asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	b38f1146e5	config: Fix ast_config_text_file_save2 writability check for missing files A patch I did back in 2014 modified ast_config_text_file_save2 to check the writability of the main file and include files before truncating and re-writing them. An unintended side-effect of this was that if a file doesn't exist, the check fails and the write is aborted. This patch causes ast_config_text_file_save2 to check the writability of the parent directory of missing files instead of checking the file itself. This allows missing files to be created again. A unit test was also added to test_config to test saving of config files. The regression was discovered when app_voicemail's passwordlocation=spooldir feature stopped working. ASTERISK-25917 #close Reported-by: Jonathan Rose Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80	9 years ago
Joshua Colp	83dadc4683	Merge "app_queue: queue members can receive multiple calls" into 13	9 years ago
kkm	29bab0d1a4	chan_sip: Make autocreated peers send PeerStatus events Since Stasis has been introduced, an attempt to send AMI messages by an autocreated peer caused a crash, and all events from autocreated peers were semi-inadvertently disabled altogether in `0b83761`. This change restores the disabled functionality. ASTERISK-25950 Change-Id: Iecc350f23db603fadb2f302064643ebe9664e974	9 years ago
zuul	6c28e8c827	Merge "Fix case sensitive actions in AMI QueueSummary and QueueStatus" into 13	9 years ago
Kevin Harwell	c345e530f4	app_queue: queue members can receive multiple calls It was possible for a queue member that is a member of at least 2 or more queues to receive mulitiple calls at the same time. This happened because of a race between when a member was being rung and when the device state notified the other queue(s) member object of the state change. This patch makes it so when a queue member is being rung it gets added to a global pool of queue members. If that same member is tried again, e.g. from another queue, and it is found to already exist in the pending member container then it will not ring that member. ASTERISK-16115 #close Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48	9 years ago
zuul	a8f8e3c340	Merge "res_agi: Prevent run_agi from eating frames it shouldn't" into 13	9 years ago
DarkS	c0688a6398	Fix case sensitive actions in AMI QueueSummary and QueueStatus ASTERISK-25954 #close Reported by: Javier Acosta Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256	9 years ago
George Joseph	eb7c581806	res_agi: Prevent run_agi from eating frames it shouldn't The run_agi function is eating control frames when it shouldn't be. This is causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond transfer. Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie answers. Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE and is left thinking he's connected to Bob. In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on Charlie's channel. The fix was to accumulate deferrable frames in the "forever" loop instead of dropping them, and re-queue them just before running the actual agi command or exiting. ASTERISK-25951 #close Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645	9 years ago
zuul	13ee3402ed	Merge "func_odbc: Use one connection per DSN." into 13	9 years ago
zuul	84d2e4fb42	Merge "Remove reference to non-existent sip.conf option" into 13	9 years ago
zuul	77ae5cd7fd	Merge "res_stasis: Handle re-enter stasis bridge with swap channel." into 13	9 years ago
zuul	09f8f8daa1	Merge "bridge: Hold off more than one imparting channel at a time." into 13	9 years ago
Richard Mudgett	ebf0724a83	test_message.c: Wait longer in case dialplan also processes the test message. Bumped the wait from 1 second to 5 seconds. The test message was hitting my default call handler and failing the test because it took longer. Change-Id: I3a03737f25e92983de00548fcc7bbc50dd7544ba	9 years ago
Richard Mudgett	ba63aa7c9e	Manager: Short circuit AMI message processing. Improve AMI message processing performance if there are no consumers listening for the messages. We now skip creating the AMI event message text strings. Change-Id: I7b22fc5ec4e500d00635c1a467aa8ea68a1bb2b3	9 years ago
Richard Mudgett	d5ee6acf28	manager.c: Eliminate most RAII_VAR usage. * Made ast_manager_event_blob_create() not allocate the ao2 event object with a lock as it is not needed. Change-Id: I8e11bfedd22c21316012e0b9dd79f5918f644b7c	9 years ago
Richard Mudgett	7303e3dc96	manager_channels.c: Fix allocation failure crash. An earlier allocation failure failed to create a channel snapshot for the AMI HangupRequest/SoftHangupRequest event which resulted in a crash in channel_hangup_request_cb(). Where the stasis message gets generated cannot tell if the NULL snapshot returned was because of an allocation failure or the channel was a dummy channel. * Made channel_hangup_request_cb() check if the channel blob has a snapshot and exit if it doesn't. * Eliminated the RAII_VAR usage in channel_hangup_request_cb(). Change-Id: I0b6a1c4e95cbb7d80b2a7054c6eadecc169dfd24	9 years ago
Richard Mudgett	1e93f3d723	Bridge system: Fix memory leaks and double frees on impart failure. You cannot reference the passed in features struct after calling ast_bridge_impart(). Even if the call fails. Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21	9 years ago
Richard Mudgett	5e388d4188	bridge_softmix.c: Fix crash if channel fails to join mixing tech. softmix_bridge_join() failed because of an allocation failure. To address this, the softmix bridge technology now checks if the channel failed to join softmix successfully. In addition, the bridge now begins the process of kicking the channel out of the bridge so we don't have channels partially in the bridge for very long. * Fix the test_channel_feature_hooks.c unit tests. The test channel must have a valid codec to join the simple_bridge technology. This patch makes joining a bridge more strict by not allowing partially joined channels to remain in the bridge. Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b	9 years ago
Mark Michelson	068ae54c76	func_odbc: Use one connection per DSN. res_odbc was changed in Asterisk 13.8.0 to remove connection management, opting instead to let unixodbc maintain open connections and return those to Asterisk as requested. This was a boon for realtime, since it meant that multiple threads could potentially run parallel queries since they could each be using their own database connections. However, on the user-facing side, func_odbc, there were some inherent behaviors being relied on that no longer hold true after the change. One such reported behavior was that MySQL's LAST_INSERTED_ID() works per-connection. This means that if Asterisk uses separate connections for every database operation, whereas before it used one connection for everything, we have broken expectations and functionality. The fix provided in this patch is to make func_odbc use a single database connection per DSN. This way, user-facing database usage will have the same behavior as it did pre-13.8.0. However, realtime, which is the real workhorse of database interaction, will continue to let unixodbc manage connections. ASTERISK-25938 #close Reported by Edwin Vandamme Change-Id: Iac961fe79154c6211569afcdfec843c0c24c46dc	9 years ago
Leif Madsen	6aeefa89bc	Remove reference to non-existent sip.conf option Option was removed in commit `7f883ef495` ASTERISK-25927 #close Change-Id: I92f9b0196d9fc41d1d58354c07340c465ef1fcf8	9 years ago
Joshua Colp	6112a94d03	Merge "res_pjsip_callerid: Clear out display name if id->name is not valid" into 13	9 years ago
Diederik de Groot	e750ea9b5b	lock.c: Check lt before dereferencing it lt is NULL if t->tracking == 0 ASTERISK-25948 #close Change-Id: I4a81af28f9c82a74aa82413d772a7dc8fa6f45ba	9 years ago
Richard Mudgett	a036c35903	res_stasis: Handle re-enter stasis bridge with swap channel. We lose the fact that there is a swap channel if there is one. We currently wind up rejoining the stasis bridge as a normal join after the swap channel has already been kicked from the bridge. This patch preserves the swap channel so the AMI/ARI events can note that the channel joining the bridge is swapping with another channel. Another benefit to swaqpping in one operation is if there are any channels that get lonely (MOH, bridge playback, and bridge record channels). The lonely channels won't leave before the joining channel has a chance to come back in under stasis if the swap channel is the only reason the lonely channels are staying in the bridge. ASTERISK-25947 #close Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee	9 years ago
Richard Mudgett	9942d50aa5	bridge: Hold off more than one imparting channel at a time. An earlier patch blocked the ast_bridge_impart() call until the channel either entered the target bridge or it failed. Unfortuantely, if the target bridge is stasis and the imprted channel is not a stasis channel, stasis bounces the channel out of the bridge to come back into the bridge as a proper stasis channel. When the channel is bounced out, that released the block on ast_bridge_impart() to continue. If the impart was a result of a transfer, then it became a race to see if the swap channel would get hung up before the imparted channel could come back into the stasis bridge. If the imparted channel won then everything is fine. If the swap channel gets hung up first then the transfer will fail because the swap channel is leaving the bridge. * Allow a chain of ast_bridge_impart()'s to happen before any are unblocked to prevent the race condition described above. When the channel finally joins the bridge or completely fails to join the bridge then the ast_bridge_impart() instances are unblocked. ASTERISK-25947 Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1	9 years ago
Joshua Colp	b1b3460783	Merge "pjproject: Add patch for removing strip of '[]' from header params" into 13	9 years ago
George Joseph	516c626a7d	res_pjsip_callerid: Clear out display name if id->name is not valid When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning the From header, then it overwrites the display name and uri from the channel's connected.id. If the connected.id.name wasn't valid, create_new_id_hdr was leaving the display name from the From header in the new RPID or PAI header. On an attended transfer where the originator had a caller id number set but not a display name, the re-INVITE to the final transferee had the number of the originator but the display name of the transferer. Added a check to clear out the display name in the new header if connected.id.name was invalid. ASTERISK-25942 #close Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b	9 years ago
Joshua Colp	08f6408dc6	Merge "PJSIP: Remove PJSIP parsing functions from uri length validation." into 13	9 years ago
Joshua Colp	ded3794fc6	app_talkdetect: Make the module core supported. This module is used as part of testsuite tests to confirm stuff works. I'm accordingly marking it as core as it is required by those tests. Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88	9 years ago
Mark Michelson	efae187217	PJSIP: Remove PJSIP parsing functions from uri length validation. The PJSIP parsing functions provide a nice concise way to check the length of a hostname in a SIP URI. The problem is that in order to use those parsing functions, it's required to use them from a thread that has registered with PJLib. On startup, when parsing AOR configuration, the permanent URI handler may not be run from a PJLib-registered thread. Specifically, this could happen when Asterisk was started in daemon mode rather than console-mode. If PJProject were compiled with assertions enabled, then this would cause Asterisk to crash on startup. The solution presented here is to do our own parsing of the contact URI in order to ensure that the hostname in the URI is not too long. The parsing does not attempt to perform a full SIP URI parse/validation, since the hostname in the URI is what is important. ASTERISK-25928 #close Reported by Joshua Colp Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60	9 years ago
Joshua Colp	9a22ef81af	Merge "app_queue: Frequent segfaults in function can_ring_entry()" into 13	9 years ago
Joshua Colp	47adff8977	Merge "stasis_bridge.c: Update stasis bridge push diagnostic messages." into 13	9 years ago
Joshua Colp	a048a0ffbd	Merge "res_pjsip_transport_management: Allow unload to occur." into 13	9 years ago
Joshua Colp	c922846c6d	Merge "bridge_channel.c: Ignore role setup failure in channel push." into 13	9 years ago
Mark Michelson	f436b9ab11	res_pjsip_registrar: Fix bad memory-ness with user_agent. Recent changes to the PJSIP registrar resulted in tests failing due to missing AOR_CONTACT_ADDED test events. The reason for this was that the user_agent string had junk values in it, resulting in being unable to generate the event. I'm going to be honest here, I have no idea why this was happening. Here are the steps needed for the user_agent variable to get messed up: * REGISTER is received * First contact in the REGISTER results in a contact being removed * Second contact in the REGISTER results in a contact being added * The contact, AOR, expiration, and user agent all have to be passed as format parameters to the creation of a string. Any subset of those parameters would not be enough to cause the problem. Looking into what was happening, the thing that struck me as odd was that the user_agent variable was meant to be set to the value of the User-Agent SIP header in the incoming REGISTER. However, when removing a contact, the user_agent variable would be set (via ast_strdupa inside a loop) to the stored contact's user_agent. This means that the user_agent's value would be incorrect when attempting to process further contacts in the incoming REGISTER. The fix here is to use a different variable for the stored user agent when removing a contact. Correcting the behavior to be correct also means the memory usage is less weird, and the issue no longer occurs. ASTERISK-25929 #close Reported by Joshua Colp Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08	9 years ago
Joshua Colp	49bfdc9ac0	res_pjsip_transport_management: Allow unload to occur. At shutdown it is possible for modules to be unloaded that wouldn't normally be unloaded. This allows the environment to be cleaned up. The res_pjsip_transport_management module did not have the unload logic in it to clean itself up causing the res_pjsip module to not get unloaded. As a result the res_pjsip monitor thread kept going processing traffic and timers when it shouldn't. Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a	9 years ago
Richard Mudgett	f4693d1897	bridge_channel.c: Ignore role setup failure in channel push. We have to setup the channel roles after the bridge class push is called because the bridge class push callback may have set roles on the incoming channel. Since we have already partially pushed the channel into the bridge and reversing what we have already done could be problematic, the only thing we can do is press on to complete pushing the channel into the bridge. * Ignore any channel role setup errors after pushing the channel into a bridge. The channel may behave incorrectly in the bridge but we can no longer abort the push at this time. Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00	9 years ago
Jaco Kroon	22335fe18a	chan_sip: Don't verify table if rtupdate=no If rtupdate=no do not verify sipregs/peers table has updatable fields. ASTERISK-25934 #close Change-Id: Iaa2c53037b93daccc7e7333c40d61861847b856d	9 years ago
Joshua Colp	c7732a2600	Merge "Codecs: strip codec name while parsing allow/disallow options" into 13	9 years ago
ibercom	3b9d8b60b2	app_queue: Frequent segfaults in function can_ring_entry() ASTERISK-25888 #close Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117	9 years ago
Richard Mudgett	724acb6ce7	stasis_bridge.c: Update stasis bridge push diagnostic messages. Change-Id: I195b14994c9dcccb9452491ca20a885d2a54605a	9 years ago
Joshua Colp	56c8182913	Merge "app_voicemail/IMAP: function 'save_to_folder' creates wrong folder" into 13	9 years ago
Mark Michelson	5f78801859	transport management: Register thread with PJProject. The scheduler thread that kills idle TCP connections was not registering with PJProject properly and causing assertions if PJProject was built in debug mode. This change registers the thread with PJProject the first time that the scheduler callback executes. AST-2016-005 Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283	9 years ago
George Joseph	9740277713	res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c) There are several places that do scheduled tasks or periodic housecleaning, each with its own implementation: * res_pjsip_keepalive has a thread that sends keepalives. * pjsip_distributor has a thread that cleans up expired unidentified requests. * res_pjsip_registrar_expire has a thread that cleans up expired contacts. * res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task. * res_pjsip_sdp_rtp also uses ast_sched to send keepalives. There are also places where we should be doing scheduled work but aren't. A good example are the places we have sorcery observers to start registration or qualify. These don't work when changes are made to a backend database without a pjsip reload. We need to check periodically. As a first step to solving these issues, a new ast_sip_sched facility has been created. ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue. When a task is ready to run, ast_sip_task_pusk is called for it. This ensures that the task is executed in a PJLIB registered thread and doesn't hold up the ast_sched thread so it can immediately continue processing the queue. The serializer used by ast_sip_sched is one of your choosing or a random one from the res_pjsip pool if you don't choose one. Another feature is the ability to automatically clean up the task_data when the task expires (if ever). If it's an ao2 object, it will be dereferenced, if it's a malloc'd object it will be freed. This is selectable when the task is scheduled. Even if you choose to not auto dereference an ao2 task data object, the scheduler itself maintains a reference to it while the task is under it's control. This prevents the data from disappearing out from under the task. There are two scheduling models. AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at the specific interval. That is, every "interval" milliseconds, regardless of how long the task takes. If the task takes longer than the interval, it will be scheduled at the next available multiple of interval. For exmaple: If the task has an interval of 60 secs and the task takes 70 secs (it better not), the next invocation will happen at 120 seconds. AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should start "interval" milliseconds after the current invocation has finished. Also, the same ast_sched facility for fixed or variable intervals exists. The task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time. One res_pjsip.h housekeeping change was made. The pjsip header files were added to the top. There have been a few cases lately where I've needed res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because I didn't add the pjsip header files to my source even though I never referenced any pjsip calls. Finally, a few new convenience APIs were added to astobj2 to make things a little easier in the scheduler. ao2_ref_and_lock() calls ao2_ref() and ao2_lock() in one go. ao2_unlock_and_unref() does the reverse. A few macros were also copied from res_phoneprov because I got tired of having to duplicate the same hash, sort and compare functions over and over again. The AO2_STRING_FIELD_(HASH\|SORT\|CMP)_FN macros will insert functions suitable for aor_container_alloc into your source. This facility can be used immediately for the situations where we already have a thread that wakes up periodically or do some scheduled work. For the registration and qualify issues, additional sorcery and schema changes would need to be made so that we can easily detect changed objects on a periodic basis without having to pull the entire database back to check. I'm thinking of a last-updated timestamp on the rows but more on this later. Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c	9 years ago
Joshua Colp	13cb5ea73f	Merge "res_pjsip_transport_management: Kill idle TCP connections." into 13	9 years ago
Joshua Colp	120493d5c0	Merge "Rename res_pjsip_keepalive res_pjsip_transport_management" into 13	9 years ago
Joshua Colp	6c9c714bb6	Merge "AST-2016-004: Fix crash on REGISTER with long URI." into 13	9 years ago
Mark Michelson	7fb3724a77	res_pjsip_transport_management: Kill idle TCP connections. "Idle" here means that someone connects to us and does not send a SIP request. PJProject will not automatically time out such connections, so it's up to Asterisk to do it instead. When we receive an incoming TCP connection, we will start a timer (equivalent to transaction timer D) waiting to receive an incoming request. If we do not receive a request in that timeframe, then we will shut down the TCP connection. ASTERISK-25796 #close Reported by George Joseph AST-2016-005 Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6	9 years ago
Mark Michelson	707fd4dcd0	Rename res_pjsip_keepalive res_pjsip_transport_management ASTERISK-25796 Reported by George Joseph AST-2016-005 Change-Id: Id322a05f927392293570599730050bc677d99433	9 years ago
Mark Michelson	0b4bb19e0b	AST-2016-004: Fix crash on REGISTER with long URI. Due to some ignored return values, Asterisk could crash if processing an incoming REGISTER whose contact URI was above a certain length. ASTERISK-25707 #close Reported by George Joseph Patches: 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch AST-2016-004 Change-Id: I0ed3898fe7ab10121b76c8c79046692de3a1be55	9 years ago
Richard Mudgett	f6e080c6a4	bridge_softmix.c: Fix crash if could not allocate the dsp. Fix off nominal crash where we could not setup the channel to process frames for the softmix bridge technology because of allocation failure. Change-Id: Ic307a8386e46bf551e48fcd1eb97276714d56372	9 years ago

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