4822 Commits (e9520dbe0db9a4f3c5c2cd90a9740880b18d1e90)

Author	SHA1	Message	Date
Walter Doekes	49cbfa7de6	Fix typo's (retrieve, specified, address). ... ........ Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430998 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9bff4eeca3	Bridge core: Pass a ref with the swap channel when joining a bridge. ... When code imparts a channel into a bridge to swap with another channel, a ref needs to be held on the swap channel to ensure that it cannot dissapear before finding it in the bridge. * The ast_bridge_join() swap channel parameter now always steals a ref for the swap channel. This is the only change to the bridge framework's public API semantics. * bridge_channel_internal_join() now requires the bridge_channel->swap channel to pass in a ref. ASTERISK-24649 Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4354/ ........ Merged revisions 430975 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	228fdb3f4e	main/rtp_engine: Format NTP timestamps as unsigned longs ... When the RTCP reports are created, the NTP timestamps are stored as strings, as JSON does not have an integer type long enough to store the value. However, on 32-bit systems, a signed long may overflow for some portion of the timestamp. This patch corrects the overflow by formatting the timestamps as unsigned longs. ........ Merged revisions 430840 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	7dc784ffa9	Call extension state callbacks at hint creation. ... When a hint gets created, any subsequent device or presence state changes result in extension status events getting sent out to interested parties. However, at the time of hint creation, no such event gets sent out, so watchers of extension state are potentially left in the dark until the first state change after hint creation. Patch contributed by John Hardin (License #6512) ........ Merged revisions 430776 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	07e2a48ab1	REVERTING res_pjsip: make it unloadable ... Due to the original patch causing memory corruptions the patch is being removed until the problem can be resolved. ........ Merged revisions 430734 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	49542a794b	res_pjsip: make it unloadable ... The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue (listed below) by Corey Farrell with a few modifications. Namely, removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. This patch is the first step (should hopefully be followed by another/others at some point) in allowing res_pjsip and the modules that depend on it to be unloadable. At this time, res_pjsip and some of the modules that depend on res_pjsip cannot be unloaded without causing problems of some sort. The goal of this patch is to get res_pjsip and only res_pjsip to be able to unload successfully and/or shutdown without incident (crashes, leaks, etc...). Other dependent modules may still cause problems on unload. Basically made sure, with the patch applied, that res_pjsip (with no other dependent modules loaded) could be succesfully unloaded and Asterisk could shutdown without any leaks or crashes that pertained directly to res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4311/ patches: pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) ........ Merged revisions 430628 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4dd6b6ff59	AMI: Revert non-backwards compatible changes from earlier commit. ... * Reverted the change to astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. Unfortunately changing the case of a returned value is not a backward compatible change so for now FAXSessions is going to have to remain inconsistent with all of the other AMI list actions. * Reverted the minor protocol error fix in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". Caught by the testsuite. ASTERISK-24049 ........ Merged revisions 430528 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c7ea108e02	Revert -r430452 It needs to be redone for the next major AMI version change instead. ... ASTERISK-24049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	9065488ddd	main/syslog: Allow dynamic logs, such as security events, to log to the syslog ... The security event log uses a dynamic log level (SECURITY) that is registered with the Asterisk logging core. Unfortunately, the syslog would ignore log statements that had a dynamic log level associated with them. Because the syslog cannot handle ad hoc dynamic log levels, this patch treats any dynamic log entries sent to the syslog as logs with a level of NOTICE. ASTERISK-20744 #close Reported by: Michael Keuter Tested by: Michael L. Young, Jacek Konieczny patches: asterisk-20744-syslog-dynamic-logging_trunk.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 430506 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430507 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	5b30938394	app_bridge: return to the next dialplan priority ... When app_bridge grabs a channel and puts it into a bridge, the channel should then continue where it left off in the dialplan after the bridge has ended. Although it stores the current dialplan location as an after bridge goto on the channel, it was executing the same priority again instead of going to the next priority. By swapping the "specific" version of bridge_set_after_goto with bridge_set_after_go_on, the next priority in the dialplan is executed instead. ASTERISK-24637 #close Review: https://reviewboard.asterisk.org/r/4322/ Reported by: John Bigelow ........ Merged revisions 430467 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430468 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	ef34a05f21	AMI: Remove no longer used parameter from astman_send_listack(). ... Follow-up issue to -r430435 from reviewboard review. ASTERISK-24049 Review: https://reviewboard.asterisk.org/r/4315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	52a7cdb101	AMI: Make AMI actions that generate event lists consistent. ... * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	56de48107f	config: Add option to NOT preserve effective context when changing a template ... Let's say you have a template T with variable VAR1 = ON and you have a context C(T) that doesn't specify VAR1. If you read C, the effective value of VAR1 is ON. Now you change T VAR1 to OFF and call ast_config_text_file_save. The current behavior is that the file gets re-written with T/VAR1=OFF but C/VAR1=ON is added. Personally, I think this is a bug. It's preserving the effective state of C even though I didn't specify C/VAR1 in th first place. I believe the behavior should be that if I didn't specify C/VAR1 originally, then the effective value of C/VAR1 should continue to follow the inherited state. Now, if I DID explicitly specify C/VAR1, the it should be preserved even if the template changes. Even though I think the existing behavior is a bug, it's been that way forever so I'm not changing it. Instead, I've created ast_config_text_file_save2() that takes a bitmask of flags, one of which is to preserve the effective context (the current behavior). The original ast_config_text_file_save calls *2 with the preserve flag. If you want the new behavior, call *2 directly without a flag. I've also updated Manager UpdateConfig with a new parameter 'PreserveEffectiveContext' whose default is 'yes'. If you want the new behavior with UpdateConfig, set 'PreserveEffectiveContext: no'. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4297/ ........ Merged revisions 430295 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	0c5234f12a	Fix dev-mode build on recent gcc ... ........ Merged revisions 430274 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430275 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	0b8fbf9238	bridge: avoid leaking channel during blond transfer pt2 ... A blond transfer to a failed destination, when followed by a recall attempt, lead to a leak of the reference to the destination channel. In addition to correcting the regression on the previous attempt (r429826) this fixes the leak and two additional reference leaks on failures of bridge_import. ASTERISK-24513 #close Review: https://reviewboard.asterisk.org/r/4302/ ........ Merged revisions 430199 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 430200 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	8d059c3808	rtp_engine: keep payload types in correct range ... In r428708 additional codecs were added including a payload type of 128 which is outside of nominal range of 0-127. This change moves changes 128 to 96 to avoid causing a pjsip assertion when making a call to an endpoint configured with allow=all. ASTERISK-24367 #close Review: https://reviewboard.asterisk.org/r/4286/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	3a73c6c90e	main/pbx.c: Fix double lock of contexts lock introduced by r429967 ... We only need to hold the context_merge_lock once. Locking it twice will make many other parts of Asterisk very sad. ASTERISK-24641 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	1c0604e905	DTMF atxfer: Setup recall channels as if the transferee initiated the call. ... After the initial DTMF atxfer call attempt to the transfer target fails to answer during a blonde transfer, the recall callback channels do not get setup with information from the initial transferrer channel. As a result, the recall callback to the transferrer does not have callid, channel variables, datastores, accountcode, peeraccount, COLP, and CLID setup. A similar situation happens with the recall callback to the transfer target but it is less visible. The recall callback to the transfer target does not have callid, channel variables, datastores, accountcode, peeraccount, and COLP setup. * Added missing information to the recall callback channels before initiating the call. callid, channel variables, datastores, accountcode, peeraccount, COLP, and CLID * Set callid of the transferrer channel on the DTMF atxfer controller thread attended_transfer_monitor_thread(). * Added missing channel unlocks and props unref to off nominal paths in attended_transfer_properties_alloc(). ASTERISK-23841 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4259/ ........ Merged revisions 430034 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430041 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	bbd9ff122e	queue_log: Post QUEUESTART entry when Asterisk fully boots. ... The QUEUESTART log entry has historically acted like a fully booted event for the queue_log file. When the QUEUESTART entry was posted to the log was broken by the change made by ASTERISK-15863. * Made post the QUEUESTART queue_log entry when Asterisk fully boots. This restores the intent of that log entry and happens after realtime has had a chance to load. AST-1444 #close Reported by: Denis Martinez Review: https://reviewboard.asterisk.org/r/4282/ ........ Merged revisions 430009 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430010 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0c38276d6e	presencestate: Allow channel drivers to provide presence state information ... This patch adds the ability for channel drivers to supply presence information in a similar manner to device state. The patch does not provide any channel driver implementations, but it does provide the core infrastructure necessary for channel drivers to provide such information. The core handles multiple providers of presence state information. Ordering of presence state is as follows: INVALID < NOT_SET < AVAILABLE < UNAVAILABLE < CHAT < AWAY < XA < DND Each provider can trump the previous if it provides a presence state that supercedes a previous one. Review: https://reviewboard.asterisk.org/r/4050 ASTERISK-24363 #close Reported by: Gareth Palmer patches: chan_presencestate-428146.patch uploaded by Gareth Palmer (License 5169) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ba403e83bd	acl: Fix reloading of configuration if configuration file does not exist at startup. ... The named ACL code incorrectly destroyed the config options information if loading of the configuration file failed at startup. This would result in reloading also failing even if a valid configuration file was put in place. ASTERISK-23733 #close Reported by: Richard Kenner ........ Merged revisions 429893 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429894 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	07d1012383	bridge: avoid leaking channel during blond transfer ... After a blond transfer (start attended and hang up) to a destination that also hangs up without answer, the Local;1 channel was leaked and would show up on core show channels. This was happening because the attended state blond_nonfinal_enter() resetting the props->transfer_target to null while releasing it's own reference, which would later prevent props from releasing another reference during destruction. The change made here is simply to not assign the target to NULL. ASTERISK-24513 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4262/ ........ Merged revisions 429826 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 429827 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2b1f2b5c1f	Prevent possible race condition on dual redirect of channels in the same bridge. ... The AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT flag was created to prevent bridges from prematurely acting on orphaned channels in bridges. The problem with the AMI redirect action was that it was setting this flag on channels based on the presence of a PBX, not whether the channel was in a bridge. Whether a channel has a PBX is irrelevant, so the condition has been altered to check if the channel is in a bridge. ASTERISK-24536 #close Reported by Niklas Larsson Review: https://reviewboard.asterisk.org/r/4268 ........ Merged revisions 429741 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429745 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	8b6ecc449c	Fix printf problems with high ascii characters after r413586 (1.8). ... In r413586 (1.8) various casts were added to silence gcc 4.10 warnings. Those fixes included things like: -out += sprintf(out, "%%%02X", (unsigned char) *ptr); +out += sprintf(out, "%%%02X", (unsigned) *ptr); That works for low ascii characters, but for the high range that yields e.g. FFFFFFC3 when C3 is expected. This changeset: - fixes those casts to use the 'hh' unsigned char modifier instead - consistently uses %02x instead of %2.2x (or other non-standard usage) - adds a few 'h' modifiers in various places - fixes a 'replcaes' typo - dev/urandon typo (in 13+ patch) Review: https://reviewboard.asterisk.org/r/4263/ ASTERISK-24619 #close Reported by: Stefan27 (on IRC) ........ Merged revisions 429673 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429674 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 429675 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	6472568bc6	loader: Move definition of ast_module_reload from _private.h to module.h ... No functionality change. Just move the definition of ast_module_reload from _private.h to module.h so it can be public. Also removed the include of _private.h from manager.c since ast_module_load was the only reason for including it. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4251/ ........ Merged revisions 429542 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	308c1b41dd	DEBUG_THREADS: Fix regression and lock tracking initialization problems. ... This patch started with David Lee's patch at https://reviewboard.asterisk.org/r/2826/ and includes a regression fix introduced by the ASTERISK-22455 patch. The initialization of a mutex's lock tracking structure was not protected in a critical section. This is fine for any mutex that is explicitly initialized, but a static mutex may have its lock tracking double initialized if multiple threads attempt the first lock simultaneously. * Added a global mutex to properly serialize initialization of the lock tracking structure. The painful global lock can be mitigated by adding a double checked lock flag as discussed on the original review request. * Defer lock tracking initialization until first use. * Don't be "helpful" and initialize an uninitialized lock when DEBUG_THREADS is enabled. Debug code is not supposed to fix or change normal code behavior. We don't need a lock initialization race that would force a re-setup of lock tracking. Lock tracking already handles initialization on first use. * Properly handle allocation failures of the lock tracking structure. * No need to initialize tracking data in __ast_pthread_mutex_destroy() just to turn around and destroy it. The regression introduced by ASTERISK-22455 is the result of manipulating a pthread_mutex_t struct outside of the pthread library code. The pthread_mutex_t struct seems to have a global linked list pointer member that can get changed by other threads. Therefore, saving and restoring the contents of a pthread_mutex_t struct is a bad thing. Thanks to Thomas Airmont for finding this obscure regression. * Don't overwrite the struct ast_lock_track.reentr_mutex member to restore tracking data in __ast_cond_wait() and __ast_cond_timedwait(). The pthread_mutex_t struct must be treated as a read-only opaque variable. Miscellaneous other items fixed by this patch: * Match ast_suspend_lock_info() with ast_restore_lock_info() in __ast_cond_timedwait(). * Made some uninitialized lock sanity checks return EINVAL and try a DO_THREAD_CRASH. * Fix bad canlog initialization expressions. ASTERISK-24614 #close Reported by: Thomas Airmont Review: https://reviewboard.asterisk.org/r/4247/ Review: https://reviewboard.asterisk.org/r/2826/ ........ Merged revisions 429539 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	8d325be503	media: Fix crash when determining sample count of a frame during shutdown. ... When shutting down Asterisk the codecs are cleaned up. As a result anything attempting to get a codec based on ID or details will find that no codec exists. This currently occurs when determining the sample count of a frame. This code did not take this situation into account. This change fixes this by getting the codec directly from the format and eliminates the lookup. This is both faster and also provides a guarantee that the codec will exist and will be valid. ASTERISK-24604 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4260/ ........ Merged revisions 429497 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	8fe45f0f0a	core: avoid possible asterisk -r crash from long id ... When connecting to the remote console, an id string is first provided that consts of the hostname, pid, and version. This is parsed by the remote instance using a buffer that may be too short, and can allow a buffer overrun because it is not terminated. This patch adds termination and a larger buffer. Review: https://reviewboard.asterisk.org/r/4182/ ........ Merged revisions 429223 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	d673209abc	ARI/AMI: Include language in standard channel snapshot output ... The channel "language" was already part of a channel snapshot, however is was not sent out over AMI or ARI. This patch makes it so the channel "language" is included in the appropriate AMI or ARI events. ASTERISK-24553 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4245/ ........ Merged revisions 429204 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 429206 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	c17cef1c38	Direct Media calls within private network sometimes get one way audio ... When endpoints with direct_media enabled, behind a firewall (Asterisk on a separate network) and were bridged sometimes Asterisk would send the ip address of the firewall in the sdp to one of the phones in the reinvite resulting in one way audio. When sending the reinvite Asterisk will retrieve the media address from the associated rtp instance, but if frames were being read this can be overwritten with another address (in this case the firewall's). This patch ensures that Asterisk uses the original device address when using direct media. ASTERISK-24563 Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4216/ ........ Merged revisions 429195 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 429196 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	fe7671fee6	Add new AMI and ARI events for connected line changes on a channel. ... The AMI event is called NewConnectedLine and the ARI event is called ChannelConnectedLine. ASTERISK-24554 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/4231 ........ Merged revisions 429064 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	4bb556a847	Stasis: Fix StasisStart/End order and missing events ... This corrects several bugs that currently exist in the stasis application code. * After a masquerade, the resulting channels have channel topics that do not match their uniqueids ** Masquerades now swap channel topics appropriately * StasisStart and StasisEnd messages are leaked to observer applications due to being published on channel topics ** StasisStart and StasisEnd publishing is now properly restricted to controlling apps via app topics * Race conditions exist where StasisStart and StasisEnd messages due to a masquerade may be received out of order due to being published on different topics ** These messages are now published directly on the app topic so this is now a non-issue * StasisEnds are sometimes missing when sent due to masquerades and bridge swaps into and out of Stasis() ** This was due to StasisEnd processing adjusting message-sent flags after Stasis() had already exited and Stasis() had been re-entered ** This was corrected by adjusting these flags prior to sending the message while the initial Stasis() application was still shutting down Review: https://reviewboard.asterisk.org/r/4213/ ASTERISK-24537 #close Reported by: Matt DiMeo ........ Merged revisions 429061 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 429062 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429063 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d04445c24a	sorcery: Add additional observer capabilities. ... Add new global, instance and wizard observers. instance_created wizard_registered wizard_unregistered instance_destroying instance_loading instance_loaded wizard_mapped object_type_registered object_type_loading object_type_loaded wizard_loading wizard_loaded Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4215/ ........ Merged revisions 428999 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 429000 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429001 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	19992844be	main/test: Fix compilation issue on 32-bit systems ... On a 32-bit system, a type of intmax_t will result in a compilation warning when formatted as a 'long int'. Use the format specifier of %jd (which was what was used originally in manager.c) to format the JSON extracted integer on both 32-/64-bit systems. ........ Merged revisions 428972 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428973 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	343a83d7d8	main/test: Fix race condition between AMI topic and Test Suite topic ... This patch fixes a race condition between the raising of test AMI events (which drive many tests in the Asterisk Test Suite) and other AMI events. Prior to this patch, the Stasis messages published to the test topic were not forwarded to the AMI topic. Instead, the code in manager had a dedicated handler for test messages that was independent of the topics forwarded to the AMI topic. This results in no synchronization between the test messages and the rest of the Stasis messages published out over AMI. In some test with very tight timing constraints, this can result in out of order messages and spurious test failures. Properly forwarding the Test Suite topic to the AMI topic ensures that the messages are synchronized properly. This patch does that, and moves the message handling to the Stasis definition of the Test Suite message in test.c as well. Review: https://reviewboard.asterisk.org/r/4221/ ........ Merged revisions 428945 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428946 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	f128ff61ab	config: Create ast_variable_find_in_list() ... Add const char *ast_variable_find_in_list(const struct ast_variable *list, const char *variable); ast_variable_find() requires a config category to search whereas ast_variable_find_in_list() just needs the root list element which is useful if you don't have a category. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4217/ ........ Merged revisions 428733 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428734 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	4ff6bd831f	rtp_engine: Add support for transporting signed linear at 12kHz, 24kHz, 32kHz, 44kHz, 48kHz, 96kHz, and 192kHz over RTP. ... This change adds mappings in the RTP engine layer for the remaining signed linear formats. ASTERISK-24274 #close Reported by: Frankie Chin Review: https://reviewboard.asterisk.org/r/4093/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428708 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	1106e8fd0f	main/stasis: Allow subscriptions to use a threadpool for message delivery ... Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428687 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	7db3d1642b	channel: Extend size of buffer for codecs in "core show channeltype" CLI command. ... The static buffer for codecs when invoking the "core show channeltype" CLI command did not have enough room for all codecs. This has been extended so it does. ASTERISK-24542 #close Reported by: snuffy patches: channeltype-tech.diff submitted by snuffy (license 5024) Review: https://reviewboard.asterisk.org/r/4204/ ........ Merged revisions 428632 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c38ffca9a1	DTMF hooks: Leaving channels need to push any collected digits into the bridge. ... Any partially collected DTMF digits for a DTMF hook need to be pushed into the bridge when a channel leaves the bridging system as if there were a timeout. Review: https://reviewboard.asterisk.org/r/4199/ ........ Merged revisions 428601 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428602 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	3576ae47f4	manager: Fix could not extend string messages. ... When shutting down Asterisk that has an active AMI connection, you get several "failed to extend from %d to %d" messages because use of the EVENT_FLAG_SHUTDOWN attempts to add all AMI permission strings to the event. * Created MAX_AUTH_PERM_STRING to use when creating stack based struct ast_str variables used with the authority_to_str() and user_authority_to_str() functions instead of a variety of magic numbers that could be too small. * Added a special check for EVENT_FLAG_SHUTDOWN to authority_to_str() so it will not attempt to add all permission level strings. Review: https://reviewboard.asterisk.org/r/4200/ ........ Merged revisions 428570 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428571 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428572 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	4394e0431c	sorcery: Make is_object_field_registered handle field names that are regexes. ... As a result of https://reviewboard.asterisk.org/r/3305, res_sorcery_realtime was tossing database fields that didn't have an exact match to a sorcery registered field. This broke the ability to use regexes as field names which manifested itself as a failure of res_pjsip_phoneprov_provider which uses this capability. It also broke handling of fields that start with '@' in realtime but I don't think anyone noticed. This patch does the following... * Modifies ast_sorcery_fields_register to pre-compile the name regex. * Modifies ast_sorcery_is_object_field_registered to test the regex if it exists instead of doing an exact strcmp. * Modifies res_pjsip_phoneprov_provider with a few tweaks to get it to work with realtime. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4185/ ........ Merged revisions 428543 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428544 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	2be984fb11	main/bridge_basic: Fix features regressions introduced by r428165 ... In r428165, two bugs were introduced: * Prior to entering the features retry loop, the buffer that holds the collected digits is wiped. However, this inadvertently wipes out the first collected digit on the first pass through, which is obtained in ast_stream_and_wait. This caused all of the features tests to fail. * If ast_app_dtget returns a hangup (-1), the loop would retry incorrectly. If we detect a hangup, we have to stop trying the feature. This patch fixes both issues. Review: https://reviewboard.asterisk.org/r/4196/ ........ Merged revisions 428505 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428506 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2f78fde10f	Fix error with mixed address family ACLs. ... Prior to this commit, the address family of the first item in an ACL was used to compare all incoming traffic. This could lead to traffic of other IP address families bypassing ACLs. ASTERISK-24469 #close Reported by Matt Jordan Patches: ASTERISK-24469-11.diff uploaded by Matt Jordan (License #6283) AST-2014-012 ........ Merged revisions 428402 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 428417 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428422 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428425 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a7c9f4c668	ast_str: Fix improper member access to struct ast_str members. ... Accessing members of struct ast_str outside of the string manipulation API routines is invalid since struct ast_str is supposed to be treated as opaque. Review: https://reviewboard.asterisk.org/r/4194/ ........ Merged revisions 428244 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428245 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 428246 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2e750db120	Allow for transferer to retry when dialing an invalid extension. ... This allows for a configurable number of attempts for a transferer to dial an extension to transfer the call to. For Asterisk 13, the default values are such that upgrading between versions will not cause a behaivour change. For trunk, though, the defaults will be changed to be more user-friendly. Review: https://reviewboard.asterisk.org/r/4167 ........ Merged revisions 428145 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	df2090b931	Documentation: Revise explanation of cdr.conf option 'Unanswered' ... ASTERISK-24279 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4109/ ........ Merged revisions 427901 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427902 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	ba811ae1c3	stun: correct attribute string padding to match rfc ... When sending the USERNAME attribute in an RTP STUN response, the implementation in append_attr_string passed the actual length, instead of padding it up to a multiple of four bytes as required by the RFC 3489. This change adds separate variables for the string and padded attributed lengths, and performs padding correctly. Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/4139/ ........ Merged revisions 427874 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427875 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427876 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427877 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2d9471ab1f	Fix race condition that could result in ARI transfer messages not being sent. ... From reviewboard: "During blind transfer testing, it was noticed that tests were failing occasionally because the ARI blind transfer event was not being sent. After investigating, I detected a race condition in the blind transfer code. When blind transferring a single channel, the actual transfer operation (i.e. removing the transferee from the bridge and directing them to the proper dialplan location) is queued onto the transferee bridge channel. After queuing the transfer operation, the blind transfer Stasis message is published. At the time of publication, snapshots of the channels and bridge involved are created. The ARI subscriber to the blind transfer Stasis message then attempts to determine if the bridge or any of the involved channels are subscribed to by ARI applications. If so, then the blind transfer message is sent to the applications. The way that the ARI blind transfer message handler works is to first see if the transferer channel is subscribed to. If not, then iterate over all the channel IDs in the bridge snapshot and determine if any of those are subscribed to. In the test we were running, the lone transferee channel was subscribed to, so an ARI event should have been sent to our application. Occasionally, though, the bridge snapshot did not have any channels IDs on it at all. Why? The problem is that since the blind transfer operation is handled by a separate thread, it is possible that the transfer will have completed and the channels removed from the bridge before we publish the blind transfer Stasis message. Since the blind transfer has completed, the bridge on which the transfer occurred no longer has any channels on it, so the resulting bridge snapshot has no channels on it. Through investigation of the code, I found that attended transfers can have this issue too for the case where a transferee is transferred to an application." The fix employed here is to decouple the creation of snapshots for the transfer messages from the publication of the transfer messages. This way, snapshots can be created to reflect what they are at the time of the transfer operation. Review: https://reviewboard.asterisk.org/r/4135 ........ Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427870 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427873 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cc4c396647	main/rtp_engine: Fix crash when processing more than one RTCP report info block ... Asterisk - in res_rtp_asterisk - only understands a single RTCP report info block. When the RTCP information was refactored in the RTP Engine to be pushed over the Stasis message bus, I put in the hooks into the engine to handle multiple RTCP report info blocks, in the hope that a future RTP implementation would be able to provide that data. Unfortunately, res_rtp_asterisk has a tendency to "lie": (1) It will send RTCP reports with a reception_report_count greater than 1 (which is pulled directly from the RTCP packet itself, so that part is correct) (2) It will only provide a single report block When the rtp_engine goes to convert this to a JSON blob, hilarity ensues as it looks for a report block that doesn't exist. This patch updates the rtp_engine to be a bit more skeptical about what it is presented with. While this could also be fixed in res_rtp_asterisk, this patch prefers to fix it in the engine for two reasons: (1) The engine is designed to work with multiple RTP implementation, and hence having it be more robust is a good thing (tm) (2) res_rtp_asterisk's handling of RTCP information is "fun". It should report the correct reception_report_count; ideally it should also be giving us all of the blocks - but it is *definitely* not designed to do that. Going down that road is a non-trivial effort. Review: https://reviewboard.asterisk.org/r/4158/ ASTERISK-24489 #close Reported by: Gregory Malsack Tested by: Gregory Malsack ASTERISK-24498 #close Reported by: Beppo Mazzucato Tested by: Beppo Maazucato ........ Merged revisions 427762 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427763 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ec1a7654f3	Fix leak in AMI Action Bridge ... Add missing reference cleanup for newly created bridge. ASTERISK-24281 Reported by: Stefan Engström Review: https://reviewboard.asterisk.org/r/4154/ ........ Merged revisions 427736 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427737 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	dbb8f0a935	pbx: Fix off-nominal case where a freed extension may still be used. ... If during the operation of adding an extension a priority is added but fails it is possible for the extension to be freed but still exist in the PBX core. If this occurs subsequent lookups may try to access the extension and end up in freed memory. This change removes the extension from the PBX core when the priority addition fails and then frees the extension. ASTERISK-24444 #close Reported by: Leandro Dardini Review: https://reviewboard.asterisk.org/r/4162/ ........ Merged revisions 427709 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427710 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427711 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427712 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	21c41e4542	manager: Fix HTTP connection reference leaks. ... Fix reference leak that happens if (session && !blastaway). ASTERISK-24505 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4153/ ........ Merged revisions 427641 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427642 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427643 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a68baad74f	Bridge DTMF hooks: Made audio pass from the bridge while waiting for more matching digits. ... * Made collecting DTMF digits for the DTMF feature hooks pass frames from the bridge. * Made collecting DTMF digits possible by other bridge hooks if there is a need. ASTERISK-24447 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4123/ ........ Merged revisions 427493 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427494 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427495 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	4d80f223af	main/file.c: fix possible extra ast_module_unref to format modules. ... fn_wrapper only adds a reference to the format's module if the file was able to be opened. If not this causes an unmatched ast_module_unref in filestream_destructor. Move ast_module_ref to get_stream. ASTERISK-24492 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4149/ ........ Merged revisions 427464 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427465 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427466 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	7e2369310c	Fix unintential memory retention in stringfields. ... * Fix missing / unreachable calls to __ast_string_field_release_active. * Reset pool->used to zero when the current pool->active reaches zero. ASTERISK-24307 #close Reported by: Etienne Lessard Tested by: ibercom, Etienne Lessard Review: https://reviewboard.asterisk.org/r/4114/ ........ Merged revisions 427380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 427381 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427382 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d5de94201e	config: Make text_file_save and 'dialplan save' escape semicolons in values. ... When a config file is read, an unescaped semicolon signals comments which are stripped from the value before it's stored. Escaped semicolons are then unescaped and become part of the value. Both of these behaviors are normal and expected. When the config is serialized either by 'dialplan save' or AMI/UpdateConfig however, the now unescaped semicolons are written as-is. If you actually reload the file just saved, the unescaped semicolons are now treated as start of comments. Since true comments are stripped on read, any semicolons in ast_variable.value must have been escaped originally. This patch re-escapes semicolons in ast_variable.values before they're written to file either by 'dialplan save' or config/ast_config_text_file_save which is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting issues nearby in pbx_config.c Tested-by: George Joseph ASTERISK-20127 #close Review: https://reviewboard.asterisk.org/r/4132/ ........ Merged revisions 427275 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427276 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	bd42a09d7f	config: BUG: Restore ability for non-templ to be used as base objs in config. ... My recent refactor of config.c accidentally removed the capability for an object to inherit from a non-template object. This patch restores the capability to inherit from both template and non-template objects. Tested-by: George Joseph Reported-by: Scott Griepentrog ASTERISK-24487 #close Review: https://reviewboard.asterisk.org/r/4147/ ........ Merged revisions 427227 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427228 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	bdc35c77b9	Fix crash caused by merge error on review 4138 ... When merging from 12 to 13 there were conflicts, I mistakenly had the loop run ast_closestream(others[0]) when it should be ast_closestream(others[x]). ........ Merged revisions 427181 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	509c04ef38	Fix ast_writestream leaks ... Fix cleanup in __ast_play_and_record where others[x] may be leaked. This was caught where prepend != NULL && outmsg != NULL, once realfile[x] == NULL any further others[x] would be leaked. A cleanup block was also added for prepend != NULL && outmsg == NULL. 11+: Fix leak of ast_writestream recording_fs in app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/ ........ Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 427024 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427025 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427026 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	85c1822a9d	func_jitterbuffer: fix frame leaks. ... Fix code paths where it is possible for frames to leak. Fix uninitialized variable in jb_get_fixed and jb_get_adaptive. ASTERISK-22409 #related Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4128/ ........ Merged revisions 427019 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427020 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427021 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ced81afff2	audiohooks: Clean references to formats ... Cleanup references to in_translate[x].format and out_translate[x].format in ast_audiohook_detach_list. ASTERISK-24465 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4124/ ........ Merged revisions 426803 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	67e496c275	manager: Unsubscribe from acl_change_sub at shutdown. ... ASTERISK-24453 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4110/ ........ Merged revisions 426524 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426525 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426526 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8e9f593e3a	main/bridge: Destroy features struct on off nominal path during bridge impart ... When a channel is imparted to a bridge, the invocation of the function may provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart, the caller must assume that ownership has passed to the function, as in all paths the function destroys the struct prior to returning (as its purpose is to configure the behavior of the channel while in the bridge). On one off nominal path - where the channel already has a PBX thread - the struct was not being destroyed. This patch fixes that glitch. ASTERISK-24437 #close Reported by: Scott Griepentrog ........ Merged revisions 426431 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426432 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f4b4d42630	main/manager: Fix typo in AMI event documentation of "OriginateResponse" ... The parameter name is "Response", not "Resonse". ASTERISK-24430 #close Reported by: Dafi Ni ........ Merged revisions 426366 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426367 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426368 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426369 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	c084728690	Documentation: Improve documentation for ExtensionStatus AMI events ... Review: https://reviewboard.asterisk.org/r/4085/ ........ Merged revisions 426120 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2165868be7	translage.c: Fix regression when generating translation path strings. ... Fix the AMI Status action read and write translation path strings from growing for each channel in the status event list by reseting the ast string given to ast_translate_path_to_str() to fill in the given translation path. ........ Merged revisions 426079 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	dad0334cf1	AST-2014-011: Fix POODLE security issues ... There are two aspects to the vulnerability: (1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module to use TLSv1+. At this time, it does not refactor res_jabber/res_xmpp to use the TCP/TLS core, which should be done as an improvement at a latter date. (2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left unspecified, will default to the OpenSSL SSLv23_method. This method allows for all ecnryption methods, including SSLv2/SSLv3. A MITM can exploit this by forcing a fallback to SSLv3, which leaves the server vulnerable to POODLE. This patch adds WARNINGS if a user uses SSLv2/SSLv3 in their configuration, and explicitly disables SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly chosen. For TLS servers, Asterisk will no longer support SSLv2 or SSLv3. Much thanks to abelbeck for reporting the vulnerability and providing a patch for the res_jabber/res_xmpp modules. Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425 #close Reported by: abelbeck Tested by: abelbeck, opsmonitor, gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded by mjordan (License 6283) ........ Merged revisions 425987 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425991 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	f91cb1207c	AMI: Add missing VarSet events when a channel inherits variables. ... There should be AMI VarSet events when channel variables are inherited by an outgoing channel. Also local;2 should generate VarSet events when it gets all of its channel variables from channel local;1. ASTERISK-24415 #close Reported by: Richard Mudgett Patches: jira_asterisk_24415_v12.patch (license #5621) patch uploaded by Richard Mudgett Review: https://reviewboard.asterisk.org/r/4074/ ........ Merged revisions 425782 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425783 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425784 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	873d956144	main/cdr: Use 'time' when rescheduling batched CDRs as opposed to 'size' ... When refactoring CDRs to use the configuration framework, a 'whoops' was introduced where the CDR batch size was used when rescheduling a batch, as opposed to the time duration. This patch corrects that obvious mistake. ASTERISK-24426 #close Reported by: Shane Blaser ........ Merged revisions 425735 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425736 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c2ec5f0f6f	config: Fix inf loop using ast_category_browse and ast_variable_retrieve ... Fix infinite loop when calling ast_variable_retrieve inside an ast_category_browse loop when there is more than 1 category with the same name. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4089/ ........ Merged revisions 425713 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425714 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	104fca5001	config: Fix SEGV in unit test with MALLOC_DEBUG ... With MALLOC_DEBUG the /main/config config_basic_ops test was causing a SEGV while doing an ast_category_delete in an ast_category_browse loop. Apparently this never worked but was also never tested. I removed the test, added 2 notes to config.h indicating that it's not supported and added a few lines of code to ast_category_delete to prevent the SEGV should someone attempt it in the future. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4078/ ........ Merged revisions 425525 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425526 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	87b5006ff0	Scheduler: Fix a nasty scheduler caching bug which makes new tasks not execute ... Tasks that were marked for pending deletion in the scheduler would be moved to the cache for later reuse, but after being recycled the deleted mark wouldn't be removed resulting in fresh tasks being deleted without reason... and immediately moved back into the cache where they could be reused again. This could cause horrendous things to happen in just about anything that used a scheduler. ASTERISK-24321 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4071/ ........ Merged revisions 425503 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425504 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c61b66e107	stasis_channels.c: Resolve unfinished Dials when doing masquerades (Part 2) ... Masquerades into and out of channels that are involved in a dial operation don't create the expected dial end event. The missing dial end event goes against the model for things like CDRs and generating Dial end manager actions and such. There are four cases: 1) A channel masquerades into the caller channel. The case happens when performing a blonde transfer using the channel driver's protocol. 2) A channel masquerades into a callee channel. The case happens when performing a directed call pickup. 3) The caller channel masquerades out of dial. The case happens when using the Bridge application on the caller channel. 4) A callee channel masquerades out of dial. The case happens when using the Bridge application on a peer channel. As it turned out, all four cases need to be handled instead of just the first one. ASTERISK-24237 Reported by: Richard Mudgett ASTERISK-24394 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4066/ ........ Merged revisions 425430 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425455 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	c7e6b6ba3d	manager/config: Support templates and non-unique category names via AMI ... This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	98d5b7090d	bridge: During a smart bridge operation provide a more complete bridge to the old technology. ... When a smart bridge operation occurs and a bridge transitions from one technology to another the old technology is provided the channels formerly in it and told that they are leaving. Unfortunately the bridge provided along with them is incomplete. The bridge, despite there being channels in it, contains none. This forces technology implementations to have additional logic when channels are leaving or to store their own duplicated state. This change makes the bridge more complete so it contains the expected channels. Now that the bridge is complete special logic within bridge_native_rtp is no longer needed and has been removed. Review: https://reviewboard.asterisk.org/r/4057/ ........ Merged revisions 425242 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	37b5f52da7	CallerID: Fix parsing regression ... This fixes a regression in callerid parsing introduced when another bug was fixed. This bug occurred when the name was composed entirely of DTMF keys and quoted without a number section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/ ........ Merged revisions 425152 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425153 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425154 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425155 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425156 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	cc595f7353	res_phoneprov: Refactor phoneprov to allow pluggable config providers ... This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424964 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	0f50e8856b	cdr.c: Make turning on CDR debug a one step process instead of two. ... Now "cdr set debug on" doesn't also require "core set verbose 1" to see CDR debug output. ........ Merged revisions 424941 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424942 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	5e50638539	Stasis: Relegate log message to dev-mode ... This error message primarily applies to development tasks and will now only show up when dev-mode is enabled via configure. ........ Merged revisions 424850 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	3dfc485e35	Indexer: Format message types may not exist ... In Asterisk 13+, any given message type is not guaranteed to exist even if Asterisk comes up correctly since creation of the message type could be declined. The indexer should not prevent Asterisk from starting under these conditions. ........ Merged revisions 424833 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	d8bbf1ec1d	Stasis: Only log errors for non-declined types ... When message type creation is declined via stasis.conf, certain operations log errors assuming that the declined type is being used before initialization or after destruction. These error messages get quite spammy for oft used message types and should not be logged in the first place since the message type is validly NULL. Reported by: Matt DiMeo ........ Merged revisions 424769 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	f7225da08a	data: Properly access formats in capabilities structure when adding codecs. ... Formats within a capabilities structure are addressed starting at 0, not 1. Assuming 1 causes it to exceed an array. ASTERISK-24389 #close Reported by: Kevin Harwell ........ Merged revisions 424752 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424753 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	39bd5b7a70	message: Don't close an AMI connection on SendMessage action error ... If SendMessage encounters an error (such as incorrect input provided to the action), it will currently return -1. Actions should only return -1 if the connection to the AMI client should be closed. In this case, SendMessage causing the client to disconnect is inappropriate. This patch causes the action to return 0, which simply causes the action to fail. Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354 #close Reported by: Peter Katzmann patches: sendMessage.patch uploaded by Peter Katzmann (License 5968) ........ Merged revisions 424690 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424691 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c384532aa4	features.c: Fix lingering channel ref while Bridge() application is active. ... Using the Bridge application to bridge a channel that is executing an applicaiton such as Wait results in a lingering Surrogate channel in the CLI "core show channels" output even though it has already hungup. * Fix bridge_exec() to not hold onto the current_dest_chan ref once it has been put into the bridge. * Eliminated bridge_exec()'s use of RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged revisions 424668 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424669 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	3a87f32dc0	sdp_srtp: Add new lines to some WARNING messages ... ........ Merged revisions 424646 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424647 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	9611ef4f1e	Release AMI connections on shutdown. ... ASTERISK-24378 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4037/ ........ Merged revisions 424578 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424579 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424580 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424581 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	70301b0438	audiohooks: Reevaluate the bridge technology when an audiohook is added or removed. ... Adding a mixmonitor to a channel causes the bridge to change technologies from native to simple_bridge so the call can be recorded. However, when the mixmonitor is stopped the bridge does not switch back to the native technology. * Added unbridge requests to reevaluate the bridge when a channel audiohook is removed. * Moved the unbridge request into ast_audiohook_attach() ensure that the bridge reevaluates whenever an audiohook is attached. This simplified the mixmonitor and chan_spy start code as well. * Added defensive code to stop_mixmonitor_full() in case additional arguments are ever added to the StopMixMonitor application. * Made ast_framehook_detach() not do an unbridge request if the framehook does not exist. * Made ast_framehook_list_fixup() do an unbridge request if there are any framehooks. Also simplified the loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4046/ ........ Merged revisions 424506 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424507 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	0165c5f95a	chan_pjsip: Fix deadlock when masquerading PJSIP channels. ... Performing a directed call pickup resulted in a deadlock when PJSIP channels were involved. A masquerade needs to hold onto the channel locks while it swaps channel information between the two channels involved in the masquerade. With PJSIP channels, the fixup routine needed to push a fixup task onto the PJSIP channel's serializer. Unfortunately, if the serializer was also processing a task that needed to lock the channel, you get deadlock. * Added a new control frame that is used to notify the channels that a masquerade is about to start and when it has completed. * Added the ability to query taskprocessors if the current thread is the taskprocessor thread. * Added the ability to suspend/unsuspend the PJSIP serializer thread so a masquerade could fixup the PJSIP channel without using the serializer. ASTERISK-24356 #close Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/4034/ ........ Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424472 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	4967478d18	sorcery: Prevent SEGV in sorcery_wizard_create when there's no create function ... When you call ast_sorcery_create() you don't necessarily know which wizard is going to be invoked. If it happens to be a wizard like 'config' that doesn't have a 'create' virtual function you get a segfault in the sorcery_wizard_create callback. This patch catches the null function pointer, does an ast_assert, and logs an error. Review: https://reviewboard.asterisk.org/r/4044/ ........ Merged revisions 424447 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424448 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	4246652603	Manager: Add missing fields and documentation for CoreShowChannels ... This corrects some issues introduced in the responses to the CoreShowChannels AMI command as well as adding documentation for the responses. The command in Asterisk 12 was missing the following fields: Duration, Application, ApplicationData, and BridgedChannel and BridgedUniqueID (replaced with BridgeId). ASTERISK-24262 #close Reported by: Mitch Claborn Review: https://reviewboard.asterisk.org/r/4040/ ........ Merged revisions 424423 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424424 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9d2bc0675a	threadpool.c: Minor cleanup fixes. ... * Fix threadpool_alloc() prototype. * Add missing off-nominal NULL check of pool in threadpool_alloc(). * searializer_create() does not need to create the object with a lock as the lock is not used. ........ Merged revisions 424096 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424097 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	e0abb82ab8	core: Ouch, forgot to undo a test free() in r423978. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	37179a2b1f	core: Don't allow free to mean ast_free (and malloc, etc..). ... This gets rid of most old libc free/malloc/realloc and replaces them with ast_free and friends. When compiling with MALLOC_DEBUG you'll notice it when you're mistakenly using one of the libc variants. For the legacy cases you can define WRAP_LIBC_MALLOC before including asterisk.h. Even better would be if the errors were also enabled when compiling without MALLOC_DEBUG, but that's a slightly more invasive header file change. Those compiling addons/format_mp3 will need to rerun ./contrib/scripts/get_mp3_source.sh. ASTERISK-24348 #related Review: https://reviewboard.asterisk.org/r/4015/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	63a4da4a0d	cli.c: Fix tab completion "module load" when MALLOC_DEBUG is enabled. ... r421600 conflicted with r155763. ASTERISK-24348 #close ........ Merged revisions 423657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423658 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423659 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423660 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	64a9e5f001	main/channel: Unlock channel in off-nominal path ... In r423414 (13) / r423415 (trunk), an API call that determines if a format capability structure is empty was added. This returns true if the format capability structure is completely empty or "none". A check for this was added in channel.c's set_format call. Unfortunately, when this check was true, it returned from the function while still holding the channel lock. This caused the CDR unit tests - which have a tendency to create channels with no formats - to deadlock. Whoops. This patch unlocks the channel on the off-nominal path. ........ Merged revisions 423641 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	6dae345674	Stasis_channels: Resolve unfinished Dials when doing masquerades ... Masquerades into channels that are in the dialing state don't end their dial and this goes against the model for things like CDRs and generating Dial end manager actions and such. ASTERISK-24237 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3990/ ........ Merged revisions 423525 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423530 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	7f2623a26f	PJSIP: Prevent T38 framehook being put on wrong channel ... This change gives framehooks a reverse-direction masquerade callback in addition to chan_fixup_cb similar to the callback added to datastores to handle the same situation. The new callback provides the same parameters as the fixup callback, but is called on the new channel's framehooks before moving framehooks from the old channel to the new channel. This gives the framehooks an oppurtunity to decide whether they should remain on the new channel or be removed. This new callback is used to prevent the PJSIP T.38 framehook from remaining on a masqueraded channel if the new channel is not also a PJSIP channel. This was causing a crash when a local channel was masqueraded into a PJSIP channel and the framehook was executed on the local channel since the channel's tech private data was not structured as expected. Review: https://reviewboard.asterisk.org/r/4001/ ........ Merged revisions 423503 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423504 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	ad8ef9175a	utils: Create ast_strsep function that ignores separators inside quotes ... This function acts like strsep with three exceptions... * The separator is a single character instead of a string. * Separators inside quotes are treated literally instead of like separators. * You can elect to have leading and trailing whitespace and quotes stripped from the result and have '\' sequences unescaped. Like strsep, ast_strsep maintains no internal state and you can call it recursively using different separators on the same storage. Also like strsep, for consistent results, consecutive separators are not collapsed so you may get an empty string as a valid result. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3989/ ........ Merged revisions 423476 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423478 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423480 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a7add3a257	astobj2.c/refcounter.py: Fix to deal with invalid object refs. ... * Make astob2 REF_DEBUG output an invalid object line when an invalid ao2 object ref/unref is attempted. This is similar to the constructor/destructor lines. * Fixed refcounter.py to handle skewed objects that have constructor/destructor states. * Made refcounter.py highlight the invalid ao2 object refs by putting them in their own section of the processed output file. * Made refcounter.py highlight unreffing an object by more than one that results in a negative ref count and the object being destroyed. The abnormally destroyed object is reported in the invalid and finalized object sections of the output. Review: https://reviewboard.asterisk.org/r/3971/ ........ Merged revisions 423349 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 423400 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 423416 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423418 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423422 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago