asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	7d9ba4875b	Fix calls to ast_get_ip() not initializing the address family. ........ Merged revisions 346239 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346240 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	b7aee9ebc9	Fix ast_str_truncate signedness warning and documentation. Review: https://reviewboard.asterisk.org/r/1594 ........ Merged revisions 346144 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346145 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	e6ca768081	Fix res_jabber resource leaks This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 ........ Merged revisions 346086 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 346087 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2b3e28f88c	Fix dnsmgr entries to ask for the same address family each time. The dnsmgr refresh would always get the first address found regardless of the original address family requested. So if you asked for only IPv4 addresses originally, you might get an IPv6 address on refresh. * Saved the original address family requested by ast_dnsmgr_lookup() to be used when the address is refreshed. ........ Merged revisions 345976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345977 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	d777e15792	Clarify why the AST_LOG_* macros exist next to the LOG_* macros. (issue ASTERISK-17973) ........ Merged revisions 345923 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345924 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	735e48f92f	Use __alignof__ instead of sizeof for stringfield length storage. Kevin P Fleming suggested that r343157 should use __alignof__ instead of sizeof. For most systems this won't be an issue, but better fix it now while it's still fresh. Review: https://reviewboard.asterisk.org/r/1573 ........ Merged revisions 344843 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344845 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	816dc295c2	Formatting and doxygen improvements git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	7fd2a68b0e	Ensure that string field lengths are properly aligned Integers should always be aligned. For some platforms (ARM, SPARC) this is more important than for others. This changeset ensures that the string field string lengths are aligned on all platforms, not just on the SPARC for which there was a workaround. It also fixes that the length integer can be resized to 32 bits without problems if needed. (closes issue ASTERISK-17310) Reported by: radael, S Adrian Reviewed by: Tzafrir Cohen, Terry Wilson Tested by: S Adrian Review: https://reviewboard.asterisk.org/r/1549 ........ Merged revisions 343157 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343158 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	b41b49ea0e	Several fixes to the chan_sip dynamic realtime peer/user lookup There were several problems with the dynamic realtime peer/user lookup code. The lookup logic had become rather hard to read due to lots of incremental changes to the realtime_peer function. And, during the addition of the sipregs functionality, several possibilities for memory leaks had been introduced. The insecure=port matching has always been broken for anyone using the sipregs family. And, related, the broken implementation forced those using sipregs to still have an ipaddr column on their sippeers table. Thanks Terry Wilson for comprehensive testing and finding and fixing unexpected behaviour from the multientry realtime call which caused the realtime_peer to have a completely unused code path. This changeset fixes the leaks, the lookup inconsistenties and that you won't need an ipaddr column on your sippeers table anymore (when you're using sipregs). Beware that when you're using sipregs, peers with insecure=port will now start matching! (closes issue ASTERISK-17792) (closes issue ASTERISK-18356) Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1395 ........ Merged revisions 342927 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342929 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	85dbd68f87	Whitespace and some better macro variable names. * Renamed AST_LIST_TRAVERSE_SAFE_BEGIN __new_prev to __list_current. * Renamed AST_LIST_MOVE_CURRENT __list_cur to __extracted. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f7ce570c35	Fix AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable. AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an iteration or before AST_LIST_REMOVE_CURRENT() without corrupting the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the list if AST_LIST_INSERT_BEFORE_CURRENT() or AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed cut and paste error using the wrong variable in AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and AST_LIST_INSERT_LIST_AFTER(). ........ Merged revisions 342661 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342662 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	700f0a2752	Return NULL when no results returned for realtime_multientry It was not documented what the return value should be when no entries were returned with the multientry realtime callback. This change forces consistent behavior even if the backends return an empty ast_config. Review: https://reviewboard.asterisk.org/r/1521/ ........ Merged revisions 342223 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342224 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	41ade83749	Fix ao2obj.h comment typos and add missing link/unlink nolock debug defines. ........ Merged revisions 342183 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342184 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b961d57c4c	Fix AGI exec Park to honor the Park application parameters. The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) ........ Merged revisions 341717 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341718 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	2f1130e13f	Clean up ast_check_digits The code was originally copied from the is_int() function in the AEL code. wdoekes pointed out that the function should take a const char* and that their was an unneeded variable. This is now fixed. ........ Merged revisions 341529 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341530 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	5f8648892f	Don't use is_int() since it doesn't link well on all platforms Just create an normal API function in strings.h that does the same thing just to be safe. ASTERISK-17146 ........ Merged revisions 341379 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341380 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	10de040b6e	More parking issues. * Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. ........ Merged revisions 341254 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 341255 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	1ec8a9d896	Update SHA1 code to RFC 6234 RFC 6234 is an update to RFC 3174 from which the code was originally taken. It has a slightly better code, and a better phrased license (simple 3-clause BSD). * main/sha1.c is sha1.c from RFC 6234 with formatting changes only. * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234. * Removed unused include of asterisk/sha1.h from main/channels.c Review: https://reviewboard.asterisk.org/r/1503/ Merge-From: http://svn.asterisk.org/svn/asterisk/branches/1.8@340263 Merge-From: http://svn.asterisk.org/svn/asterisk/branches/10@340280 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340283 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	067250f74c	Convert registered AMI actions to ao2 objects. * Fixed race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential memory leak if an AMI action failed to get registered because is already was registered. Part of the ao2 conversion. * Fixed AMI ListCommands action not walking the actions list with a lock held. * Fix usage of ast_strdupa() and alloca() in loops. Excess stack usage. * Fix AMI Originate action Variable header requiring a space after the header colon. Reported by Yaroslav Panych on the asterisk-dev list. * Increased the number of listed variables allowed per AMI Originate action Variable header to 64. * Fixed AMI GetConfigJSON action output format. * Fixed usage of res contents outside of scope in append_channel_vars(). * Fixed inconsistency of config file channelvars option. The values no longer accumulate with every channelvars option in the config file. Only the last value is kept to be consistent with the CLI "manager show settings" command. (closes issue ASTERISK-18479) Reported by: Jaco Kroon ........ Merged revisions 340279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 340281 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340282 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	bb07ca66a1	Merged revisions 340109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r340109 \| mnicholson \| 2011-10-10 09:15:41 -0500 (Mon, 10 Oct 2011) \| 18 lines Merged revisions 340108 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r340108 \| mnicholson \| 2011-10-10 09:14:48 -0500 (Mon, 10 Oct 2011) \| 11 lines Load the proper XML documentation when multiple modules document the same application. This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@340110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	6e5f97df77	Merged revisions 339720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339720 \| rmudgett \| 2011-10-06 17:58:40 -0500 (Thu, 06 Oct 2011) \| 27 lines Merged revisions 339719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339719 \| rmudgett \| 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) \| 20 lines Fix regression in configure script for libpri capability checks. JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer 2 persistence issues with some telcos. ASTERISK-18535 attempted to fix the unexpected requirement that libpri must have that feature to work with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI optional features required. Unfortunately, I thought AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and deleted those lines for libpri. The result was the HAVE_PRI_xxx defines that control the ability to use optional libpri features were also deleted. * Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional features in a library that the source code could take advantage of if the code supports the feature. (closes issue ASTERISK-18687) Reported by: Norbert Tested by: rmudgett ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	b698038995	Add generic faxdetect framehook to res_fax Added func FAXOPT(faxdetect)=yes,cng,t38[,timeout]/no to enable dialplan faxdetect allowing more flexibility. as soon as a fax tone is detected the framehook is removed. there is a penalty involved in running this framehook on non G711 channels as they will be transcoded. CNG tone is suppresed using the SQUELCH flag to allow WaitForNoise to be run on the channel to detect Voice. (Closes issue ASTERISK-18569) Reported by: Myself Reviewed by: Matthew Nicholson, Kevin Fleming Review: https://reviewboard.asterisk.org/r/1116/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	e9736c586f	Merged revisions 338253 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r338253 \| rmudgett \| 2011-09-28 16:22:05 -0500 (Wed, 28 Sep 2011) \| 14 lines Merged revisions 338235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r338235 \| rmudgett \| 2011-09-28 16:17:45 -0500 (Wed, 28 Sep 2011) \| 7 lines Fix inconsistency in LOG_VERBOSE/AST_LOG_VERBOSE declaration. (closes issue ASTERISK-17973) Reported by: Luke H Patches: logger_h.patch (license #6278) patch uploaded by Luke H ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	55b70ae625	Merged revisions 337974 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337974 \| rmudgett \| 2011-09-26 14:35:23 -0500 (Mon, 26 Sep 2011) \| 37 lines Merged revisions 337973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 \| rmudgett \| 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) \| 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	5982bdcb7c	Merged revisions 337595,337597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r337595 \| jrose \| 2011-09-22 10:35:50 -0500 (Thu, 22 Sep 2011) \| 12 lines Generate Security events in chan_sip using new Security Events Framework Security Events Framework was added in 1.8 and support was added for AMI to generate events at that time. This patch adds support for chan_sip to generate security events. (closes issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (license #5026) by Michael L. Young Review: https://reviewboard.asterisk.org/r/1362/ ........ r337597 \| jrose \| 2011-09-22 10:47:05 -0500 (Thu, 22 Sep 2011) \| 10 lines Forgot to svn add new files to r337595 Part of Generating security events for chan_sip (issue ASTERISK-18264) Reported by: Michael L. Young Patches: security_events_chan_sip_v4.patch (License #5026) by Michael L. Young Reviewboard: https://reviewboard.asterisk.org/r/1362/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Jordan	e218748ac1	Merged revisions 337120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337120 \| mjordan \| 2011-09-20 17:49:36 -0500 (Tue, 20 Sep 2011) \| 28 lines Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 \| mjordan \| 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) \| 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	5e7121b44f	Merged revisions 336734 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336734 \| tilghman \| 2011-09-19 15:29:40 -0500 (Mon, 19 Sep 2011) \| 18 lines Merged revisions 336733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336733 \| tilghman \| 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011) \| 11 lines Various changes to allow 1.8 to compile on Mac OS X Lion (10.7) * Makefile workaround for 10.6 extended to work on 10.7 and later. * Now uses the 'weak' symbol for Lion systems, which no longer support 'weak_import' Closes ASTERISK-17612. Closes ASTERISK-18213. Tested by: tilghman, oej. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	beae2df26e	Merged revisions 336307 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r336307 \| jrose \| 2011-09-16 16:09:20 -0500 (Fri, 16 Sep 2011) \| 20 lines Merged revisions 336294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r336294 \| jrose \| 2011-09-16 14:53:40 -0500 (Fri, 16 Sep 2011) \| 13 lines Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ae4c13f4f3	Merged revisions 335912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335912 \| rmudgett \| 2011-09-14 13:31:15 -0500 (Wed, 14 Sep 2011) \| 20 lines Merged revisions 335911 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335911 \| rmudgett \| 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) \| 13 lines Remove unnecessary libpri dependency checks in the configure script. Using the --with-pri option with the configure script generated an error about not having PRI_L2_PERSISTENCE if you did not have the absolute latest libpri SVN checkout installed. The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to be for libraries that are dependent upon other libraries and not necessarily for optional/added features within a library. (closes issue ASTERISK-18535) Reported by: Michael Keuter ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	2a25779d47	Merged revisions 335510 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335510 \| russell \| 2011-09-13 02:24:34 -0500 (Tue, 13 Sep 2011) \| 22 lines Merged revisions 335497 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335497 \| russell \| 2011-09-13 02:11:36 -0500 (Tue, 13 Sep 2011) \| 15 lines Fix a crash in res_ais. This patch resolves a crash observed in a load testing environment that involved the use of the res_ais module. I observed some crashes where the event delivery callback would get called, but the length parameter incidcating how much data there was to read was 0. The code assumed (with good reason I would think) that if this callback got called, there was an event available to read. However, if the rare case that there's nothing there, catch it and return instead of blowing up. More specifically, the change always ensure that the size of the received event in the cluster is always big enough to be a real ast_event. Review: https://reviewboard.asterisk.org/r/1423/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	404151ad65	New sip.conf option for setting default tonezone for channel or individual devices Review: https://reviewboard.asterisk.org/r/1429/ (closes issue ASTERISK-18497) Thanks to russellb for peer review. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Jordan	8b5ba33fe0	Merged revisions 335078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 \| mjordan \| 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) \| 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 \| mjordan \| 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) \| 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	220bf14557	Merged revisions 334297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r334297 \| rmudgett \| 2011-09-02 12:15:08 -0500 (Fri, 02 Sep 2011) \| 46 lines Merged revisions 334296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r334296 \| rmudgett \| 2011-09-02 12:10:58 -0500 (Fri, 02 Sep 2011) \| 39 lines Fix potential memory allocation failure crashes in config.c. * Added required checks to the returned memory allocation pointers to prevent crashes. * Made ast_include_rename() create a replacement ast_variable list node if the new filename is longer than the available space. Fixes potential crash and memory leak. * Factored out ast_variable_move() from ast_variable_update() so ast_include_rename() can also use it when creating a replacement ast_variable list node. * Made the filename stuffed at the end of the struct a minimum allocated size in ast_variable_new() in case ast_include_rename() changes the stored filename. * Constify struct char pointers pointing to strings stuffed at the end of the struct for: ast_variable, cache_file_mtime, and ast_config_map. * Factored out cfmtime_new() to remove inlined code and allow some struct pointers to become const. * Removed the list lock from struct cache_file_mtime that was never used. * Added doxygen comments to several structure elements and better documented what strings are stuffed at the struct end char array. * Reworked ast_config_text_file_save() and set_fn() to handle allocation failure of the include file scratch pad object tracking blank lines. * Made ast_config_text_file_save() fn[] declared with PATH_MAX to ensure it is long enough for any filename with path. Also reduced the number of container fileset buckets from a rediculus 180,000 to 1023. JIRA AST-618 Review: https://reviewboard.asterisk.org/r/1378/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@334304 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	d9526bc6c8	Merged revisions 333786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r333786 \| rmudgett \| 2011-08-29 16:12:29 -0500 (Mon, 29 Aug 2011) \| 13 lines Merged revisions 333784-333785 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r333784 \| rmudgett \| 2011-08-29 16:05:43 -0500 (Mon, 29 Aug 2011) \| 2 lines Fix deadlock potential of chan_mobile.c:mbl_ast_hangup(). ........ r333785 \| rmudgett \| 2011-08-29 16:06:16 -0500 (Mon, 29 Aug 2011) \| 1 line Add some do not hold locks notes to channel.h ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@333789 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Jordan	3b53a9cdb3	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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	901e275c4c	Add option for logging congested calls as CONGESTION instead of NO_ANSWER in CDR This patch adds a CDR option to cdr.conf that will allow CDR files to log calls ending with congestion in a way that is unique from other unanswered calls. (closes issue ASTERISK-14842) Reported by: Alec Davis Patches: cdr_congestion.diff.txt (License #5546) patch uploaded by Alec Davis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	91d3a7d3a1	Merged revisions 332756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r332756 \| mnicholson \| 2011-08-22 11:29:45 -0500 (Mon, 22 Aug 2011) \| 4 lines add a way to disable and/or modify the gateway timeout ASTERISK-18219 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	318f0f5514	Merged revisions 332369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332369 \| tilghman \| 2011-08-17 14:24:59 -0500 (Wed, 17 Aug 2011) \| 17 lines Merged revisions 332355 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332355 \| tilghman \| 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) \| 10 lines Re-add support for spaces in pathnames, including now spaces in DESTDIR. This was initially added to 1.8 prior to release, primarily to support the standard paths on Mac OS X, but was partially reverted recently in Subversion, due to the lack of support for spaces in DESTDIR. This commit restores support for the standard paths on Mac OS X, and also includes support for spaces in DESTDIR. (closes issue ASTERISK-18290) Reported by: pabelanger Review: https://reviewboard.asterisk.org/r/1326/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	265102faf8	Merged revisions 332265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r332265 \| rmudgett \| 2011-08-17 11:01:29 -0500 (Wed, 17 Aug 2011) \| 33 lines Merged revisions 332264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332264 \| rmudgett \| 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) \| 26 lines Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards. France Telecom brings layer 2 and layer 1 down on BRI lines when the line is idle. When layer 1 goes down Asterisk cannot make outgoing calls and the HA8 and HB8 cards also get IRQ misses. The inability to make outgoing calls is because the line is in red alarm and Asterisk will not make calls over a line it considers unavailable. The IRQ misses for the HA8 and HB8 card are because the hardware is switching clock sources from the line which just brought layer 1 down to internal timing. There is a DAHDI option for the B410P card to not tell Asterisk that layer 1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards: "modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear up the IRQ misses when the telco brings layer 1 down. * Add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. The new option has three settings: 1) Use libpri default layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer brings it down. 3) Leave layer 2 down when the peer brings it down. Layer 2 will be brought up as needed for outgoing calls. JIRA AST-598 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@332270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	16acfefa74	Merged revisions 331097 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331097 \| twilson \| 2011-08-08 17:59:01 -0500 (Mon, 08 Aug 2011) \| 5 lines Bump the AMI protocol version to 1.2 As a result of converting Unlink events that were missed in the AMI 1.1 update to Bridge events, the AMI protocol version is being incremented. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	6a15e95a32	astobj2: Avoid using temporary objects + ao2_find() with OBJ_POINTER. There is a fairly common pattern making its way through the code base where we put a temporary object on the stack so we can call ao2_find() with OBJ_POINTER. The purpose is so that it can be passed into the object hash function. However, this really seems like a hack and potentially error prone. This patch is a first stab at approach to avoid having to do that. It adds a new flag, OBJ_KEY, which can be used instead of OBJ_POINTER in these situations. Then, the hash function can know whether it was given an object or some custom data to hash. The patch also changes some uses of ao2_find() for iax2_user and iax2_peer objects to reflect how OBJ_KEY would be used. So long, and thanks for all the fish. Review: https://reviewboard.asterisk.org/r/1184/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	d170e5e829	reverting 329840 due to failing tests. Going to change this feature to be purely optional. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	3ee80d6a90	Adds cdr logging of calls resulting in CONGESTION Applies a patch made a long time ago by alecdavis which adds a CDR feature for logging calls that failed due to congestion. (closes issue #15907) Reported by: alecdavis Patches: cdr_congestion.diff.txt uploaded by alecdavis (license #5546) Review: https://reviewboard.asterisk.org/r/454/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	462e0fe530	Merged revisions 329528 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 \| jrose \| 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) \| 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 \| jrose \| 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) \| 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Gregory Nietsky	3b1cc6de8d	dsp_process was enhanced to work with alaw and ulaw in addition to slin.	15 years ago
Russell Bryant	f243d129c9	Merged revisions 329257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329257 \| russell \| 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) \| 2 lines s/1.10/10.0/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kinsey Moore	1dc97eb69b	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/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	c26bb50cc3	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/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	145c174565	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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	3f44b08b7b	do v21 detection instead of CED detection for the fax gateway git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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2345 Commits (edaf970c3855d2eb805ad103bf95248d95a8e299)