09-07 22:23 +0000 [r285394-285456] Jason Parker + + * /, channels/chan_sip.c: Merged revisions 285455 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285455 | qwell | 2010-09-07 17:22:14 -0500 (Tue, 07 Sep 2010) | + 8 lines Don't automatically add domains for wildcard bindaddrs. + (closes issue #17832) Reported by: oej Patches: + 17832-wildcard.diff uploaded by qwell (license 4) Tested by: + qwell ........ + + * /, channels/chan_sip.c: Merged revisions 285369 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285369 | qwell | 2010-09-07 15:58:34 -0500 (Tue, 07 Sep 2010) | + 7 lines Add note to 'sip show settings' regarding dual-stack + support, and a :: bindaddress. (closes issue #17831) Reported by: + oej Patches: 17831-v6wildcardbind.diff uploaded by qwell (license + 4) ........ + +2010-09-07 21:21 +0000 [r285374-285390] Tilghman Lesher + + * pbx/pbx_spool.c, /: Merged revisions 285386 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285386 | tilghman | 2010-09-07 16:20:16 -0500 (Tue, 07 Sep 2010) + | 13 lines Don't notify on attribute changes, and change how the + queuing mechanism works. Fixes call spools in 1.8. (closes issue + #17337) Reported by: loloski Patches: + 20100827__issue17337.diff.txt uploaded by tilghman (license 14) + (closes issue #17924) Reported by: mkeuter Tested by: mkeuter + ........ + + * /, funcs/func_channel.c: Merged revisions 285373 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r285373 | tilghman | 2010-09-07 16:14:03 -0500 (Tue, 07 + Sep 2010) | 7 lines Add CHANNEL(checkhangup) to check whether a + channel is in the process of being hanged up. (closes issue + #17652) Reported by: kobaz Patches: func_channel.patch uploaded + by kobaz (license 834) ........ + +2010-09-07 21:12 +0000 [r285372] Richard Mudgett + + * /, main/features.c: Merged revisions 285371 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285371 | rmudgett | 2010-09-07 16:08:35 -0500 (Tue, 07 Sep 2010) + | 1 line Fix cut-n-paste error. ........ + +2010-09-07 20:56 +0000 [r285269-285368] Tilghman Lesher + + * /, pbx/pbx_config.c: Merged revisions 285367 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285367 | tilghman | 2010-09-07 15:56:07 -0500 + (Tue, 07 Sep 2010) | 23 lines Merged revisions 285366 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285366 | tilghman | 2010-09-07 15:31:41 -0500 + (Tue, 07 Sep 2010) | 16 lines Merged revisions 285365 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285365 | tilghman | 2010-09-07 15:30:22 -0500 (Tue, 07 Sep 2010) + | 9 lines Catch invalid extensions at the parser, instead of + making the core deal with them. (closes issue #17794) Reported + by: PavelL Patches: 20100820__issue17794__1.6.2.diff.txt uploaded + by tilghman (license 14) 20100820__issue17794__1.4.diff.txt + uploaded by tilghman (license 14) Tested by: PavelL ........ + ................ ................ + + * /, include/asterisk/compiler.h, addons/ooh323c/src/ooSocket.h: + Merged revisions 285336 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285336 | tilghman | 2010-09-07 14:38:12 -0500 (Tue, 07 Sep 2010) + | 2 lines Fix build on FreeBSD 8.0, take 2. ........ + + * /, main/poll.c: Merged revisions 285268 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285268 | tilghman | 2010-09-07 14:08:09 -0500 + (Tue, 07 Sep 2010) | 18 lines Merged revisions 285267 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285267 | tilghman | 2010-09-07 14:07:17 -0500 + (Tue, 07 Sep 2010) | 11 lines Merged revisions 285266 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285266 | tilghman | 2010-09-07 14:04:50 -0500 (Tue, 07 Sep 2010) + | 4 lines Use poll, if indicated to do so, in the ast_poll2 + implementation. This fixes the unit tests on FreeBSD 8.0. + ........ ................ ................ + +2010-09-07 17:57 +0000 [r285199] Brett Bryant + + * /, apps/app_voicemail.c: Merged revisions 285197 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285197 | bbryant | 2010-09-07 13:54:21 -0400 + (Tue, 07 Sep 2010) | 24 lines Merged revisions 285196 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285196 | bbryant | 2010-09-07 13:49:07 -0400 + (Tue, 07 Sep 2010) | 17 lines Merged revisions 285194 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r285194 | bbryant | 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) + | 10 lines Fixes voicemail.conf issues where mailboxes with + passwords that don't precede a comma would throw unnecessary + error messages. (closes issue #15726) Reported by: 298 Patches: + M15726.diff uploaded by junky (license 177) Tested by: junky + Review: [full review board URL with trailing slash] ........ + ................ ................ + +2010-09-07 17:55 +0000 [r285198] Richard Mudgett + + * channels/chan_misdn.c, /: Merged revisions 285195 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285195 | rmudgett | 2010-09-07 12:47:34 -0500 + (Tue, 07 Sep 2010) | 20 lines Merged revisions 285193 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier + ........ Merged revisions 285192 via svnmerge from + https://origsvn.digium.com/svn/asterisk/be/branches/C.3 ........ + r285192 | rmudgett | 2010-09-07 11:58:57 -0500 (Tue, 07 Sep 2010) + | 8 lines COLP/CONP and chan_misdn missing update chan_misdn does + not update the caller id of the channel if a new connected number + or ECT-INFORM (w/ new peer number on call transfer) is received. + JIRA ABE-2502 JIRA SWP-2058 ........ ........ ................ + +2010-09-06 20:10 +0000 [r285163] Russell Bryant + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac: + Merged revisions 285161-285162 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285161 | russell | 2010-09-06 15:10:03 -0500 (Mon, 06 Sep 2010) + | 4 lines Fix libsrtp -fPIC check for when non-standard prefix is + used. Thanks to loompek in #asterisk for reporting the issue and + testing this patch. ........ r285162 | russell | 2010-09-06 + 15:10:24 -0500 (Mon, 06 Sep 2010) | 1 line regenerate configure + script. ........ + +2010-09-06 06:57 +0000 [r285091] Tilghman Lesher + + * /, BSDmakefile (added), makeopts.in: Merged revisions 285090 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r285090 | tilghman | 2010-09-06 01:56:07 -0500 + (Mon, 06 Sep 2010) | 16 lines Merged revisions 285089 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r285089 | tilghman | 2010-09-06 01:55:17 -0500 + (Mon, 06 Sep 2010) | 9 lines Merged revisions 285088 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r285088 | tilghman | 2010-09-06 01:54:18 -0500 (Mon, 06 + Sep 2010) | 2 lines Silly convenience script for BSD platforms. + ........ ................ ................ + +2010-09-04 18:10 +0000 [r285058] Russell Bryant + + * /, include/asterisk/cli.h: Merged revisions 285057 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r285057 | russell | 2010-09-04 13:08:19 -0500 (Sat, 04 + Sep 2010) | 2 lines Add a C++ compatible version of + AST_CLI_DEFINE(). ........ + +2010-09-03 23:23 +0000 [r285029] Terry Wilson + + * /, channels/chan_sip.c: Merged revisions 285017 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r285017 | twilson | 2010-09-03 18:19:54 -0500 (Fri, 03 Sep 2010) + | 4 lines Call correct lock function as transferer is a sip_pvt + not a channel Both functions are #defined to ao2_lock, but + still... ........ + +2010-09-03 22:23 +0000 [r285007] David Vossel + + * /, channels/chan_sip.c, configs/sip.conf.sample, + channels/sip/include/sip.h: Merged revisions 285006 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r285006 | dvossel | 2010-09-03 17:21:50 -0500 (Fri, 03 + Sep 2010) | 9 lines Disables auth_options_request option by + default. The auth_options_request option was created to do + authentication on OPTIONS request just like INVITES are done. + Since it has been noted that some endpoints use OPTIONS requests + as a way of qualifying a peer and that a 401 authentication + response could result in interoperability issues, this option has + been disabled by default. ........ + +2010-09-03 18:21 +0000 [r284973] Brett Bryant + + * /, channels/chan_iax2.c: Merged revisions 284967 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284967 | bbryant | 2010-09-03 14:19:53 -0400 + (Fri, 03 Sep 2010) | 15 lines Merged revisions 284958 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r284958 | bbryant | 2010-09-03 14:15:49 -0400 (Fri, 03 Sep 2010) + | 8 lines This is a patch provided for issue #17935 to add the + ActionID to the IAXregistry AMI response. (closes issue #17935) + Reported by: alexkuklin Patches: iaxshowreg uploaded by + alexkuklin (license 1115) Tested by: alexkuklin ........ + ................ + +2010-09-03 18:04 +0000 [r284951-284953] David Vossel + + * /, channels/chan_sip.c: Merged revisions 284952 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284952 | dvossel | 2010-09-03 13:03:23 -0500 (Fri, 03 Sep 2010) + | 2 lines During OPTIONS authentication, the authpeer does not + need to be returned for any reason. ........ + + * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + channels/sip/include/sip.h: Merged revisions 284950 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r284950 | dvossel | 2010-09-03 12:29:02 -0500 (Fri, 03 + Sep 2010) | 14 lines authenticate OPTIONS requests just like we + would an INVITE OPTIONS requests should be treated the same as an + INVITE This includes authentication. This patch adds the ability + for incoming out of dialog OPTION requests to be authenticated + before providing a response indicating whether an extension is + available or not. The authentication routine works the exact same + way as it does for incoming INVITEs. This means that if a peer + has 'insecure=invite' in their peer definition, the same will be + true for the processing of the OPTIONS request. Review: + https://reviewboard.asterisk.org/r/881/ ........ + +2010-09-03 16:42 +0000 [r284922] Terry Wilson + + * /, apps/app_chanspy.c: Merged revisions 284921 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284921 | twilson | 2010-09-03 11:28:18 -0500 + (Fri, 03 Sep 2010) | 19 lines Merged revisions 284897 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284897 | twilson | 2010-09-03 11:20:45 -0500 + (Fri, 03 Sep 2010) | 12 lines Merged revisions 284881 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r284881 | twilson | 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) + | 5 lines Properly detect when a sound file doesn't exist + ast_fileexists returns -1 for error and 0 for a non-existant + file. The existing code treated missing files as though they + existed. ........ ................ ................ + +2010-09-03 13:09 +0000 [r284851-284853] Jan Kalab + + * /, res/res_calendar_ews.c: Merge of strdupa() fix for calendars + categories priorities + + * res/res_calendar_icalendar.c, /, res/res_calendar_caldav.c, + include/asterisk/calendar.h, res/res_calendar_ews.c, + res/res_calendar.c: Support for calendar events priorities and + categories (with ISO C90 fix) See RFC 5545 ch. 3.8.1.2 and 9. + (closes issue #17837) Review: + https://reviewboard.asterisk.org/r/880/ + +2010-09-02 21:08 +0000 [r284782] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /, channels/sig_pri.c: + Merged revisions 284779-284780 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284779 | rmudgett | 2010-09-02 15:59:12 -0500 (Thu, 02 Sep 2010) + | 8 lines Made output libpri event names if pri debugging is + enabled when sig_pri processes them. * Simplified CLI "pri debug + xx span xx" command code and removed redundant debugging enabled + messages. * Made CLI "pri debug xx span xx" command only close + the debugging log file if it was opened. ........ r284780 | + rmudgett | 2010-09-02 16:02:54 -0500 (Thu, 02 Sep 2010) | 2 lines + Simplified pri_dchannel() poll timeout duration code. ........ + +2010-09-02 16:57 +0000 [r284706] David Vossel + + * /, channels/chan_sip.c: Merged revisions 284705 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284705 | dvossel | 2010-09-02 11:56:43 -0500 + (Thu, 02 Sep 2010) | 20 lines Merged revisions 284704 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284704 | dvossel | 2010-09-02 11:48:51 -0500 + (Thu, 02 Sep 2010) | 13 lines Merged revisions 284703 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r284703 | dvossel | 2010-09-02 11:47:15 -0500 (Thu, 02 Sep 2010) + | 7 lines Removed relatedpeer code from sip_autodestruct Handling + of the relatedpeer structure associated with a sip_pvt should be + done during the final sip_destruction function, not in + sip_autodestruct. ........ ................ ................ + +2010-09-02 16:44 +0000 [r284702] Jason Parker + + * /, formats/format_wav.c: Merged revisions 284701 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r284701 | qwell | 2010-09-02 11:43:09 -0500 (Thu, 02 Sep + 2010) | 8 lines Add slin16 support for format_wav (new wav16 file + extension) (closes issue #15029) Reported by: andrew Patches: + wav16.patch uploaded by andrew (license 240) Tested by: qwell, + andrew ........ + +2010-09-02 16:36 +0000 [r284700] Tilghman Lesher + + * /, addons/ooh323c/src/oochannels.c: Merged revisions 284696 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284696 | tilghman | 2010-09-02 11:27:52 -0500 (Thu, 02 Sep 2010) + | 2 lines Fixing build ........ + +2010-09-02 16:35 +0000 [r284699] Richard Mudgett + + * /, doc/tex/channelvariables.tex, doc/tex/partymanip.tex (added), + doc/tex/asterisk.tex: Merged revisions 284698 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284698 | rmudgett | 2010-09-02 11:34:32 -0500 (Thu, 02 Sep 2010) + | 5 lines Added documentation for CONNECTEDLINE and REDIRECTING + functions. (closes issue #17808) Reported by: jtodd Review: + https://reviewboard.asterisk.org/r/875/ ........ + +2010-09-02 16:12 +0000 [r284598-284667] Tilghman Lesher + + * channels/chan_usbradio.c, /: Merged revisions 284666 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284666 | tilghman | 2010-09-02 11:11:15 -0500 + (Thu, 02 Sep 2010) | 9 lines Merged revisions 284665 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r284665 | tilghman | 2010-09-02 11:07:19 -0500 (Thu, 02 + Sep 2010) | 2 lines Fixing build. ........ ................ + + * /, apps/app_queue.c: Merged revisions 284632 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284632 | tilghman | 2010-09-02 00:31:02 -0500 + (Thu, 02 Sep 2010) | 14 lines Merged revisions 284631 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r284631 | tilghman | 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010) + | 7 lines Don't reset queue stats on a module reload. (closes + issue #17535) Reported by: raarts Patches: + 20100819__issue17535.diff.txt uploaded by tilghman (license 14) + ........ ................ + + * /, channels/chan_sip.c, channels/chan_agent.c, + channels/chan_iax2.c, apps/app_queue.c, apps/app_getcpeid.c, + apps/app_followme.c, apps/app_speech_utils.c, main/loader.c, + pbx/pbx_loopback.c, channels/chan_dahdi.c, funcs/func_aes.c, + include/asterisk/module.h, pbx/pbx_realtime.c, pbx/pbx_dundi.c, + apps/app_stack.c, channels/chan_mgcp.c, apps/app_adsiprog.c, + apps/app_voicemail.c: Merged revisions 284610 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) + | 10 lines When optional_api is non-optional, force dependent + modules to be loaded. (closes issue #17707) Reported by: ira + Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by + tilghman (license 14) Tested by: tilghman Review: + https://reviewboard.asterisk.org/r/876/ ........ + + * main/stun.c, res/res_ais.c, /, include/asterisk/autoconfig.h.in, + configure.ac, channels/console_video.c, + include/asterisk/channel.h, res/res_jabber.c, res/res_pktccops.c, + main/poll.c, channels/chan_usbradio.c, include/asterisk/select.h + (added), channels/chan_phone.c, channels/chan_misdn.c, configure, + main/features.c, include/asterisk/poll-compat.h, + tests/test_poll.c (added), addons/ooh323c/src/oochannels.c, + main/asterisk.c, addons/ooh323c/src/ooSocket.h: Merged revisions + 284597 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284597 | tilghman | 2010-09-02 00:00:34 -0500 + (Thu, 02 Sep 2010) | 29 lines Merged revisions 284593,284595 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284593 | tilghman | 2010-09-01 17:59:50 -0500 + (Wed, 01 Sep 2010) | 18 lines Merged revisions 284478 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) + | 11 lines Ensure that all areas that previously used select(2) + now use poll(2), with implementations that need poll(2) + implemented with select(2) safe against 1024-bit overflows. This + is a followup to the fix for the pthread timer in 1.6.2 and + beyond, fixing a potential crash bug in all supported releases. + (closes issue #17678) Reported by: russell Branch: + https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select + Review: https://reviewboard.asterisk.org/r/824/ ........ + ................ r284595 | tilghman | 2010-09-01 22:57:43 -0500 + (Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after + last commit ................ ................ + +2010-09-01 21:48 +0000 [r284562] David Vossel + + * /, channels/chan_sip.c: Merged revisions 284561 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284561 | dvossel | 2010-09-01 16:47:01 -0500 (Wed, 01 Sep 2010) + | 9 lines During request to dialog matching, verify init_ruri is + present before comparing. During request to dialog matching, we + attempt a best effort routine for fork detection which requires + several elements to be in place. The dialog's initial request uri + is one of those elements. Since it is best effort, if the + init_ruri is not present for some reason we can not proceed with + that routine. ........ + +2010-09-01 18:52 +0000 [r284479] Terry Wilson + + * res/res_rtp_asterisk.c, include/asterisk/res_srtp.h, + main/rtp_engine.c, /, channels/chan_sip.c, res/res_srtp.c: Merged + revisions 284477 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284477 | twilson | 2010-09-01 13:44:36 -0500 (Wed, 01 Sep 2010) + | 17 lines Fix SRTP for changing SSRC and multiple a=crypto SDP + lines Adding code to Asterisk that changed the SSRC during + bridges and masquerades broke SRTP functionality. Also broken was + handling the situation where an incoming INVITE had more than one + crypto offer. This patch caches the SRTP policies the we use so + that we can change the ssrc and inform libsrtp of the new + streams. It also uses the first acceptable a=crypto line from the + incoming INVITE. (closes issue #17563) Reported by: Alexcr + Patches: srtp.diff uploaded by twilson (license 396) Tested by: + twilson Review: https://reviewboard.asterisk.org/r/878/ ........ + +2010-09-01 18:19 +0000 [r284440-284474] Tilghman Lesher + + * res/res_config_pgsql.c, /: Merged revisions 284473 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284473 | tilghman | 2010-09-01 13:16:37 -0500 + (Wed, 01 Sep 2010) | 12 lines Merged revisions 284472 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r284472 | tilghman | 2010-09-01 13:13:35 -0500 (Wed, 01 Sep 2010) + | 5 lines Don't warn on floats and timestamps (closes issue + #17082) Reported by: coolmig ........ ................ + + * /, channels/chan_sip.c: Merged revisions 284415 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284415 | tilghman | 2010-08-31 15:22:10 -0500 + (Tue, 31 Aug 2010) | 21 lines Merged revisions 284399 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284399 | tilghman | 2010-08-31 15:18:32 -0500 + (Tue, 31 Aug 2010) | 14 lines Merged revisions 284393 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r284393 | tilghman | 2010-08-31 15:13:21 -0500 (Tue, 31 Aug 2010) + | 7 lines Don't send a devstate change on poke_noanswer if the + state did not change. (closes issue #17741) Reported by: schmidts + Patches: chan_sip.c.patch uploaded by schmidts (license 1077) + ........ ................ ................ + +2010-08-31 19:01 +0000 [r284315-284319] Leif Madsen + + * /, configs/say.conf.sample: Merged revisions 284318 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284318 | lmadsen | 2010-08-31 14:00:15 -0500 + (Tue, 31 Aug 2010) | 22 lines Merged revisions 284317 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284317 | lmadsen | 2010-08-31 13:59:31 -0500 + (Tue, 31 Aug 2010) | 15 lines Merged revisions 284316 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r284316 | lmadsen | 2010-08-31 13:57:59 -0500 (Tue, 31 Aug 2010) + | 7 lines Update say.conf.sample to match the rules in say.c + (closes issue #17835) Reported by: RoadKill Patches: + say.conf.sample.patch.rules uploaded by RoadKill (license 933) + Tested by: RoadKill ........ ................ ................ + + * channels/chan_sip.c: Add trustrpid and sendrpid global values to + 'sip show settings' (closes issue #17860) Reported by: jtodd + Patches: __20100816-chan_sip-sip-show-settings.txt uploaded by + lmadsen (license 10) Tested by: lmadsen, russell + +2010-08-30 22:30 +0000 [r284282] Tilghman Lesher + + * /, apps/app_festival.c: Merged revisions 284281 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284281 | tilghman | 2010-08-30 17:28:47 -0500 + (Mon, 30 Aug 2010) | 18 lines Merged revisions 284280 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r284280 | tilghman | 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010) + | 11 lines Fix 3 coding errors: 1) After we close FD, we should + not be trying to write to it. 2) Call _exit(0), not exit(0), to + avoid running shutdown routines in a child. 3) Use endian, not + processor, detection to ensure bytes are written in the correct + order. (closes issue #15706) Reported by: modelnine Patches: + asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine + (license 865) Tested by: gmartinez ........ ................ + +2010-08-30 09:32 +0000 [r284189-284248] Olle Johansson + + * main/file.c: Small doxygen fix and doc addition + + * main/say.c: Clean upp doxygen documentation + + * include/asterisk/say.h: Doxygen formatting You can't write "same + as above" in hypertext documentation. Above doesn't make sense in + hyperspace. + + * apps/app_playback.c: Add doxygen documentation + +2010-08-29 07:06 +0000 [r284097-284159] Tilghman Lesher + + * /, configs/res_curl.conf.sample (added): Merged revisions 284158 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284158 | tilghman | 2010-08-29 02:05:27 -0500 (Sun, 29 Aug 2010) + | 2 lines Missed adding this file ........ + + * /, sounds: Merged revisions 284127 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284127 | tilghman | 2010-08-29 00:17:37 -0500 (Sun, 29 Aug 2010) + | 2 lines Also ignore the checksums ........ + + * cel/cel_odbc.c (added), /, configs/cel_adaptive_odbc.conf.sample + (removed), configs/cel_odbc.conf.sample (added), + cel/cel_adaptive_odbc.c (removed): Merged revisions 284096 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284096 | tilghman | 2010-08-28 21:51:14 -0500 (Sat, 28 Aug 2010) + | 3 lines Rename CEL adaptive driver to plain driver, since there + isn't another ODBC driver (and the other CEL drivers have + adaptive capabilities, anyway). ........ + +2010-08-28 21:30 +0000 [r284066] Russell Bryant + + * main/manager.c, /: Merged revisions 284065 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r284065 | russell | 2010-08-28 16:29:45 -0500 (Sat, 28 Aug 2010) + | 13 lines Be more flexible with whitespace on AMI action + headers. Previously, this code required exactly one space to be + after the ':' in headers for an AMI action. This now makes + whitespace optional, and allows whitespace that is there to vary + in amount. (closes issue #17862) Reported by: cmoye Patches: + manager.c.patch_trunk uploaded by cmoye (license 858) + manager.c.patch_1.8 uploaded by cmoye (license 858) Tested by: + cmoye ........ + +2010-08-27 22:39 +0000 [r284033] David Vossel + + * /, channels/chan_sip.c: Merged revisions 284032 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r284032 | dvossel | 2010-08-27 17:37:11 -0500 + (Fri, 27 Aug 2010) | 21 lines Merged revisions 284002 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r284002 | dvossel | 2010-08-27 17:27:50 -0500 + (Fri, 27 Aug 2010) | 14 lines Merged revisions 283960 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r283960 | dvossel | 2010-08-27 17:17:26 -0500 (Fri, 27 Aug 2010) + | 8 lines Parse all "Accept" headers for SIP SUBSCRIBE requests. + (closes issue #17758) Reported by: ibc Patches: + multiple_accept_headers_1.4.diff uploaded by dvossel (license + 671) ........ ................ ................ + +2010-08-27 21:50 +0000 [r283958] Russell Bryant + + * /, pbx/pbx_realtime.c: Merged revisions 283951 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283951 | russell | 2010-08-27 16:33:55 -0500 (Fri, 27 Aug 2010) + | 2 lines Print exten@context:priority in verbose messages from + pbx_realtime. ........ + +2010-08-27 20:32 +0000 [r283883] Jason Parker + + * res/res_config_odbc.c, /, main/config.c, + addons/res_config_mysql.c: Merged revisions 283882 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283882 | qwell | 2010-08-27 15:31:55 -0500 + (Fri, 27 Aug 2010) | 22 lines Merged revisions 283881 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r283881 | qwell | 2010-08-27 15:30:27 -0500 + (Fri, 27 Aug 2010) | 15 lines Merged revisions 283880 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r283880 | qwell | 2010-08-27 15:29:11 -0500 (Fri, 27 Aug 2010) | + 8 lines Fix issue with decoding ^-escaped characters in realtime. + (closes issue #17790) Reported by: denzs Patches: + 17790-chunky.diff uploaded by qwell (license 4) Tested by: qwell, + denzs ........ ................ ................ + +2010-08-27 14:01 +0000 [r283803] Olle Johansson + + * main/manager.c: Doxygen formatting changes + +2010-08-26 23:51 +0000 [r283771] Tilghman Lesher + + * /, res/res_musiconhold.c: Merged revisions 283770 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r283770 | tilghman | 2010-08-26 18:47:02 -0500 (Thu, 26 + Aug 2010) | 8 lines Convert MOH to use generic timers. (closes + issue #17726) Reported by: lmadsen Patches: + 20100825__issue17726__2.diff.txt uploaded by tilghman (license + 14) Tested by: tilghman ........ + +2010-08-26 15:28 +0000 [r283693] David Vossel + + * /, channels/chan_sip.c: Merged revisions 283692 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283692 | dvossel | 2010-08-26 10:26:37 -0500 + (Thu, 26 Aug 2010) | 32 lines Merged revisions 283691 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r283691 | dvossel | 2010-08-26 10:24:40 -0500 + (Thu, 26 Aug 2010) | 25 lines Merged revisions 283690 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r283690 | dvossel | 2010-08-26 10:22:28 -0500 (Thu, 26 Aug 2010) + | 19 lines Fixed how Asterisk destroys a dialog on channel hangup + before invite receives a response. If an ast_channel with a SIP + tech pvt hangs up before the sip dialog gets a response to its + outgoing INVITE, Asterisk used to pretend_ack the INVITE. This is + not rfc compliant and results in confusion at the other endpoint. + sip_pretend_ack will ack and remove all the packets in the + retransmit queue. This means that the INVITE will stop + retransmitting, and that any response to that INVITE that comes + after the pretend_ack occurs will be ignored. Instead of faking + any sort of acknowledgement for an outgoing INVITE during an + internal hangup, we should let the protocol stack process the + INVITE transaction and terminate the dialog properly. This is + achieved by setting the PENDING_BYE flag. When this flag is used, + once the dialog proceeds to an escapable state the transaction + will either be canceled with a SIP_CANCEL or completed followed + immediately by a BYE. Attempting to do this any other way is + incorrect. If the endpoint is not responding to the INVITE + request, the INVITE must continue to be retransmitted until it + times out which will result in the dialog being destroyed. + ........ ................ ................ + +2010-08-26 13:28 +0000 [r283628-283660] Russell Bryant + + * /, res/res_odbc.c: Merged revisions 283659 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283659 | russell | 2010-08-26 08:26:14 -0500 (Thu, 26 Aug 2010) + | 2 lines Slight improvement to a debug message. ........ + + * Makefile, /, keys/iaxtel.pub (removed), keys/freeworlddialup.pub + (removed): Merged revisions 283629 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283629 | russell | 2010-08-26 07:48:45 -0500 (Thu, 26 Aug 2010) + | 2 lines Remove public keys that are no longer useful. ........ + + * /, configs/manager.conf.sample: Merged revisions 283627 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283627 | russell | 2010-08-26 07:26:22 -0500 (Thu, 26 Aug 2010) + | 2 lines Move httptimeout out from in between port and bindaddr. + ........ + +2010-08-25 22:59 +0000 [r283596] David Vossel + + * /, channels/chan_sip.c: Merged revisions 283595 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283595 | dvossel | 2010-08-25 17:57:56 -0500 + (Wed, 25 Aug 2010) | 14 lines Merged revisions 283594 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r283594 | dvossel | 2010-08-25 17:56:42 -0500 (Wed, 25 Aug 2010) + | 7 lines Add to and from tags to NOTIFY dialog-info xml body so + pickup can occur. When pedantic mode is used, the dialog-info xml + generated during a ringing event must contain the to and from tag + values. Otherwise if a pickup occurs using INVITE with replaces, + Astrisk will not be able to locate the subscription. ........ + ................ + +2010-08-25 16:14 +0000 [r283562] Tilghman Lesher + + * /, res/res_odbc.c: Merged revisions 283561 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283561 | tilghman | 2010-08-25 11:12:43 -0500 (Wed, 25 Aug 2010) + | 5 lines Initialize connect timeout on each time through the + loop. (closes issue #17911) Reported by: wurstsalat ........ + +2010-08-25 15:56 +0000 [r283560] David Vossel + + * /, channels/chan_sip.c, channels/sip/include/sip.h: Merged + revisions 283559 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283559 | dvossel | 2010-08-25 10:54:11 -0500 + (Wed, 25 Aug 2010) | 16 lines Merged revisions 283558 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r283558 | dvossel | 2010-08-25 10:52:54 -0500 (Wed, 25 Aug 2010) + | 10 lines Asterisk will not advertise session timers are + supported when 'session-timers=refuse' is used. Asterisk now + dynamically builds the "Supported" header depending on what is + enabled/disabled in sip.conf. Session timers used to always be + advertised as being supported even when they were disabled in the + configuration. This caused problems with some end points. (issue + #17005) ........ ................ + +2010-08-25 14:55 +0000 [r283528] Russell Bryant + + * /, channels/chan_sip.c: Merged revisions 283527 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283527 | russell | 2010-08-25 09:55:00 -0500 (Wed, 25 Aug 2010) + | 2 lines Convert ast_log(LOG_DEBUG, ...) to ast_debug(...) + ........ + +2010-08-24 20:42 +0000 [r283495] Damien Wedhorn + + * channels/chan_skinny.c: Ignore redial hard button when no + previous number. (closes issue #17887) Reported by: salecha + Patches: skinny.redial.diff uploaded by wedhorn (license 30) + Tested by: wedhorn, salecha + +2010-08-24 20:36 +0000 [r283494] David Vossel + + * /, configs/sip.conf.sample, UPGRADE-1.8.txt, + channels/sip/include/sip.h: Merged revisions 283493 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24 + Aug 2010) | 2 lines Changes the default behavior for sip.conf's + pedantic option from "no" to "yes". ........ + +2010-08-24 18:58 +0000 [r283458] Leif Madsen + + * res/res_rtp_asterisk.c, /, channels/chan_sip.c: Merged revisions + 283457 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283457 | lmadsen | 2010-08-24 13:56:29 -0500 (Tue, 24 Aug 2010) + | 9 lines Fix issue where TOS is no longer set on RTP packets. + Fix issue where the tos is no longer being set on RTP packets + through res_rtp_asterisk. (closes issue #17890) Reported by: + elguero Patches: qos_18.diff uploaded by elguero (license 37) + Review: https://reviewboard.asterisk.org/r/868 ........ + +2010-08-24 18:45 +0000 [r283383-283456] David Vossel + + * res/res_stun_monitor.c: This fix downgrades the ERROR message + indicating no res_stun_monitor.conf to a WARNING message. + + * /, channels/chan_sip.c: Merged revisions 283382 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283382 | dvossel | 2010-08-24 11:11:18 -0500 + (Tue, 24 Aug 2010) | 25 lines Merged revisions 283381 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r283381 | dvossel | 2010-08-24 11:07:37 -0500 + (Tue, 24 Aug 2010) | 18 lines Merged revisions 283380 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r283380 | dvossel | 2010-08-24 11:01:51 -0500 (Tue, 24 Aug 2010) + | 11 lines This fix makes sure the ast_channel hangs up correctly + when the dialog's PENDING_BYE flag is set. When the pending bye + flag is used, it is possible that the dialog will terminate and + leave the sip_pvt->owner channel up. This is because we never + hangup the ast_channel after sending the SIP_BYE request. When we + receive the response for the SIP_BYE we set need_destroy which we + would expect to destroy the dialog on the next do_monitor loop, + but this is not the case. The dialog will only be destroyed once + the owner is hungup even with the need_destroy flag set. This + patch sets the softhangup flag on the ast_channel when a SIP_BYE + request is sent as a result of the pending bye flag. ........ + ................ ................ + +2010-08-24 12:51 +0000 [r283351] Russell Bryant + + * /, funcs/func_odbc.c: Merged revisions 283350 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283350 | russell | 2010-08-24 07:49:41 -0500 (Tue, 24 Aug 2010) + | 2 lines Don't attempt to release a NULL ODBC handle. ........ + +2010-08-23 21:35 +0000 [r283320] Tilghman Lesher + + * /, cdr/cdr_adaptive_odbc.c, cdr/cdr_odbc.c, + cel/cel_adaptive_odbc.c: Merged revisions 283319 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283319 | tilghman | 2010-08-23 16:33:47 -0500 + (Mon, 23 Aug 2010) | 9 lines Merged revisions 283318 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r283318 | tilghman | 2010-08-23 16:32:14 -0500 (Mon, 23 + Aug 2010) | 2 lines CDR drivers depend upon res_odbc, not + directly on the ODBC libraries ........ ................ + +2010-08-23 20:50 +0000 [r283287-283289] Damien Wedhorn + + * channels/chan_skinny.c: Hack to allow easy debugging of skinny in + trunk. + + * channels/chan_skinny.c: Add additional AST_CONTROL_ states to + control2str. + + * channels/chan_skinny.c: Fixes display issues on 7910 and older + phones. Also correct the callinfo provided in skinny_answer. + (closes issue #17876) Reported by: salecha Patches: + skinny_cnd3.diff uploaded by wedhorn (license 30) Tested by: + salecha, wedhorn Review: NA + +2010-08-23 13:35 +0000 [r283178-283242] Russell Bryant + + * configs/cel.conf.sample, /: Merged revisions 283241 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r283241 | russell | 2010-08-23 08:35:35 -0500 (Mon, 23 + Aug 2010) | 2 lines Add sample configuration for cel_radius. + ........ + + * /, include/asterisk/cel.h, main/cel.c: Merged revisions 283230 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283230 | russell | 2010-08-23 08:23:12 -0500 (Mon, 23 Aug 2010) + | 7 lines Make the AST_CEL_AMA enum match up with the AST_CDR_ + ama flag values. Really, having 2 enums for this is silly and + error prone, demonstrated by the crash that I hit because there + was an assumption in the code that the values in each matched up. + However, this is a quick fix to get them to match up so it will + work. ........ + + * /, main/cel.c: Merged revisions 283209 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283209 | russell | 2010-08-23 08:06:57 -0500 (Mon, 23 Aug 2010) + | 2 lines Don't blow up on an invalid AMA flag. ........ + + * /, configs/cel_custom.conf.sample: Merged revisions 283207 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283207 | russell | 2010-08-23 07:31:20 -0500 (Mon, 23 Aug 2010) + | 2 lines Tack on ${eventextra} to the sample cel_custom.conf. + ........ + + * /, configs/cel_custom.conf.sample: Merged revisions 283177 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283177 | russell | 2010-08-23 07:12:53 -0500 (Mon, 23 Aug 2010) + | 2 lines Cut down on excessive quotation. ........ + +2010-08-23 12:09 +0000 [r283176] Tilghman Lesher + + * /, res/res_stun_monitor.c: Merged revisions 283175 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r283175 | tilghman | 2010-08-23 07:06:26 -0500 (Mon, 23 + Aug 2010) | 2 lines Don't fail to start if the config file is + missing. ........ + +2010-08-23 11:59 +0000 [r283174] Russell Bryant + + * /, configs/cel_custom.conf.sample: Merged revisions 283173 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283173 | russell | 2010-08-23 06:58:34 -0500 (Mon, 23 Aug 2010) + | 5 lines Expand cel_custom.conf.sample. Include the usage of + CSV_QUOTE() to ensure data has valid CSV formatting. Also list + the special CEL variables that are available for use in the + mapping. ........ + +2010-08-20 15:39 +0000 [r283051] Richard Mudgett + + * /, channels/sig_pri.c: Merged revisions 283050 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r283050 | rmudgett | 2010-08-20 10:35:38 -0500 + (Fri, 20 Aug 2010) | 36 lines Merged revisions 283049 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r283049 | rmudgett | 2010-08-20 10:31:03 -0500 + (Fri, 20 Aug 2010) | 29 lines Merged revisions 283048 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r283048 | rmudgett | 2010-08-20 10:24:36 -0500 (Fri, 20 Aug 2010) + | 22 lines Q931 - Sending PROGRESS after sending ALERTING is a + protocol error The PRI layer in chan_dadhi will check if a + PROGRESS message has already been sent, and not allow sending + another (although that is technically allowed by the Q931 spec), + however it does not protect against sending an ALERTING and then + sending a PROGRESS message, which is a violation of the + specification. Most switches don't seem to care too deeply about + this, but some do, and will disconnect the call when receiving + this invalid sequence. Protocol specification reference: + T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview + protocol control (network side) point-point (sheet 3 of 8)" + (closes issue #17874) Reported by: nic_bellamy Patches: + asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by + nic bellamy (license 299) + asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded + by nic bellamy (license 299) + asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded + by nic bellamy (license 299) ........ ................ + ................ + +2010-08-20 12:45 +0000 [r282980-283015] Russell Bryant + + * /, configs/cel_adaptive_odbc.conf.sample: Merged revisions 283013 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r283013 | russell | 2010-08-20 07:45:12 -0500 (Fri, 20 Aug 2010) + | 2 lines Fix a typo in a column name. ........ + + * /, apps/app_celgenuserevent.c: Merged revisions 282979 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282979 | russell | 2010-08-20 06:52:37 -0500 (Fri, 20 Aug 2010) + | 2 lines Add an argument missing from the CELGenUserEvent + documentation. ........ + + * channels/chan_multicast_rtp.c, /: Merged revisions 282638 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282638 | russell | 2010-08-18 07:30:40 -0500 (Wed, 18 Aug 2010) + | 4 lines Split _all_ arguments before parsing them. This fixes + multicast RTP paging using linksys mode. ........ + +2010-08-19 21:08 +0000 [r282892-282896] David Vossel + + * /, channels/chan_sip.c: Merged revisions 282895 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282895 | dvossel | 2010-08-19 16:07:20 -0500 + (Thu, 19 Aug 2010) | 25 lines Merged revisions 282894 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r282894 | dvossel | 2010-08-19 16:05:54 -0500 + (Thu, 19 Aug 2010) | 18 lines Merged revisions 282893 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) + | 11 lines tos_sip option was not being set correctly When + tos_sip is used, the tos of the sip socket is only set correctly + if the socket binding changes on a reload. If the binding stays + the same but the TOS changes, the new tos value would not take + into effect. This patch fixes that. (closes issue #17712) + Reported by: nickb ........ ................ ................ + + * /, channels/chan_sip.c: Merged revisions 282891 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282891 | dvossel | 2010-08-19 15:34:41 -0500 + (Thu, 19 Aug 2010) | 11 lines Merged revisions 282890 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r282890 | dvossel | 2010-08-19 15:31:22 -0500 (Thu, 19 Aug 2010) + | 5 lines fixes sip peer memory leaks in the peer_by_ip table + (issue #17798) ........ ................ + +2010-08-19 20:02 +0000 [r282861] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 282860 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282860 | mnicholson | 2010-08-19 15:01:11 -0500 + (Thu, 19 Aug 2010) | 30 lines Merged revisions 282859 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r282859 | mnicholson | 2010-08-19 14:44:00 -0500 + (Thu, 19 Aug 2010) | 23 lines Merged revisions 277944 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r277944 | pabelanger | 2010-07-19 15:56:07 -0500 (Mon, 19 Jul + 2010) | 16 lines Regression with T.38 negotiation Prior to + 1.4.26.3 T.38 negotiation worked properly, in the case of the + reporter. (issue #16852) Reported by: cfc (closes issue #16705) + Reported by: mpiazzatnetbug Patches: issue16705_2.diff uploaded + by ebroad (license 878) Tested by: vrban, ebroad, c0rnoTa, + samdell3 Review: https://reviewboard.asterisk.org/r/754/ ........ + ................ ................ + +2010-08-19 14:46 +0000 [r282827] Tilghman Lesher + + * main/netsock2.c, /: Merged revisions 282826 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282826 | tilghman | 2010-08-19 09:44:51 -0500 (Thu, 19 Aug 2010) + | 2 lines Only output debugging if the debug level is on. + ........ + +2010-08-19 12:13 +0000 [r282798] Russell Bryant + + * include/asterisk/cel.h: Add a todo item for CEL. + +2010-08-19 02:20 +0000 [r282751] Terry Wilson + + * /, configs/sip.conf.sample: Merged revisions 282740 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282740 | twilson | 2010-08-18 21:18:50 -0500 + (Wed, 18 Aug 2010) | 16 lines Merged revisions 282730 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r282730 | twilson | 2010-08-18 21:14:28 -0500 + (Wed, 18 Aug 2010) | 9 lines Merged revisions 282729 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r282729 | twilson | 2010-08-18 21:12:55 -0500 (Wed, 18 + Aug 2010) | 2 lines Add some documentation about codec + negotiation to sip.conf ........ ................ + ................ + +2010-08-18 21:34 +0000 [r282701] Damien Wedhorn + + * channels/chan_skinny.c: Cleanup: consolidate offhook (new call). + Consolidates all offhook (new call with dialtone) to + setsubstate_offhook. This should be roughly equivalent to + existing code, although a couple of calls now run through the + full offhook sequence rather than an abbreviated one. (closes + issue #17812) Reported by: wedhorn Patches: + cleanup.stateoffhook.diff uploaded by wedhorn (license 30) Tested + by: salecha, wedhorn Review: NA + +2010-08-18 15:35 +0000 [r282673] Richard Mudgett + + * channels/sig_pri.h, channels/chan_dahdi.c, /: Merged revisions + 282671-282672 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282671 | rmudgett | 2010-08-18 10:27:51 -0500 (Wed, 18 Aug 2010) + | 1 line Use the correct operator when calculating the PRI span + devstate. ........ r282672 | rmudgett | 2010-08-18 10:28:27 -0500 + (Wed, 18 Aug 2010) | 1 line Use the correct type for + aoce_delayhangup bit field. ........ + +2010-08-18 13:11 +0000 [r282640] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 282639 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282639 | mnicholson | 2010-08-18 08:10:39 -0500 (Wed, 18 Aug + 2010) | 13 lines Properly handle 200 and unknown responses + conatined in NOTIFY requests received in response to REFER + requests. This patch fixes the way asterisk handles NOTIFY + requests received in response to REFER requests. These changes to + NOTIFY handler were first introduced in r217482. This new change + properly handles the 200 response by queueing an + AST_TRANSFER_SUCCESS control frame and also prevents that control + frame from being queued when provisional and unknown responses + are received. (issue #17486) Reported by: davidw Tested by: + mnicholson (issue #12713) Reported by: davidw Review: + https://reviewboard.asterisk.org/r/860/ ........ + +2010-08-18 07:50 +0000 [r282609] Tilghman Lesher + + * /, channels/sig_pri.c: Merged revisions 282608 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282608 | tilghman | 2010-08-18 02:49:04 -0500 + (Wed, 18 Aug 2010) | 16 lines Merged revisions 282607 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r282607 | tilghman | 2010-08-18 02:43:14 -0500 (Wed, 18 Aug 2010) + | 9 lines Don't warn on callerid when completely text, instead of + numeric with localdialplan prefixes. (closes issue #16770) + Reported by: jamicque Patches: 20100413__issue16770.diff.txt + uploaded by tilghman (license 14) 20100811__issue16770.diff.txt + uploaded by tilghman (license 14) Tested by: jamicque ........ + ................ + +2010-08-17 21:37 +0000 [r282544-282578] David Vossel + + * /, channels/chan_sip.c: Merged revisions 282577 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282577 | dvossel | 2010-08-17 16:36:57 -0500 + (Tue, 17 Aug 2010) | 16 lines Merged revisions 282576 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r282576 | dvossel | 2010-08-17 16:35:17 -0500 (Tue, 17 Aug 2010) + | 9 lines fixes no default transport for temp peer creation in + chan_sip (closes issue #17829) Reported by: falves11 Patches: + issue_17829.rev1.txt uploaded by russell (license 2) + issue_17829.diff uploaded by dvossel (license 671) Tested by: + falves11 ........ ................ + + * /, channels/chan_iax2.c: Merged revisions 282545 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r282545 | dvossel | 2010-08-17 15:08:56 -0500 (Tue, 17 + Aug 2010) | 6 lines ACCEPT message should respond with the new + FORMAT2 ie (closes issue #17804) Reported by: tpanton ........ + + * /, include/asterisk/unaligned.h: Merged revisions 282543 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282543 | dvossel | 2010-08-17 14:34:06 -0500 (Tue, 17 Aug 2010) + | 4 lines fixes truncated uint64_t value in + put_unaligned_uint64_t() function (issue #17804) ........ + +2010-08-16 20:40 +0000 [r282502] Terry Wilson + + * main/channel.c, /: Merged revisions 282468 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282468 | twilson | 2010-08-16 12:53:44 -0500 + (Mon, 16 Aug 2010) | 30 lines Merged revisions 282467 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r282467 | twilson | 2010-08-16 12:32:01 -0500 + (Mon, 16 Aug 2010) | 23 lines Merged revisions 282430 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r282430 | twilson | 2010-08-16 12:06:37 -0500 (Mon, 16 Aug 2010) + | 16 lines Send a SRCCHANGE indication when we masquerade + Masquerading a channel means that the src of the audio is + potentially changing, so send a SRCCHANGE so that RTP-based media + streams can get a new SSRC generated to reflect the change. + Original patch by addix (along with lots of testing--thanks!). + (closes issue #17007) Reported by: addix Patches: + 1001-reset-SSRC-original-channel.diff uploaded by addix (license + 1006) srcchange.diff uploaded by twilson (license 396) Tested by: + addix, twilson Review: https://reviewboard.asterisk.org/r/862/ + ........ ................ ................ + +2010-08-16 18:02 +0000 [r282471] Leif Madsen + + * doc/tex/sounds.tex (added), /, doc/tex/asterisk.tex: Merged + revisions 282470 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282470 | lmadsen | 2010-08-16 13:01:00 -0500 + (Mon, 16 Aug 2010) | 15 lines Merged revisions 282469 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r282469 | lmadsen | 2010-08-16 13:00:09 -0500 (Mon, 16 Aug 2010) + | 7 lines Add information about creating sounds files using the + sounds tools publically available so that others can create their + own sounds prompts using the same tools we use to generate sounds + releases. This allows people creating their own prompts to sound + consistent with the prompts available from the open source + project. SWP-595 ........ ................ + +2010-08-15 13:08 +0000 [r282397] Tzafrir Cohen + + * utils/muted.c, configure, main/Makefile, configure.ac, + main/acl.c, channels/chan_oss.c, main/netsock.c: Support for + GNU/kFreeBSD kFreeBSD is GNU (with glibc) on to of a FreeBSD + kernel. See http://glibc-bsd.alioth.debian.org/porting/PORTING + This patch gets Asterisk close to building on Debian kFreeBSD + i386, mainly by adding an extra test for __GLIBC__ in one or two + (or more) places. OSARCH is set to 'kfreebsd-gnu' DAHDI support + (and support for chan_vpb) was not tested. Review: + https://reviewboard.asterisk.org/r/858/ + +2010-08-14 04:58 +0000 [r282367] Tilghman Lesher + + * /, include/asterisk/sched.h, channels/chan_iax2.c: Merged + revisions 282366 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282366 | tilghman | 2010-08-13 23:53:58 -0500 (Fri, 13 Aug 2010) + | 4 lines Fix our FRACKing issue with chan_iax2 a different way. + Review: https://reviewboard.asterisk.org/r/861/ ........ + +2010-08-13 23:57 +0000 [r282335] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 282334 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r282334 | rmudgett | 2010-08-13 18:53:36 -0500 (Fri, 13 + Aug 2010) | 6 lines PRI CCSS may use a stale dial string for the + recall dial string. If an outgoing call negotiates a different B + channel than initially requested, the saved original dial string + was not transferred to the new B channel. CCSS uses that dial + string to generate the recall dial string. ........ + +2010-08-13 22:27 +0000 [r282237-282304] David Vossel + + * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES, + UPGRADE-1.8.txt: Merged revisions 282302 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) + | 10 lines remove current STUN support from chan_sip.c This patch + removes the current broken/useless stun support from chan_sip. + (closes issue #17622) Reported by: philipp2 Review: + https://reviewboard.asterisk.org/r/855/ ........ + + * /, CHANGES: Merged revisions 282271 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282271 | dvossel | 2010-08-13 15:11:58 -0500 (Fri, 13 Aug 2010) + | 2 lines res_stun_monitor and corresponding options CHANGES + documentation ........ + + * configs/iax.conf.sample, /, channels/chan_sip.c, + include/asterisk/event_defs.h, res/res_stun_monitor.c (added), + configs/res_stun_monitor.conf.sample (added), + configs/sip.conf.sample, channels/chan_iax2.c: Merged revisions + 282269 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282269 | dvossel | 2010-08-13 15:03:56 -0500 (Fri, 13 Aug 2010) + | 4 lines res_stun_monitor for monitoring network changes behind + a NAT device Review: https://reviewboard.asterisk.org/r/854 + ........ + + * /, channels/chan_sip.c: Merged revisions 282236 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282236 | dvossel | 2010-08-13 13:58:10 -0500 + (Fri, 13 Aug 2010) | 23 lines Merged revisions 282235 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r282235 | dvossel | 2010-08-13 13:54:53 -0500 (Fri, 13 Aug 2010) + | 16 lines only do magic pickup when notifycid is enabled A new + way of doing BLF pickup was introduced into 1.6.2. This feature + adds a call-id value into the XML of a SIP_NOTIFY message sent to + alert a subscriber that a device is ringing. This option should + only be enabled when the new 'notifycid' option is set... but + this was not the case. Instead the call-id value was included for + every RINGING Notify message, which caused a regression for + people who used other methods for call pickup. (closes issue + #17633) Reported by: urosh Patches: chan_sip.txt uploaded by + urosh (license ) blf_cid_issue.diff uploaded by dvossel (license + 671) Tested by: dvossel, urosh, okrief, alecdavis ........ + ................ + +2010-08-13 16:08 +0000 [r282202] Terry Wilson + + * /, configure, configure.ac: Merged revisions 282200-282201 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282200 | twilson | 2010-08-13 11:00:02 -0500 (Fri, 13 Aug 2010) + | 10 lines Detect when libsrtp cannot be linked in a shared + library The libsrtp build system currently does not produce a + shared library or a static library compiled with -fPIC, so on + 64-bit systems it is possible that we will get a compile error if + libsrtp is installed and res_srtp is selected in menuselect. This + patch attempts to detect this situation and provide the user with + instructions to work around the problem. ........ r282201 | + twilson | 2010-08-13 11:02:20 -0500 (Fri, 13 Aug 2010) | 2 lines + Whitespace fix :-/ ........ + +2010-08-12 22:52 +0000 [r282132] Jason Parker + + * /, pbx/pbx_config.c: Merged revisions 282131 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r282131 | qwell | 2010-08-12 17:51:44 -0500 + (Thu, 12 Aug 2010) | 16 lines Merged revisions 282130 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r282130 | qwell | 2010-08-12 17:50:54 -0500 + (Thu, 12 Aug 2010) | 9 lines Merged revisions 282129 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r282129 | qwell | 2010-08-12 17:49:28 -0500 (Thu, 12 Aug + 2010) | 1 line Register CLI commands before parsing config, in + case there is a config error. ........ ................ + ................ + +2010-08-12 22:10 +0000 [r282099] Richard Mudgett + + * /, main/ccss.c, include/asterisk/ccss.h: Merged revisions 282098 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282098 | rmudgett | 2010-08-12 17:06:06 -0500 (Thu, 12 Aug 2010) + | 7 lines Separate call completion config parameter allocation + and default initialization. If you ever have a need to reset the + call completion config parameters to defaults, now you can. And + no Virginia, C++ idioms do not always work in C. ........ + +2010-08-12 20:44 +0000 [r282067] Russell Bryant + + * /, CHANGES, main/cli.c: Merged revisions 282066 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282066 | russell | 2010-08-12 15:41:17 -0500 (Thu, 12 Aug 2010) + | 4 lines Add a "core reload" CLI command. Review: + https://reviewboard.asterisk.org/r/859/ ........ + +2010-08-12 20:17 +0000 [r282048] David Vossel + + * /, main/translate.c, CHANGES, include/asterisk/translate.h, + main/cli.c: Merged revisions 282047 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282047 | dvossel | 2010-08-12 15:15:41 -0500 (Thu, 12 Aug 2010) + | 35 lines improved translation paths for wideband codecs The + problem I'm addressing is that Asterisk's current method of + building the least cost translation paths between codecs does not + take into account sample rate. For instance, it was possible for + siren14 (a 32khz codec), to contain the a translation path to + siren7 (a 16khz audio codec) that goes through slin at 8khz. In + this case Asterisk takes a 32khz codec, down samples it to 8khz + and then up samples it to 16khz which is terrible regardless if + it is computationally less expensive. This patch now builds + translation paths that give priority to maintaining the best + possible sample rate before taking into consideration + computational cost. This patch also adds cli commands to expose + what translation paths are actually being used. Changes: 1. + Translation paths will never contain a step that changes the + sample rate unless absolutely necessary. 2. When choosing the + best codec to make two channels compatible. Shared codecs with + the highest sample rate are given priority. 3. A new cli command + to show all translation paths available for a specific codec + 'core show translation paths [codec name]' has been added. 4. + 'core show translation' which displays the translation matrix now + includes the new higher bit audio codecs in the table. 5. 'core + show channel [channel name]' now displays the translation paths + if translation is used. (closes issue #16841) Reported by: + dvossel Review: https://reviewboard.asterisk.org/r/842/ ........ + +2010-08-12 18:04 +0000 [r281983-282016] Russell Bryant + + * main/pbx.c, /: Merged revisions 282015 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r282015 | russell | 2010-08-12 13:03:56 -0500 (Thu, 12 Aug 2010) + | 2 lines Put back pointer value output for ast_debug(), such + that it is only removed for verbose output. ........ + + * main/pbx.c, /: Merged revisions 281982 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281982 | russell | 2010-08-12 11:33:30 -0500 (Thu, 12 Aug 2010) + | 5 lines Remove debugging output from verbose messages. Pointer + values to internal objects is not terribly useful to users in the + verbose messages about adding extensions and contexts. ........ + +2010-08-12 03:08 +0000 [r281914] Jeff Peeler + + * main/channel.c, /: Merged revisions 281913 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281913 | jpeeler | 2010-08-11 22:03:37 -0500 + (Wed, 11 Aug 2010) | 34 lines Merged revisions 281912 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281912 | jpeeler | 2010-08-11 22:01:38 -0500 + (Wed, 11 Aug 2010) | 27 lines Merged revisions 281911 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281911 | jpeeler | 2010-08-11 22:00:14 -0500 (Wed, 11 Aug 2010) + | 20 lines Ensure SSRC is changed when media source is changed to + resolve audio delay. This change causes the SSRC to change right + before the channels are bridged, which is what used to happen. It + seems that fixes were made to attempt limiting SSRC changes, + targeted mainly at sending DTMF. DTMF is not affecting the SSRC + with this change. There are two other control frames sent in + ast_channel_bridge that probably should also be changed to + AST_CONTROL_SRCCHANGE as well, but I'm going to leave this change + up to the discretion of resolving issue #17007. For reference - + old review implementing new control frame SRCCHANGE: + https://reviewboard.asterisk.org/r/540 (closes issue #17404) + Reported by: sdolloff Patches: bug17404.patch uploaded by jpeeler + (license 325) Tested by: sdolloff ........ ................ + ................ + +2010-08-11 21:13 +0000 [r281877] Leif Madsen + + * /, configs/say.conf.sample: Merged revisions 281875 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281875 | lmadsen | 2010-08-11 16:12:13 -0500 + (Wed, 11 Aug 2010) | 21 lines Merged revisions 281873 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281873 | lmadsen | 2010-08-11 16:09:47 -0500 + (Wed, 11 Aug 2010) | 14 lines Merged revisions 281819 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281819 | lmadsen | 2010-08-11 13:28:10 -0500 (Wed, 11 Aug 2010) + | 6 lines Add Danish support to say.conf.sample (closes issue + #17836) Reported by: RoadKill Patches: say.conf.sample.patch.dk + uploaded by RoadKill (license 933) ........ ................ + ................ + +2010-08-11 21:12 +0000 [r281876] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 281874 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281874 | mnicholson | 2010-08-11 16:11:54 -0500 (Wed, 11 Aug + 2010) | 10 lines handle all possible responses to REFER requests + (closes issue #17486) Reported by: davidw Patches: + Issue17486-counterbid.diff.txt uploaded by davidw (license 780) + Tested by: davidw Review: https://reviewboard.asterisk.org/r/837/ + ........ + +2010-08-11 20:38 +0000 [r281871] Richard Mudgett + + * channels/sig_analog.c, /, channels/sig_analog.h: Merged revisions + 281870 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281870 | rmudgett | 2010-08-11 15:30:29 -0500 (Wed, 11 Aug 2010) + | 4 lines Fix a call to analog_set_pulsedial() not setting 0 or 1 + only. * Also a couple minor tweaks. ........ + +2010-08-11 17:55 +0000 [r281765] Leif Madsen + + * /, configs/say.conf.sample: Merged revisions 281764 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281764 | lmadsen | 2010-08-11 12:54:56 -0500 + (Wed, 11 Aug 2010) | 21 lines Merged revisions 281763 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281763 | lmadsen | 2010-08-11 12:54:09 -0500 + (Wed, 11 Aug 2010) | 14 lines Merged revisions 281762 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281762 | lmadsen | 2010-08-11 12:51:40 -0500 (Wed, 11 Aug 2010) + | 6 lines Allow say.conf to handle large numbers ending with + multiple zeros. (closes issue #17833) Reported by: RoadKill + Patches: say.conf.sample.patch.largenumbers uploaded by RoadKill + (license 933) ........ ................ ................ + +2010-08-11 17:29 +0000 [r281761] Matthew Nicholson + + * /, channels/chan_sip.c: Merged revisions 281760 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281760 | mnicholson | 2010-08-11 12:27:59 -0500 (Wed, 11 Aug + 2010) | 4 lines Avoid a deadlock in add_header_max_forwards(). + Related to r276951 ........ + +2010-08-11 15:20 +0000 [r281726] Tilghman Lesher + + * /, apps/app_readexten.c: Merged revisions 281723 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281723 | tilghman | 2010-08-11 10:18:40 -0500 + (Wed, 11 Aug 2010) | 14 lines Merged revisions 281722 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r281722 | tilghman | 2010-08-11 10:17:20 -0500 (Wed, 11 Aug 2010) + | 7 lines Only set status TIMEOUT, if we have no digits. (closes + issue #15188) Reported by: jcovert Patches: + app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license + 551) ........ ................ + +2010-08-11 13:31 +0000 [r281688] + + * main/netsock2.c, /, include/asterisk/netsock2.h, + configs/sip.conf.sample, channels/sip/config_parser.c: Merged + revisions 281687 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281687 | simon.perreault | 2010-08-11 09:30:59 -0400 (Wed, 11 + Aug 2010) | 9 lines Fix parsing of IPv6 address literals in + outboundproxy (closes issue #17757) Reported by: oej Patches: + 17757.diff uploaded by sperreault (license 252) sip.conf.diff + uploaded by sperreault (license 252) Tested by: oej ........ + +2010-08-10 21:50 +0000 [r281530-281651] Russell Bryant + + * /, configs/sip.conf.sample, UPGRADE-1.8.txt, + channels/sip/include/sip.h: Merged revisions 281650 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r281650 | russell | 2010-08-10 16:47:31 -0500 (Tue, 10 + Aug 2010) | 5 lines Change the default value for alwaysauthreject + in sip.conf to "yes". (closes issue #17756) Reported by: oej + ........ + + * /, main/sched.c: Merged revisions 281575 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281575 | russell | 2010-08-10 13:05:07 -0500 + (Tue, 10 Aug 2010) | 16 lines Merged revisions 281574 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r281574 | russell | 2010-08-10 13:04:32 -0500 (Tue, 10 Aug 2010) + | 9 lines Don't move the time threshold for running scheduled + events on every iteration. Instead, only calculate the time + threshold each time ast_sched_runq() is called. (closes issue + #17742) Reported by: schmidts Patches: sched.c.patch uploaded by + schmidts (license 1077) ........ ................ + + * apps/app_dial.c, /: Merged revisions 281568 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281568 | russell | 2010-08-10 12:48:42 -0500 + (Tue, 10 Aug 2010) | 22 lines Merged revisions 281567 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281567 | russell | 2010-08-10 12:47:13 -0500 + (Tue, 10 Aug 2010) | 15 lines Merged revisions 281566 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281566 | russell | 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) + | 8 lines Reset visible indication after answer. (closes issue + #17641) Reported by: klaus3000 Patches: + ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by + klaus3000 (license 65) Tested by: schmidts ........ + ................ ................ + + * /, channels/chan_sip.c: Merged revisions 281532 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281532 | russell | 2010-08-10 11:54:20 -0500 (Tue, 10 Aug 2010) + | 8 lines Ensure that the proper external address is used for the + RTP destination. (closes issue #17044) Reported by: ebroad Tested + by: ebroad Review: https://reviewboard.asterisk.org/r/566/ + ........ + + * /, main/cli.c: Merged revisions 281529 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281529 | russell | 2010-08-10 11:21:58 -0500 (Tue, 10 Aug 2010) + | 8 lines Resolve a problem with channel name tab completion. + Hitting tab without typing any part of a channel name resulted in + no results. This now results in getting a full list of active + channels, just as it did in previous versions of Asterisk. + Review: https://reviewboard.asterisk.org/r/818/ ........ + +2010-08-10 07:26 +0000 [r281498] TransNexus OSP Development + + * apps/app_osplookup.c: Fixed the issue caused by EXTEN including + user parameters. + +2010-08-09 23:04 +0000 [r281467] Jeff Peeler + + * channels/chan_local.c, /: Merged revisions 281466 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r281466 | jpeeler | 2010-08-09 18:04:02 -0500 (Mon, 09 + Aug 2010) | 2 lines Add some more stuff to copy from 281429. + ........ + +2010-08-09 20:49 +0000 [r281433] David Vossel + + * /, channels/chan_sip.c: Merged revisions 281432 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281432 | dvossel | 2010-08-09 15:47:53 -0500 + (Mon, 09 Aug 2010) | 20 lines Merged revisions 281430 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r281430 | dvossel | 2010-08-09 15:46:50 -0500 (Mon, 09 Aug 2010) + | 13 lines fixes SIP peers memory leak We zeroed out the peer's + addr before it was removed from the peers_by_ip container. This + made it impossible to be removed from the container as the addr + is the key used by the container to find the peer. (closes issue + #17774) Reported by: kkm Patches: + 017774-sip-peer-leak-1.6.2.10.diff uploaded by kkm (license 888) + 017774-sip-peer-leak-1.8.diff uploaded by kkm (license 888) + ........ ................ + +2010-08-09 20:46 +0000 [r281431] Jeff Peeler + + * channels/chan_local.c, /: Merged revisions 281429 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281429 | jpeeler | 2010-08-09 15:43:54 -0500 + (Mon, 09 Aug 2010) | 27 lines Merged revisions 281391 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r281391 | jpeeler | 2010-08-09 15:07:29 -0500 + (Mon, 09 Aug 2010) | 20 lines Merged revisions 281390 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r281390 | jpeeler | 2010-08-09 15:04:30 -0500 (Mon, 09 Aug 2010) + | 13 lines Prevent loss of Caller ID information set on local + channel after masquerade. Caller ID set on the channel before a + masquerade occurs when using a local channel would cause the + information to be lost. The problem was that the information was + set on a channel destined to be hung up. The somewhat confusing + fix is to detect if any Caller ID has been set on the channel and + if so preswap the Caller ID data so that basically the masquerade + puts the data back. (closes issue #17138) Reported by: kobaz + Review: https://reviewboard.asterisk.org/r/847/ ........ + ................ ................ + +2010-08-09 14:52 +0000 [r281359] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 281358 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281358 | mnicholson | 2010-08-09 09:49:38 -0500 (Mon, 09 Aug + 2010) | 4 lines Validate minrate, maxrate, and modem settings + before attempting a fax session. FAX-224 ........ + +2010-08-09 14:32 +0000 [r281357] + + * /, configs/sip.conf.sample: Merged revisions 281356 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r281356 | simon.perreault | 2010-08-09 10:31:40 -0400 + (Mon, 09 Aug 2010) | 2 lines Added comment about IPv4-mapped IPv6 + addresses and the output of netstat. ........ + +2010-08-09 12:52 +0000 [r281295-281326] Russell Bryant + + * /, configs/cdr.conf.sample: Merged revisions 281325 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r281325 | russell | 2010-08-09 07:51:43 -0500 (Mon, 09 + Aug 2010) | 2 lines Add a couple of default values to the + documentation of cdr.conf. ........ + + * /, configs/cdr.conf.sample: Merged revisions 281294 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r281294 | russell | 2010-08-09 07:14:34 -0500 (Mon, 09 + Aug 2010) | 5 lines Reorder some options in cdr.conf.sample. Put + all of the options that affect the contents of CDRs together, + instead of having the batch mode options in the middle of them. + ........ + +2010-08-07 22:36 +0000 [r281226-281257] Damien Wedhorn + + * channels/chan_skinny.c: Fix up handling and indications during + transfer. Cleaned up handling of onhook indications and added + indications if more than one sub on device. Also fixes issue in + 12324 so that the phone can call itself without locking up. + (closes issue #17692) Reported by: jmhunter Patches: + chan_skinny-transfer-v4.txt uploaded by DEA (license 3) + skinnytransfver.v8.diff uploaded by wedhorn (license 30) Tested + by: jmhunter, salecha, wedhorn Review: NA + + * channels/chan_skinny.c: Move call answering stuff into new + setsubstate_connected. Move call answering stuff into new + setsubstate_connected. Also add sub->substate var and set it to + SUBSTATE_CONNECTED in setsubstate_connected. (closes issue + #17772) Reported by: wedhorn Patches: + cleanup.stateconnected2.diff uploaded by wedhorn (license 30) + Tested by: wedhorn, salecha Review: NA + + * channels/chan_skinny.c: Start rtp on answer before the answer is + queued (closes issue #17770) Reported by: salecha Patches: + skinny.answercrash.diff uploaded by wedhorn (license 30) Tested + by: salecha Review: NA + +2010-08-06 18:58 +0000 [r281086] Tilghman Lesher + + * /, main/utils.c: Merged revisions 281085 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r281085 | tilghman | 2010-08-06 13:57:10 -0500 (Fri, 06 Aug 2010) + | 8 lines Fix alignment of stringfields on the SPARC architecture + (closes issue #17789) Reported by: Ian Mason Patches: + 20100806__issue17789__2.diff.txt uploaded by tilghman (license + 14) Tested by: Ian_Mason ........ + +2010-08-05 13:19 +0000 [r281054] Russell Bryant + + * main/cdr.c, /: Merged revisions 281052 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r281052 | russell | 2010-08-05 08:16:11 -0500 + (Thu, 05 Aug 2010) | 16 lines Merged revisions 281051 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r281051 | russell | 2010-08-05 08:11:32 -0500 (Thu, 05 Aug 2010) + | 9 lines Cleanup default option value handling for cdr.conf + [general]. The default values would differ depending on whether + or not cdr.conf exists. That is no longer the case. Apply a + default value to the unanswered option. Define all default values + as named constants. ........ ................ + +2010-08-05 07:47 +0000 [r280985] Tilghman Lesher + + * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 280984 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280984 | tilghman | 2010-08-05 02:46:36 -0500 + (Thu, 05 Aug 2010) | 22 lines Merged revisions 280983 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280983 | tilghman | 2010-08-05 02:40:47 -0500 + (Thu, 05 Aug 2010) | 15 lines Merged revisions 280982 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280982 | tilghman | 2010-08-05 02:28:33 -0500 (Thu, 05 Aug 2010) + | 8 lines Change context lock back to a mutex, because + functionality depends upon the lock being recursive. (closes + issue #17643) Reported by: zerohalo Patches: + 20100726__issue17643.diff.txt uploaded by tilghman (license 14) + Tested by: zerohalo ........ ................ ................ + +2010-08-04 15:22 +0000 [r280910] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 280909 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280909 | mnicholson | 2010-08-04 10:11:13 -0500 (Wed, 04 Aug + 2010) | 2 lines Initialize FAXOPT() status variables in sendfax + and receivefax instead of when the details structure is created. + ........ + +2010-08-04 14:05 +0000 [r280810-280880] Tilghman Lesher + + * /, channels/chan_mgcp.c: Merged revisions 280879 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280879 | tilghman | 2010-08-04 09:04:07 -0500 (Wed, 04 + Aug 2010) | 14 lines Check cur value before attempting a deref. + (closes issue #17775) Reported by: svinson Patches: + 20100804__issue17775.diff.txt uploaded by tilghman (license 14) + Tested by: svinson (closes issue #17743) Reported by: tgruenberg + Patches: 20100804__issue17775.diff.txt uploaded by tilghman + (license 14) Tested by: tgruenberg ........ + + * /, funcs/func_strings.c, CHANGES: Merged revisions 280809 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280809 | tilghman | 2010-08-03 15:25:10 -0500 (Tue, 03 Aug 2010) + | 12 lines Sneak FIELDNUM() into 1.8. Returns a 1-based index + into a list of a specified item. Matches up with FIELDQTY() and + CUT(). (closes issue #17713) Reported by: gareth Patches: + svn-279754.diff uploaded by gareth (license 208) Tested by: + gareth, tilghman Review: https://reviewboard.asterisk.org/r/810/ + ........ + +2010-08-03 19:59 +0000 [r280745-280780] + + * /, channels/chan_sip.c: Merged revisions 280778 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280778 | simon.perreault | 2010-08-03 15:54:03 -0400 (Tue, 03 + Aug 2010) | 9 lines Fixed IPv6-related SIP parsing bugs. (closes + issue #17663) Reported by: oej Patches: diff uploaded by + sperreault (license 252) diff2 uploaded by sperreault (license + 252) get_domain.diff uploaded by sperreault (license 252) + ........ + + * /, configs/sip.conf.sample: Merged revisions 280777 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280777 | simon.perreault | 2010-08-03 15:53:07 -0400 + (Tue, 03 Aug 2010) | 8 lines Better documentation related to + IPv6. (closes issue #17737) Reported by: oej Patches: doc.diff + uploaded by sperreault (license 252) Tested by: mmichelson + ........ + + * contrib/realtime/mysql/voicemail.sql, channels/chan_sip.c, + contrib/realtime/mysql/iaxfriends.sql, + contrib/realtime/mysql/meetme.sql, + contrib/realtime/postgresql/realtime.sql, + configs/sip.conf.sample, contrib/realtime/mysql/sipfriends.sql: + Reverted r280706 and r280707. Will commit in branch 1.8 and merge + to trunk properly. + +2010-08-03 18:50 +0000 [r280743] Russell Bryant + + * contrib/scripts/get_mp3_source.sh (added), /, addons/Makefile, + addons/mp3 (removed): Merged revisions 280742 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280742 | russell | 2010-08-03 13:48:45 -0500 (Tue, 03 Aug 2010) + | 4 lines Remove the MP3 decoder source code and replace it with + a small shell script. Review: + https://reviewboard.asterisk.org/r/836/ ........ + +2010-08-03 18:43 +0000 [r280741] Tilghman Lesher + + * /, doc/asterisk.8, doc/Makefile (added), doc/asterisk.sgml: + Merged revisions 280740 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280740 | tilghman | 2010-08-03 13:42:24 -0500 + (Tue, 03 Aug 2010) | 9 lines Merged revisions 280739 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r280739 | tilghman | 2010-08-03 13:39:28 -0500 (Tue, 03 + Aug 2010) | 2 lines Document -B and -W flags and regenerate + manpage from sgml ........ ................ + +2010-08-03 16:52 +0000 [r280706-280707] + + * channels/chan_sip.c: Fixed IPv6-related SIP parsing bugs. (closes + issue #17663) Reported by: oej Patches: diff uploaded by + sperreault (license 252) diff2 uploaded by sperreault (license + 252) get_domain.diff uploaded by sperreault (license 252) + + * configs/sip.conf.sample: Better documentation related to IPv6. + +2010-08-02 21:28 +0000 [r280629-280673] Tilghman Lesher + + * /, apps/app_voicemail.c: Merged revisions 280672 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280672 | tilghman | 2010-08-02 16:27:25 -0500 + (Mon, 02 Aug 2010) | 9 lines Merged revisions 280671 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02 + Aug 2010) | 2 lines Allow the pipe, but also allow the comma + ........ ................ + + * /, main/Makefile: Merged revisions 280628 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280628 | tilghman | 2010-08-02 09:41:46 -0500 (Mon, 02 Aug 2010) + | 2 lines Make this a little more deterministic... we want the + latest value, not just a 1 somewhere. ........ + +2010-08-02 14:30 +0000 [r280627] David Vossel + + * channels/chan_sip.c: if totag is not present for an ACK request, + do not send an error response + +2010-08-02 14:28 +0000 [r280626] Tilghman Lesher + + * /, main/Makefile: Merged revisions 280624 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280624 | tilghman | 2010-08-02 09:27:20 -0500 (Mon, 02 Aug 2010) + | 2 lines Apparently, the values in makeopts are sometimes 1:1 + and sometimes 1. Compensate for this. ........ + +2010-07-30 09:12 +0000 [r280589] Damien Wedhorn + + * channels/chan_skinny.c: Cleanup transmit_ for handle_register and + keepalives Moved inline packet sending to transmit_ subs. Removed + handle_keep_alive and handle_register_message to inline in + handle_message. Also moved transmit_response(d) to + transmit_response_bysessions(s) and created a wrapper + transmit_response(d) that calls + transmit_response_bysession(d->session). (closes issue #16980) + Reported by: wedhorn Patches: skinny-clean06b.diff uploaded by + wedhorn (license 30) Tested by: wedhorn, DEA Review: NA + +2010-07-29 21:08 +0000 [r280559] Matthew Nicholson + + * /, res/res_fax.c: Merged revisions 280557 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280557 | mnicholson | 2010-07-29 16:07:21 -0500 (Thu, 29 Jul + 2010) | 4 lines Fix regression introduced in r1664. Give the fax + stack time to shutdown and populate the FAXOPT output variables. + FAX-222 ........ + +2010-07-29 21:06 +0000 [r280555] Paul Belanger + + * CHANGES, channels/chan_iax2.c: PeerStatus now includes Address + and Port (closes issue #17730) Reported by: jkroon Patches: + iax2-peerstate-address.patch uploaded by jkroon (license 714) + Tested by: lmadsen + +2010-07-29 20:44 +0000 [r280553] David Vossel + + * /, channels/chan_sip.c: Merged revisions 280552 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280552 | dvossel | 2010-07-29 15:43:47 -0500 + (Thu, 29 Jul 2010) | 17 lines Merged revisions 280551 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280551 | dvossel | 2010-07-29 15:42:29 -0500 (Thu, 29 Jul 2010) + | 11 lines fixes wrong SRV query for TLS connection (closes issue + #17612) Reported by: marcelloceschia Patches: + chan-sip_srvQuery.patch uploaded by marcelloceschia (license + 1079) chan-sip_Trunk_srvQuery.patch uploaded by st (license 907) + chan-sip_asterisk18b1_srvQuery.patch uploaded by marcelloceschia + (license 1079) Tested by: marcelloceschia, st, pabelanger + ........ ................ + +2010-07-29 20:36 +0000 [r280550] Russell Bryant + + * /, configs/ccss.conf.sample: Merged revisions 280549 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280549 | russell | 2010-07-29 15:35:30 -0500 (Thu, 29 + Jul 2010) | 5 lines Add header to ccss.conf to appease oej. + (closes issue #17755) Reported by: oej ........ + +2010-07-29 19:48 +0000 [r280520] Sean Bright + + * /, channels/sig_pri.c: Merged revisions 280519 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280519 | seanbright | 2010-07-29 15:47:16 -0400 (Thu, 29 Jul + 2010) | 7 lines Fix compilation error in chan_dahdi (strdupa -> + ast_strdupa). (closes issue #17751) Reported by: b11d Patches: + strdupa_oops.diff uploaded by malcolmd (license 924) ........ + +2010-07-29 19:35 +0000 [r280459-280518] David Vossel + + * channels/chan_sip.c: respond with 481 when request requiring + totag has no totag to match against + + * main/channel.c, /: Merged revisions 280450 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280450 | dvossel | 2010-07-29 14:13:27 -0500 + (Thu, 29 Jul 2010) | 25 lines Merged revisions 280449 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280449 | dvossel | 2010-07-29 14:05:25 -0500 + (Thu, 29 Jul 2010) | 18 lines Merged revisions 280448 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280448 | dvossel | 2010-07-29 14:04:23 -0500 (Thu, 29 Jul 2010) + | 12 lines fixes issue with translator frame not getting freed A + translator frame even if it local storage so the translation path + can be freed. This issue prevented g729 licenses from being freed + up. (closes issue #17630) Reported by: manvirr Patches: + encoder_fix.diff uploaded by dvossel (license 671) Tested by: + manvirr, dvossel ........ ................ ................ + +2010-07-29 18:51 +0000 [r280447] Paul Belanger + + * /, tests/test_utils.c: Merged revisions 280446 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280446 | pabelanger | 2010-07-29 14:37:32 -0400 (Thu, 29 Jul + 2010) | 2 lines Remove res_crypto dependency. ........ + +2010-07-29 16:47 +0000 [r280416] Jean Galarneau + + * /, apps/app_meetme.c: Merged revisions 280346 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280346 | jeang | 2010-07-29 11:07:16 -0500 + (Thu, 29 Jul 2010) | 17 lines Merged revisions 280345 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280345 | jeang | 2010-07-29 11:01:35 -0500 + (Thu, 29 Jul 2010) | 10 lines Merged revisions 280341 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | + 2 lines Fix a dsp structure leak occuring when a local channel is + put into a meetme conference, then masquaraded away. ABE-2422 + ........ ................ ................ + +2010-07-29 16:45 +0000 [r280415] Paul Belanger + + * /, tests/test_utils.c: Merged revisions 280414 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280414 | pabelanger | 2010-07-29 12:44:22 -0400 (Thu, 29 Jul + 2010) | 2 lines crypto_loaded_test depends on res_crypto, else + test will fail. ........ + +2010-07-29 16:26 +0000 [r280395] Russell Bryant + + * main/rtp_engine.c, /: Merged revisions 280391 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280391 | russell | 2010-07-29 11:25:43 -0500 (Thu, 29 Jul 2010) + | 2 lines Don't blow up if get_codec() was not provided in the + RTP glue. ........ + +2010-07-29 15:58 +0000 [r280308-280344] Matthew Nicholson + + * channels/chan_usbradio.c, /: Merged revisions 280343 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280343 | mnicholson | 2010-07-29 10:57:57 -0500 (Thu, + 29 Jul 2010) | 4 lines Use PRIx64 instead of PRId64 in format + string. related to r280302 ........ + + * channels/chan_usbradio.c, /: Merged revisions 280302 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280302 | pabelanger | 2010-07-28 19:45:34 -0500 (Wed, + 28 Jul 2010) | 2 lines Use PRId64 with format_t ........ + + * channels/chan_usbradio.c: Make chan_usbradio.c build on 64bit + platforms. + + * main/channel.c, channels/chan_local.c, /: Merged revisions 280307 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280307 | mnicholson | 2010-07-29 08:56:35 -0500 + (Thu, 29 Jul 2010) | 11 lines Merged revisions 280306 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280306 | mnicholson | 2010-07-29 08:45:11 -0500 (Thu, 29 Jul + 2010) | 2 lines Implement support for ast_channel_queryoption on + local channels. Currently only AST_OPTION_T38_STATE is supported. + ABE-2229 Review: https://reviewboard.asterisk.org/r/813/ ........ + Additionally, pass AST_CONTROL_T38_PARAMETERS control frames + through generic bridges. This change appears to have been + unintentionally left out of rev 203699. ................ + +2010-07-28 20:50 +0000 [r280270] Jeff Peeler + + * /, channels/sip/reqresp_parser.c: Merged revisions 280269 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280269 | jpeeler | 2010-07-28 15:49:26 -0500 (Wed, 28 Jul 2010) + | 2 lines Give test category missing leading slash ........ + +2010-07-28 20:19 +0000 [r280247] Richard Mudgett + + * channels/chan_dahdi.c, /: Merged revisions 280235 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280235 | rmudgett | 2010-07-28 15:12:16 -0500 + (Wed, 28 Jul 2010) | 9 lines Merged revisions 280229 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r280229 | rmudgett | 2010-07-28 14:57:49 -0500 (Wed, 28 + Jul 2010) | 2 lines Add missing enum value "unknown" to the SS7 + called_nai and calling_nai config options. ........ + ................ + +2010-07-28 20:04 +0000 [r280234] Jason Parker + + * /, sounds/Makefile: Merged revisions 280233 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280233 | qwell | 2010-07-28 15:03:22 -0500 + (Wed, 28 Jul 2010) | 13 lines Merged revisions 280231 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280231 | qwell | 2010-07-28 15:02:27 -0500 (Wed, 28 Jul 2010) | + 6 lines Work around some silly behavior on BSD. A non-zero exit + from a subshell should make the build fail. (closes issue #17621) + ........ ................ + +2010-07-28 19:37 +0000 [r280226] Terry Wilson + + * res/res_rtp_asterisk.c, /: Merged revisions 280225 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280225 | twilson | 2010-07-28 12:34:42 -0700 (Wed, 28 + Jul 2010) | 3 lines Do rtp/rtcp debugging when it is turned on + w/o filtering ........ + +2010-07-28 18:25 +0000 [r280196] Jason Parker + + * /, sounds/Makefile: Merged revisions 280195 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280195 | qwell | 2010-07-28 13:24:29 -0500 + (Wed, 28 Jul 2010) | 16 lines Merged revisions 280193 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280193 | qwell | 2010-07-28 13:05:54 -0500 (Wed, 28 Jul 2010) | + 9 lines Remove unnecessary subshells. Attempt to make + checksumming work. Also improves readability. (issue #17621) + Reported by: bjm Review: https://reviewboard.asterisk.org/r/808/ + ........ ................ + +2010-07-28 16:53 +0000 [r280162] Sean Bright + + * /, apps/app_queue.c: Merged revisions 280161 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280161 | seanbright | 2010-07-28 12:52:12 -0400 + (Wed, 28 Jul 2010) | 15 lines Merged revisions 280160 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul + 2010) | 8 lines Plug a reference leak in app_queue when adding + members dynamically. (closes issue #17738) Reported by: + bobwienholt Patches: issue17738.patch uploaded by bobwienholt + (license 950) Tested by: bobwienholt, seanbright ........ + ................ + +2010-07-28 14:14 +0000 [r280093] Olle Johansson + + * channels/chan_sip.c: Formatting changes + +2010-07-28 13:53 +0000 [r280091] Leif Madsen + + * contrib/scripts/live_ast, /: Merged revisions 280090 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r280090 | lmadsen | 2010-07-28 08:52:50 -0500 + (Wed, 28 Jul 2010) | 16 lines Merged revisions 280089 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r280089 | lmadsen | 2010-07-28 08:51:16 -0500 + (Wed, 28 Jul 2010) | 9 lines Merged revisions 280088 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.4 + ........ r280088 | lmadsen | 2010-07-28 08:50:38 -0500 (Wed, 28 + Jul 2010) | 1 line Update help text to be less confusing. + ........ ................ ................ + +2010-07-28 13:02 +0000 [r280059] Russell Bryant + + * /, res/res_crypto.c: Merged revisions 280058 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r280058 | russell | 2010-07-28 08:01:15 -0500 (Wed, 28 Jul 2010) + | 2 lines s/init keys/keys init/ ........ + +2010-07-28 01:39 +0000 [r280024] Paul Belanger + + * channels/chan_usbradio.c, /: Merged revisions 280023 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r280023 | pabelanger | 2010-07-27 21:37:10 -0400 (Tue, + 27 Jul 2010) | 5 lines Resolve compiler warning about formatting + (closes issue #17732) Reported by: pabelanger ........ + +2010-07-27 21:16 +0000 [r279954] Russell Bryant + + * /, utils, codecs, main/db1-ast/mpool, Makefile.rules, cdr, + formats, codecs/gsm/src, bridges, codecs/lpc10, configure, + main/editline, channels/sip, pbx, res/ael, channels, + main/stdtime, main/editline/np, main/db1-ast/hash, cel, apps, + configure.ac, main/db1-ast/db, res/ais, res/snmp, funcs, + main/db1-ast/btree, codecs/g722, main, main/db1-ast/recno, + makeopts.in, res: Merged revisions 279953 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279953 | russell | 2010-07-27 16:16:05 -0500 (Tue, 27 Jul 2010) + | 5 lines Add --enable-coverage option to configure script. This + option enables the proper compiler flags for tracking code + coverage, which is useful along side automated testing. ........ + +2010-07-27 20:59 +0000 [r279951] David Vossel + + * main/channel.c, /, include/asterisk/audiohook.h, + main/audiohook.c: Merged revisions 279949 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279949 | dvossel | 2010-07-27 15:57:00 -0500 + (Tue, 27 Jul 2010) | 31 lines Merged revisions 279946 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r279946 | dvossel | 2010-07-27 15:54:32 -0500 + (Tue, 27 Jul 2010) | 24 lines Merged revisions 279945 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r279945 | dvossel | 2010-07-27 15:33:40 -0500 (Tue, 27 Jul 2010) + | 19 lines remove empty audiohook write list on channel If a + channel has an audiohook write list created on it, that list + stays on the channel until the channel is destroyed. There is no + reason to keep that list on the channel if it becomes empty. If + it is empty that just means we are doing needless translating for + every ast_read and ast_write. This patch removes the audiohook + list from the channel once it is detected to be empty on either a + read or write. If a audiohook is added back to the channel after + this list is destroyed, the list just gets recreated as if it + never existed to begin with. (closes issue #17630) Reported by: + manvirr Review: https://reviewboard.asterisk.org/r/799/ ........ + ................ ................ + +2010-07-27 19:55 +0000 [r279917] Russell Bryant + + * channels/chan_dahdi.c, /, channels/sig_pri.c: Merged revisions + 279916 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279916 | russell | 2010-07-27 14:50:56 -0500 (Tue, 27 Jul 2010) + | 12 lines Fix inband DTMF detection on outgoing ISDN calls. This + is a regression from the sig_pri split from chan_dahdi. When a + call is first initiated, the inband DTMF detector is not enabled + if it's an outgoing ISDN call. However, it needs to be turned on + once the media path starts up. This handling was put back in the + open_media() callback of chan_dahdi. In sig_pri, open_media() + calls were added to a few places where it was needed, including + handling of PRI_EVENT_RINGING, PRI_EVENT_PROGRESS, and + PRI_EVENT_PROCEEDING. Thanks to rmudgett for helping me with the + patch! ........ + +2010-07-27 18:55 +0000 [r279888] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 279887 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279887 | mmichelson | 2010-07-27 13:54:07 -0500 (Tue, 27 Jul + 2010) | 16 lines Fix parsing error in sip_sipredirect(). The code + was written in a way that did a bad job of parsing the port out + of a URI. Specifically, it would do badly when dealing with an + IPv6 address. In this particular scenario, there was no value + from parsing the port out, so I just removed that logic. And + while I was messing around in the function, I changed some + variable names to be more descriptive. (closes issue #17661) + Reported by: oej Patches: 17661.diff uploaded by mmichelson + (license 60) ........ + +2010-07-27 16:41 +0000 [r279851] Jason Parker + + * /, sounds/Makefile: Merged revisions 279850 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279850 | qwell | 2010-07-27 11:40:05 -0500 + (Tue, 27 Jul 2010) | 9 lines Merged revisions 279849 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r279849 | qwell | 2010-07-27 11:39:16 -0500 (Tue, 27 Jul + 2010) | 1 line Simply sounds/Makefile some more. ........ + ................ + +2010-07-27 16:11 +0000 [r279818] David Vossel + + * main/netsock2.c, /, channels/chan_sip.c: Merged revisions 279817 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279817 | dvossel | 2010-07-27 11:09:15 -0500 (Tue, 27 Jul 2010) + | 2 lines fix sip transaction match with authentication, fix + confusing log message when using getaddrinfo ........ + +2010-07-27 16:08 +0000 [r279816] Russell Bryant + + * main/channel.c, channels/chan_dahdi.c, /: Merged revisions + 279636,279815 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279636 | russell | 2010-07-26 16:53:30 -0500 (Mon, 26 Jul 2010) + | 2 lines Ignore a control subclass of -1 in + ast_waitfordigit_full(). ........ r279815 | russell | 2010-07-27 + 11:06:58 -0500 (Tue, 27 Jul 2010) | 4 lines Support "channels" in + addition to "channel" in chan_dahdi.conf. Review: + https://reviewboard.asterisk.org/r/804 ........ + +2010-07-27 15:16 +0000 [r279786] Mark Michelson + + * /, channels/chan_sip.c: Merged revisions 279785 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279785 | mmichelson | 2010-07-27 10:15:22 -0500 + (Tue, 27 Jul 2010) | 20 lines Merged revisions 279784 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r279784 | mmichelson | 2010-07-27 10:13:24 -0500 (Tue, 27 Jul + 2010) | 14 lines Fix bad behavior of dynamic_exclude_static + option in sip.conf. We were attempting to create a contactdeny + rule based on the peer's IP address before the peer's IP address + had been set. By moving the processing further down in the + function, we can ensure stuff works as we expect for it to. + (closes issue #17717) Reported by: mmichelson Patches: + 17717.patch uploaded by mmichelson (license 60) Tested by: + DennisD ........ ................ + +2010-07-27 03:02 +0000 [r279727-279756] Paul Belanger + + * channels/chan_dahdi.c, /: Merged revisions 279755 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r279755 | pabelanger | 2010-07-26 22:57:33 -0400 (Mon, + 26 Jul 2010) | 10 lines If dringXcontext is null, fallback to + default context value. (closes issue #17693) Reported by: + iasgoscouk Patches: issue17693.patch uploaded by pabelanger + (license 224) Tested by: iasgoscouk Review: + https://reviewboard.asterisk.org/r/803/ ........ + + * /, main/http.c: Merged revisions 279726 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279726 | pabelanger | 2010-07-26 21:53:38 -0400 (Mon, 26 Jul + 2010) | 9 lines Use ast_sockaddr_setnull() when http is not + enabled. Otherwise, ast_tcptls_server_start() will still start + http. (closes issue #17708) Reported by: pabelanger Patches: + http.patch uploaded by pabelanger (license 224) ........ + +2010-07-27 01:39 +0000 [r279725] Russell Bryant + + * CHANGES: Make a formatting change. (Demonstrating the commit IRC + bot to pabelanger) + +2010-07-26 23:35 +0000 [r279692] Paul Belanger + + * /, CHANGES, UPGRADE-1.8.txt: Merged revisions 279689 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r279689 | pabelanger | 2010-07-26 19:29:34 -0400 (Mon, + 26 Jul 2010) | 2 lines Updated documentation for FAX logger + level. ........ + +2010-07-26 23:06 +0000 [r279659] Jason Parker + + * /, sounds/Makefile.380 (removed), configure, + include/asterisk/autoconfig.h.in, sounds/Makefile.381 (removed), + configure.ac, sounds/Makefile (added): Merged revisions 279658 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279658 | qwell | 2010-07-26 18:03:38 -0500 + (Mon, 26 Jul 2010) | 12 lines Merged revisions 279657 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r279657 | qwell | 2010-07-26 17:59:52 -0500 (Mon, 26 Jul 2010) | + 5 lines Really fix sounds Makefile (and make it readableish). + There was a rather large syntax error that should have caused ALL + versions of GNU make to fail. I don't know how it worked. + ........ ................ + +2010-07-26 21:21 +0000 [r279602-279624] Tilghman Lesher + + * /, configure, configure.ac: Merged revisions 279619 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279619 | tilghman | 2010-07-26 16:20:12 -0500 + (Mon, 26 Jul 2010) | 9 lines Merged revisions 279609 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r279609 | tilghman | 2010-07-26 16:18:17 -0500 (Mon, 26 + Jul 2010) | 2 lines Dunno why this worked on my machine, but it + works better this way. ........ ................ + + * /, res/res_config_ldap.c: Merged revisions 279601 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279601 | tilghman | 2010-07-26 16:07:45 -0500 + (Mon, 26 Jul 2010) | 19 lines Merged revisions 279597 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r279597 | ghenry | 2010-07-26 15:25:54 -0500 (Mon, 26 Jul 2010) | + 13 lines Apply all patches in: + https://issues.asterisk.org/view.php?id=13573 (closes issue + #13573) Reported by: navkumar Patches: + res_config_ldap-category.diff uploaded by navkumar (license 580) + res_config_ldap.patch uploaded by bencer (license 961) + res_config_ldap uploaded by bencer (license 961) Tested by: + suretec ........ ................ + +2010-07-26 21:07 +0000 [r279600] Gavin Henry + + * /: Merged revisions 279598 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279598 | ghenry | 2010-07-26 21:58:12 +0100 (Mon, 26 Jul 2010) | + 21 lines Merged revisions 279597 via svnmerge from + https://origsvn.digium.com/svn/asterisk/1.6.2 + ----------------------------------------------------------------------- + r279597 | ghenry | 2010-07-26 15:25:53 -0500 (Mon, 26 Jul 2010) | + 13 lines Apply all patches in: + https://issues.asterisk.org/view.php?id=13573 [^] (closes issue + 0013573) Reported by: navkumar Patches: + res_config_ldap-category.diff uploaded by navkumar (license 580) + res_config_ldap.patch uploaded by bencer (license 961) + res_config_ldap uploaded by bencer (license 961) Tested by: + suretec + ------------------------------------------------------------------------ + ........ + +2010-07-26 20:00 +0000 [r279569] David Vossel + + * /, channels/chan_sip.c, channels/sip/reqresp_parser.c, + channels/sip/include/sip.h, + channels/sip/include/reqresp_parser.h: Merged revisions 279568 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279568 | dvossel | 2010-07-26 14:59:03 -0500 (Mon, 26 Jul 2010) + | 21 lines transaction matching using top most Via header This + patch modifies the way chan_sip.c does transaction to dialog + matching. Asterisk now stores information in the top most Via + header of the initial incoming request and compares that against + other Requests that have the same call-id. This results in + Asterisk being able to detect a forked call in which it has + received multiple legs of the fork. I completely stripped out the + previous matching code and made the comparisons a little more + explicit and easier to understand. My comments in the code should + offer all the details involving this patch. This patch also fixes + a bug with the usage of the OBJ-MULTIPLE flag to find multiple + dialogs with the same call-id. Since the callback function was + returning (CMP_MATCH | CMP_STOP) only the first item found was + being returned. I fixed this by making a new callback function + for finding multiple dialogs that only returns (CMP_MATCH) on a + match allowing for multiple items to be returned. Review: + https://reviewboard.asterisk.org/r/776/ ........ + +2010-07-26 19:58 +0000 [r279567] Paul Belanger + + * /, CHANGES, UPGRADE-1.8.txt, configs/logger.conf.sample: Merged + revisions 279566 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279566 | pabelanger | 2010-07-26 15:51:39 -0400 (Mon, 26 Jul + 2010) | 8 lines Add documentation for FAX logger level. (closes + issue #17715) Reported by: vrban Patches: 17715.patch uploaded by + pabelanger (license 224) Tested by: vrban ........ + +2010-07-26 19:20 +0000 [r279564] Tilghman Lesher + + * /, sounds/Makefile.380 (added), configure, + include/asterisk/autoconfig.h.in, sounds/Makefile.381 (added), + configure.ac, sounds/Makefile (removed): Merged revisions 279562 + via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279562 | tilghman | 2010-07-26 14:18:26 -0500 + (Mon, 26 Jul 2010) | 9 lines Merged revisions 279561 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ........ r279561 | tilghman | 2010-07-26 14:15:59 -0500 (Mon, 26 + Jul 2010) | 2 lines Use a special Makefile for noobs who still + have GNU Make 3.80. ........ ................ + +2010-07-26 16:44 +0000 [r279533] Mark Michelson + + * /, configure, include/asterisk/autoconfig.h.in, configure.ac, + channels/sip/reqresp_parser.c: Merged revisions 279504 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279504 | mmichelson | 2010-07-26 11:04:09 -0500 (Mon, 26 Jul + 2010) | 14 lines Allow for systems without locale support to be + usable. A recent change to SIP URI comparison code added a + locale-specific string comparison to the mix, and certain systems + do not support such functions. This fix allows for those systems + to still use Asterisk 1.8 (closes issue #17697) Reported by: + pprindeville Patches: asterisk-trunk-bugid17697.patch uploaded by + pprindeville (license 347) Tested by: mmichelson ........ + +2010-07-26 15:44 +0000 [r279503] Sean Bright + + * /, autoconf/ast_ext_lib.m4: Merged revisions 279502 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279502 | seanbright | 2010-07-26 11:43:54 -0400 + (Mon, 26 Jul 2010) | 12 lines Merged revisions 279501 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ + r279501 | seanbright | 2010-07-26 11:41:13 -0400 (Mon, 26 Jul + 2010) | 5 lines Expand the correct value within AST_OPTION_ONLY. + (closes issue #17703) Reported by: stuarth ........ + ................ + +2010-07-26 03:28 +0000 [r279473] Tilghman Lesher + + * /, formats/format_sln.c, formats/format_wav.c, + formats/format_ogg_vorbis.c, formats/format_wav_gsm.c, + formats/format_sln16.c, formats/format_siren7.c, + formats/format_ilbc.c, formats/format_vox.c, + formats/format_pcm.c, formats/format_h263.c, + formats/format_g723.c, formats/format_h264.c, + formats/format_siren14.c, formats/format_jpeg.c, + formats/format_g726.c, formats/format_gsm.c, + formats/format_g719.c, formats/format_g729.c: Merged revisions + 279472 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279472 | tilghman | 2010-07-25 22:27:06 -0500 (Sun, 25 Jul 2010) + | 2 lines Formats need to load before apps, because some apps + call ast_format_str_reduce() at load time. ........ + +2010-07-25 21:28 +0000 [r279443] Paul Belanger + + * /, tests/test_func_file.c: Merged revisions 279442 via svnmerge + from https://origsvn.digium.com/svn/asterisk/branches/1.8 + ........ r279442 | pabelanger | 2010-07-25 17:26:42 -0400 (Sun, + 25 Jul 2010) | 2 lines Add trailing backslash to silence warning + message. ........ + +2010-07-25 18:22 +0000 [r279391-279413] Tilghman Lesher + + * /, cdr/cdr_odbc.c: Merged revisions 279410 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279410 | tilghman | 2010-07-25 13:21:27 -0500 (Sun, 25 Jul 2010) + | 8 lines Don't re-register CDR module on reload. (closes issue + #17304) Reported by: jnemeth Patches: + 20100507__issue17304.diff.txt uploaded by tilghman (license 14) + Tested by: jnemeth ........ + + * /, main/logger.c: Merged revisions 279390 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279390 | tilghman | 2010-07-25 12:32:21 -0500 (Sun, 25 Jul 2010) + | 8 lines Don't assume qlog is open. (closes issue #17704) + Reported by: vrban Patches: issue17704.patch uploaded by + pabelanger (license 224) Tested by: vrban ........ + +2010-07-24 20:49 +0000 [r279274-279315] Paul Belanger + + * Makefile, /: Merged revisions 279314 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279314 | pabelanger | 2010-07-24 16:47:52 -0400 (Sat, 24 Jul + 2010) | 7 lines Remove duplicate -c flag when using $(INSTALL) + (closes issue #17695) Reported by: pabelanger Patches: + Makefile.diff uploaded by pabelanger (license 224) ........ + + * /, include/asterisk/netsock2.h: Merged revisions 279280 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279280 | pabelanger | 2010-07-24 14:18:43 -0400 (Sat, 24 Jul + 2010) | 8 lines Check if ast_sockaddr is NULL then return. + (closes issue #17677) Reported by: outcast Patches: + issue0017677.patch uploaded by pabelanger (license 224) Tested + by: elguero ........ + + * main/manager.c, /: Merged revisions 279273 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279273 | pabelanger | 2010-07-24 13:36:42 -0400 (Sat, 24 Jul + 2010) | 6 lines Default sin_family to AF_INET for TCP / TLS + Bindaddress. Otherwise, 'manager show settings' will generate + errors if manager is not enabled. ........ + +2010-07-23 22:24 +0000 [r279156-279245] Richard Mudgett + + * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 279227 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 + ................ r279227 | rmudgett | 2010-07-23 17:20:47 -0500 + (Fri, 23 Jul 2010) | 21 lines Merged revisions 279207 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.6.2 + ................ r279207 | rmudgett | 2010-07-23 17:11:23 -0500 + (Fri, 23 Jul 2010) | 14 lines Merged revisions 279206 via + svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ + r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) + | 7 lines SIP promiscuous redirect could fail to dial the + redirect. The ast_channel was created with one variable to + ast_request() but the call to ast_call() that initiates the + outgoing call was using a different variable. The two variables + are not equivalent if the call_forward string included a channel + technology specifier. e.g., SIP/200 ........ ................ + ................ + + * channels/chan_dahdi.c: Make "dahdi show channels" show an + outgoing called number. The "dahdi show channels" extension + column previously only showed the called number of an incoming + call. It now shows the called number for an outgoing call as + well. (closes issue #17653) Reported by: amazinzay Patches: + issue17653_trunk.txt uploaded by rmudgett (license 664) + +2010-07-23 19:17 +0000 [r279116-279118] Russell Bryant + + * UPGRADE.txt, UPGRADE-1.8.txt (added): Shuffle UPGRADE.txt files + for 1.10. + + * CHANGES: Start a new section in CHANGES for 1.10. + +2010-07-23 18:56 +0000 [r279115] Tilghman Lesher + + * /, res/res_odbc.c: Merged revisions 279113 via svnmerge from + https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ + r279113 | tilghman | 2010-07-23 13:56:04 -0500 (Fri, 23 Jul 2010) + | 2 lines Silly 64-bit compilers (who uses 64-bit anyway?) + ........ + +2010-07-23 18:22 +0000 [r279063-279084] Russell Bryant + + * /: Remove old properties. + + * /: Add branch-1.8-merged and branch-1.8-blocked properties to + trunk. + -- 2.47.3