asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	43bcf51204	utils dir: Remove no longer needed traces of refcounter except in the clean make target. * Removed no longer needed files from the svn:ignore property to make them visible. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412213 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	d6e2c50058	bridging: Ensure locking during snapshot creation While the vast majority of bridge snapshot creation is locked properly, there are currently some instances that are not. This adds the missing locking to ensure bridge state is not malleable during snapshot creation. (closes issue ASTERISK-22904) Review: https://reviewboard.asterisk.org/r/3415/ Reported by: Matt Jordan ........ Merged revisions 412193 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Olle Johansson	f65dd23bf4	Formatting: Remove invisible characters git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Olle Johansson	2a4205df2c	Formatting only. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	4f30c7e91f	main/astobj2: Make REF_DEBUG a menuselect item; improve REF_DEBUG output This patch does the following: (1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables REF_DEBUG globally throughout Asterisk. (2) The ref debug log file is now created in the AST_LOG_DIR directory. Every run will now blow away the previous run (as large ref files sometimes caused issues). We now also no longer open/close the file on each write, instead relying on fflush to make sure data gets written to the file (in case the ao2 call being performed is about to cause a crash) (3) It goes with a comma delineated format for the ref debug file. This makes parsing much easier. This also now includes the thread ID of the thread that caused ref change. (4) A new python script instead for refcounting has been added in the contrib/scripts folder. (5) The old refcounter implementation in utils/ has been removed. Review: https://reviewboard.asterisk.org/r/3377/ ........ Merged revisions 412114 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412115 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412153 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	af95d8c1d2	monitor: use app options parsing helper code This app is pretty ancient, so it was never converted to use the option parsing helper code. I'd like to add an option to this app that takes an argument, and that's a pain to do when not using this helper, so start by doing this conversion. Review: https://reviewboard.asterisk.org/r/3429/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0c1342bd2b	res_hep_pjsip: Use the channel name instead of the call ID when it is available During discussions with Alexandr Dubovikov at Kamailio World, it became apparent that while the SIP call ID is a useful identifier prior to an Asterisk channel being created, it is far more preferable to use the channel name (or some channel based identifier) when the channel is available. Homer is smart enough to tie the various messages together. This patch opts to use the channel name when it is available, falling back to the call ID otherwise. ........ Merged revisions 412088 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	6905ac0f5e	res_pjsip_pubsub: Set the body generation result to 0 for a valid path The result of the "ast_sip_pubsub_generate_body_content" was not set/initialized. Consequently, the nominal path potentially returned an invalid value, thus not sending mwi notifications. ........ Merged revisions 412074 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412075 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	755696dcd0	Add a Command header to the AMI Mixmonitor action. This fixes a parsing error that occurred during the processing of the AMI action. The error did not result in MixMonitor itself misbehaving, but it could result in the AMI response not giving correct information back. The new header allows for one to specify a post-process command to run when recording finishes. Previously, in order to do this, the post-process command would have to be placed at the end of the Options: header. Patches: mixmonitor_command_2.patch by jhardin (License #6512) ........ Merged revisions 412048 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	c613ef6b30	res_stasis_answer: Add missing newlines ........ Merged revisions 412034 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412035 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	5ca5d42646	Internal timing: Add notice that the -I and internal_timing option are no longer needed. Add notice messages during execution that the -I command line option and the astersik.conf internal_timing option are no longer needed. The internal timing functionality is now always enabled if there is a timing module loaded. NOTE: Since the command line options and the asterisk.conf config file are processed before the logging system is initialized, the messages are output to stderr. Change requested as a result of asterisk-dev list comments about the commit for ASTERISK-22846 that removed the -I and internal_timing options. Review: https://reviewboard.asterisk.org/r/3423/ ........ Merged revisions 411964 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411974 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411985 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	551b2d1183	config: Fix CB_ADD_LEN() to work as originally intended. Fix a long standing bug in CB_ADD_LEN() behaving like CB_ADD(). ASTERISK-23546 #close Reported by: Walter Doekes ........ Merged revisions 411960 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411961 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411962 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	158bd5dd74	app_confbridge: Fix confbridge.conf dsp_talking_threshold option setting wrong parameter. Fixed copy pasta error. ASTERISK-23545 #close Reported by: John Knott ........ Merged revisions 411944 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411945 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	909f835066	res_pjsip: Ignore explicit transport configuration if a WebSocket transport is specified. This change makes it so if a transport is configured on an endpoint that is a WebSocket type the option will be ignored. In practice this is fine because the WebSocket transport can not create outgoing connections, it can only reuse existing ones. By ignoring the option the existing PJSIP logic for using the existing connection will be invoked and stuff will proceed. (closes issue ASTERISK-23584) Reported by: Rusty Newton ........ Merged revisions 411927 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411928 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	607c6e9772	func_periodic_hook: List more modules as dependencies This module makes use of some existing Asterisk components. app_chanspy was already listed as a dependency. There are a few function modules used, as well, so list them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411897 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	fcb04d889a	PJSIP: Ensure test event has new state The change that fixed the pubsub test event's use of a dangling pointer also changed when it was processed relative to the pjsip subscription state change processing. This change corrects the order of events while holding a reference to the pointer that was previously dangling. ........ Merged revisions 411883 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	c0a812e143	AGI/Manager: Prevent multiple NewExten events during AGI application changes AGI applications would trigger NewExten events every time the state of the AGI application changed. This has historically not been the behavior and this behavior was introduced with a CDR patch. This patch corrects that. (closes issue ASTERISK-23390) Reported by: Benjamin Keith Ford Review: https://reviewboard.asterisk.org/r/3406/ ........ Merged revisions 411868 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	76d5c4ed43	app_queue: Re-add HoldTime to QueueCallerAbandon event (simple typo during ast12 refactor). Reported by: Ibrahim22 (on IRC) Tested by: Ibrahim22 ........ Merged revisions 411811 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	a625471bf1	Blocked revisions 411809 ........ configs: Clean up long line and typo in res_odbc.conf.sample. ........ Merged revisions 411807 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411808 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	62e2bf68f0	Stasis: Fix Stasis() bridge refcount issue The Stasis() dialplan application monitors what bridge a channel is in and so necessarily holds on to a bridge pointer. This change ensures that it also holds on to a reference for that bridge to prevent the bridge pointer from becoming a dangling pointer. ........ Merged revisions 411804 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	5d9a1281ee	PJSIP: Fix crash introduced in r411671 The test event introduced in revision 411671 uses a dangling pointer to access information about pubsub state changes. This moves the event to within the lifetime of the pointer. ........ Merged revisions 411790 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Russell Bryant	ea290b2c3e	func_periodic_hook: New function for periodic hooks. This commit introduces a new dialplan function, PERIODIC_HOOK(). It allows you run to a dialplan hook on a channel periodically. The original use case that inspired this was the ability to play a beep periodically into a call being recorded. The implementation is much more generic though and could be used for many other things. The implementation makes heavy use of existing Asterisk components. It uses a combination of Local channels and ChanSpy() to run some custom dialplan and inject any audio it generates into an active call. The other important bit of the implementation is how it figures out when to trigger the beep playback. This implementation uses the audiohook API, even though it's not actually touching the audio in any way. It's a convenient way to get a callback and check if it's time to kick off another beep. It would be nice if this was timer event based instead of polling based, but unfortunately I don't see a way to do it that won't interfere with other things. Review: https://reviewboard.asterisk.org/r/3362/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	03beadb6e9	internal_timing: Remove the option and always make it enabled if a timing module is loaded. The masquerade supertest frequently fails because either the local channel chain doesn't completely optimize out or the DTMF handshake doesn't completely get accross. Local channel optimization requires frames flowing to trigger when optimization can happen. When optimization happens the media frame that triggered the optimization is dropped. Sending DTMF requires frames to flow in the other direction for timing purposes while sending nothing. If internal timing is not enabled when MOH is playing, Asterisk switches to received timing when an audio frame is received. With optimization dropping media frames and MOH not sending frames unless it receives frames, occasionaly there are no more frames being passed and the test fails. * The asterisk command line -I option and the asterisk.conf internal_timing option are removed. Asterisk now always uses internal timing when needed if any timing module is loaded. The issue ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken if other internal timing modules besides DAHDI are used. The ast_read_generator_actions() now only does received timing if it has no choice for frame generators like MOH, silence, and playback streaming. * Cleaned up some code dealing with frame generators in ast_deactivate_generator(), generator_write_format_change(), ast_activate_generator(), and ast_channel_stop_silence_generator(). * Removed ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and ast_opt_internal_timing. ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........ Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411716 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411717 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9be438299d	Add some asserts that were handy when looking for a stasis cache problem. * Assert if a channel is destroyed but has the snapshot staging flag set. In this case the final channel destruction snapshot would never get taken. * Assert if what we just got out of the stasis cache is not what we were looking for. This assert would have saved several days searching for a bug and a lot of my hair. * Assert if the music on hold message posts could not find the associated channel. A crash will happen later when manager tries to send the MOH AMI message. This assert catches the problem when the stasis message is posted instead of by the thread processing the defective message. * Always generate a backtrace when an ast_assert() fails. Review: https://reviewboard.asterisk.org/r/3411/ ........ Merged revisions 411701 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	73f337d97b	http: Fix spurious ERROR message in responses with no content When a response has a content length of 0, fwrite would be called to write a buffer with no data in it. This resulted in the following classic error message: [Apr 3 11:49:17] ERROR[26421] http.c: fwrite() failed: Success This patch makes it so that we only attempt to write out the content if the calculated content_length is non-zero. ........ Merged revisions 411687 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	045285f8e3	res_pjsip_pubsub: Add test event for state change This adds a test event when subscription state changes so that integration tests may trigger new actions at the appropriate times. Review: https://reviewboard.asterisk.org/r/3383/ ........ Merged revisions 411670 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	db5bd60c2a	res_hep: Fix crash when hep.conf not available Parts of res_hep properly checked for a valid configuration object before attempting to access the configuration. A check, however, was missed when a packet is sent. This patch fixes the crash caused by not checking if the configuration object is valid. ........ Merged revisions 411668 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411669 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	eefcb79bfb	Prevent duplicate sorcery wizards from being applied to sorcery object types. This commit contains several changes to sorcery: 1) Application of sorcery configuration based on module name is automatically performed when sorcery is opened for a module. 2) Sorcery will not attempt to apply the same wizard to an object type more than once. 3) Sorcery gives more exact results when attempting to apply a wizard, whether as the default or based on configuration. Sorcery unit tests still pass for me after making these changes. Review: https://reviewboard.asterisk.org/r/3326 ........ Merged revisions 411159 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c704795dcb	res_parking: Minor tweaks. * Use ast_bridge_channel_lock()/ast_bridge_channel_unlock() instead of ao2_lock()/ao2_unlock() for struct ast_bridge_channel variables. * Use ast_copy_string() instead of inlining it. * Remove an already done TODO comment. * Some whitespace tweaks. ........ Merged revisions 411638 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411639 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	7542361f4a	stasis_channels.c: Eliminate another overuse of RAII_VAR(). ........ Merged revisions 411636 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	c7b8633c26	app_queue: Fix a bug where realtime members would be deleted during reload causing waiting callers to get ejected. This patch causes realtime queue members to remain in queues during the reload process. Previously these members would be removed causing any waiting callers to be ejected from the queue with a reason of "EXITEMPTY". ASTERISK-23547 #close ASTERISK-23547 #comment Patch app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo Rossi (license 6409) Review: https://reviewboard.asterisk.org/r/3404/ ........ Merged revisions 411584 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411585 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411586 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ef0c9fe4d8	res_hep/res_hep_pjsip: Add a HEPv3 capture agent module and a logger for PJSIP This patch adds the following: (1) A new module, res_hep, which implements a generic packet capture agent for the Homer Encapsulation Protocol (HEP) version 3. Note that this code is based on a patch provided by Alexandr Dubovikov; I basically just wrapped it up, added configuration via the configuration framework, and threw in a taskprocessor. (2) A new module, res_hep_pjsip, which forwards all SIP message traffic that passes through the res_pjsip stack over to res_hep for encapsulation and transmission to a HEPv3 capture server. Much thanks to Alexandr for his Asterisk patch for this code and for a lot of patience waiting for me to port it to 12/trunk. Due to some dithering on my part, this has taken the better part of a year to port forward (I still blame CDRs for the delay). ASTERISK-23557 #close Review: https://reviewboard.asterisk.org/r/3207/ ........ Merged revisions 411534 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411556 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Alexandr Anikin	a35ce3924b	process stack command even if gatekeeper client isn't register don't destroy gatekeeper client if it is not started don't destroy gatekeeper client in some sort of gatekeeper errors signal rtp create condition when call cleared before rtp structure created (closes issue ASTERISK-23460) Reported by: Dmitry Melekhov Patches: ASTERISK-23460-2.patch Tested by: Dmitry Melekhov ........ Merged revisions 411531 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411532 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	597f25db69	Update API versions and UPGRADE/CHANGES for 12.2.0 This patch does the following: * It updates the AMI version to 2.2.0 to indicate backwards compatible changes have been made since the last release * It updates the ARI version to 1.2.0 to indicate backwards compatible changes have been made since the last release * It updates the UPGRADE/CHANGES files with changes that were not mentioned ........ Merged revisions 411529 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a438a0e65f	res_config_odbc: Fix for nullable integer columns and keyfield existence check in update_odbc. This patch fixes setting nullable integer columns to NULL instead of an empty string, which fails for PostgreSQL, for example. The current code is supposed to do so, but the check is broken. The patch also allows the first column in the list to be a nullable integer. Also, the check for existence of a mandatory column checked for the first column in the list instead of the key field lookup column. This patch fixes that issue as well. Finally, the compatibility option allow_empty_string_in_nontext, which was added to previous revisions to allow for some database backends with certain schemas to function, has been removed. Review: https://reviewboard.asterisk.org/r/3335 ASTERISK-23459 #close ASTERISK-23351 #close (closes issue ASTERISK-23459) Reported by: zvision patches: res_config_odbc.diff uploaded by zvision (License 5755) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411515 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	24452b852d	Blocked revisions 411512 ........ res_config_odbc/res_odbc: Fix handling of non-text columns updates with empty values. This patch fixes setting nullable integer columns to NULL instead of an empty string, which fails for PostgreSQL, for example. The current code is supposed to do so, but the check is broken. The patch also allows the first column in the list to be a nullable integer. This patch also adds a compatibility setting in res_odbc.conf, allow_empty_string_in_nontext. It is enabled by default. It should be disabled for database backends (such as PostgreSQL) that require NULL instead of an empty string for Integer columns. Review: https://reviewboard.asterisk.org/r/3375 (issue ASTERISK-23459) Reported by: zvision patches: res_config_odbc.diff uploaded by zvision (License 5755) ........ Merged revisions 411399 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411408 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	0d057e6791	http: response body often missing after specific request This patch works around a problem with the HTTP body being dropped from the response to a specific client and under specific circumstances: a) Client request comes from node.js user agent "Shred" via use of swagger-client library. b) Asterisk and Client are not on the same host or TCP/IP stack In testing this problem, it has been determined that the write of the HTTP body is lost, even if the data is written using low level write function. The only solution found is to instruct the TCP stack with the shutdown function to flush the last write and finish the transmission. See review for more details. ASTERISK-23548 #close (closes issue ASTERISK-23548) Reported by: Sam Galarneau Review: https://reviewboard.asterisk.org/r/3402/ ........ Merged revisions 411462 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411463 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411465 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	2cfa9b4b77	UPGRADE: Note IAX2 compatibility issue between 1.4 and 1.8+ systems. ........ Merged revisions 411457 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411458 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411459 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411460 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	2dae3d6ea3	contrib/realtime: Remove empty SQL script files Since the relatime scripts are now managed by Alembic, the previous realtime scripts were previously removed. However, the removal process messed up, as the files were still in the repository. The contents were just empty. This removes the files from the tree. ........ Merged revisions 411442 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	562182f5b0	chan_sip: Add MESSAGE request to allowed methods The allowed methods advertised by chan_sip did not previously note the MESSAGE request. Even in Asterisk 1.8, we do accept in-dialog MESSAGE requests; we should advertise that we support MESSAGE requests. ASTERISK-23504 #close ASTERISK-23504 #comment Reported by: Martin Kontsek ASTERISK-23504 #comment Patch sip.h_patch.diff uploaded by Martin Kontsek (license 6587) Review: https://reviewboard.asterisk.org/r/3396/ ........ Merged revisions 411372 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411373 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411374 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	fbe0dfaf44	Fix dialplan function NULL channel safety issues (closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	44409401ec	main/formats: Fix crash in ast_format_cmp during non-clean shutdown. * Update asterisk.h to reflect availability of ast_register_cleanup in 11.9. * Use ast_register_cleanup for format_attr_shutdown. (closes issue ASTERISK-23103) Reported by: JoshE ........ Merged revisions 411310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	a8629e53c1	Give sorcery instances a reference to their wizards. On graceful shutdown, sorcery wizards are all killed off, but it is possible for sorcery instances to still have dangling pointers after this, possibly causing a crash. Giving the sorcery instances a reference to their wizards ensures that the wizard reference will remain valid for the lifetime of the sorcery instance. Review: https://reviewboard.asterisk.org/r/3401 ........ Merged revisions 411295 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	7dddd694cb	say: Fix a bug where SayNumber in Polish tries to play incorrect sound. This change fixes a bug where calling SayNumber with a number divisible by 100 using the Polish language would cause the code to attempt to play a sound file with an empty name. (closes issue ASTERISK-23509) Reported by: zvision Review: https://reviewboard.asterisk.org/r/3378/ ........ Merged revisions 411243 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411244 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411245 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	fa3a2f8eca	chan_sip: Send real CallerID information with P-Assserted-Identity (RFC-3325) Prior too this patch, the P-Asserted-Identity header would include anonymous caller id information which seems to go against the point of the P-Asserted-Identity header. Now the real caller ID information will be included in this header. Also, no privacy header would be included. This patch adds 'Privacy: id' to outgoing SIP messages that include the P-Asserted-Identity header. (closes issue AST-1301) ........ Merged revisions 411189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411190 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411193 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4b18b3bb4d	Fix 'alembic branches' merge conflict as described by the web page. ........ Merged revisions 411191 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	c32fe8b8e5	ARI: Don't complain about missing ARI users when we aren't enabled Currently, if ARI is not enabled it will still complain that there are no configured users. This patch checks to see if ARI is enabled before logging and error or iterating the container to validate the users. Review: https://reviewboard.asterisk.org/r/3391/ ........ Merged revisions 411173 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2bf37a417d	Add a "message_context" option for PJSIP endpoints. ........ Merged revisions 411157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c1c8300e27	res_pjsip: Fix contact authenticate_qualify endpoint lookup when qualifing a contact. * Fixed bad use of ao2_find() in on_endpoint(). * Replaced use of find_endpoints() with find_an_endpoint() since only the first found endpoint is ever needed. * Fixed qualify_contact_cb() to update the contact with the aor authenticate_qualify setting. Otherwise, permanent contacts in the aor type sections would have a config line order dependancy. * Fixed off nominal path contact ref leak in qualify_contact(). The comment saying the unref is not needed was wrong. * Fixed off nominal path use of the endpoint parameter if it is NULL in send_out_of_dialog_request(). * Added missing off nominal path unref of pjsip tdata in send_out_of_dialog_request(). * Fixed off nominal path failing to call the callback in send_request_cb() when the request is challenged for authentication. * Eliminated silly RAII_VAR() use in qualify_contact_cb(). * Updated ast_sip_send_request() doxygen to better reflect reality. (closes issue ASTERISK-23254) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/3381/ ........ Merged revisions 411141 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411142 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	a4890eddfb	chan_sip: Fix incorrect use of timers If update_provisional_keepalive() is called while send_provisional_keepalive_full() is waiting on the PVT lock, then pvt->provisional_keepalive_sched_id will be changed to a new sched_id value by update_provisional_keepalive(), but that new sched_id then may be overwritten with -1 by send_provisional_keepalive_full(), killing the pvt's reference to a schedule and "leaking" the reference. (closes issue ASTERISK-22079) Review: https://reviewboard.asterisk.org/r/3368/ Reported by: Jamuel Starkey, Matteo, Leif Madsen, Steve Davies Patches: provisional_keepalive_fix.diff uploaded by Steve Davies (license 5012) ........ Merged revisions 411088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411089 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411091 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago

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