asterisk

Commit Graph

Author	SHA1	Message	Date
Diederik de Groot	a2f19d82a8	configs/basic-pbx/asterisk.conf: contains incorrect path separator Note: When packagers use these files (as an example) the paths are never really used when they are split using '='. Note: Thirdparty applications will also have trouble parsing the file when expecting '=>'. Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004	9 years ago
zuul	69ced86c6c	Merge "pjproject_bundled: Disable PJSIP_UNESCAPE_IN_PLACE" into 13	9 years ago
Joshua Colp	d65023b5a5	Merge "res_pjsip: Start body generator users after suppliers." into 13	9 years ago
zuul	3e5666eadc	Merge "res_pjsip_pubsub.c: Fix body generator registration race." into 13	9 years ago
zuul	72ee8c1bf9	Merge "chan_sip: Make autocreated peers send PeerStatus events" into 13	9 years ago
Joshua Colp	1ce30f1fb5	Merge "res_pjsip_outbound_publish.c: Remove redundant flag check." into 13	9 years ago
zuul	cc8a50631e	Merge "res_pjsip_pubsub.c: Add useful information to some messages." into 13	9 years ago
zuul	2ae097f144	Merge "res_pjsip_pubsub.h: Fix doxygen association." into 13	9 years ago
zuul	28e6978e82	Merge "res_pjsip: Add ability to identify by Authorization username" into 13	9 years ago
Richard Mudgett	7992923c70	res_pjsip: Start body generator users after suppliers. Change-Id: I8f0b57841feaab56c8a4e821b5ccb4e05e5fbadb	9 years ago
Richard Mudgett	5dc0e082b2	res_pjsip_pubsub.c: Add useful information to some messages. Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a	9 years ago
Richard Mudgett	f9e416f053	res_pjsip_pubsub.c: Fix body generator registration race. Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67	9 years ago
Richard Mudgett	b1b2019046	res_pjsip_pubsub.h: Fix doxygen association. Change-Id: I110d3e3572598289fcd4215d966cf0c858f98632	9 years ago
Richard Mudgett	b7f07fdff5	res_pjsip_outbound_publish.c: Remove redundant flag check. Change-Id: I0da80a3c3e0eae0c52ff27e7412ba027d6f52353	9 years ago
George Joseph	719ece5659	pjproject_bundled: Disable PJSIP_UNESCAPE_IN_PLACE When pjsip_parse_uri is called with PJSIP_UNESCAPE_IN_PLACE enabled, the input uri string will become corrupted if it contains escape sequences. It's not possible to automatically strdup or strdupa the input string because the output uri pj_str_t's will have pointers to chunks of the input string. Getting around this would require more memory management code and wouldn't be worth the savings of doing the unescape in place. ASTERISK-25970 #close Reported-by: Dmitriy Serov Change-Id: I28dc0e599b5108f7959b9c46dc8278371b372f88	9 years ago
Joshua Colp	86fb803257	Merge "func_odbc: Check connection status before executing queries." into 13	9 years ago
George Joseph	38bed4515d	res_pjsip: Add ability to identify by Authorization username A feature of chan_sip that service providers relied upon was the ability to identify by the Authorization username. This is most often used when customers have a PBX that needs to register rather than identify by IP address. From my own experiance, this is pretty common with small businesses who otherwise don't need a static IP. In this scenario, a register from the customer's PBX may succeed because From will usually contain the PBXs account id but an INVITE will contain the caller id. With nothing recognizable in From, the service provider's Asterisk can never match to an endpoint and the INVITE just stays unauthorized. The fixes: A new value "auth_username" has been added to endpoint/identify_by that will use the username and digest fields in the Authorization header instead of username and domain in the the From header to match an endpoint, or the To header to match an aor. This code as added to res_pjsip_endpoint_identifier_user rather than creating a new module. Although identify_by was always a comma-separated list, there was only 1 choice so order wasn't preserved. So to keep the order, a vector was added to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar to find the aor. The res_pjsip_endpoint_identifier_* modules are called in globals/endpoint_identifier_order. Along the way, the logic in res_pjsip_registrar was corrected to match most-specific to least-specific as res_pjsip_endpoint_identifier_user does. The order is: username@domain username@domain_alias username Auth by username does present 1 problem however, the first INVITE won't have an Authorization header so the distributor, not finding a match on anything, sends a securty_alert. It still sends a 401 with a challenge so the next INVITE will have the Authorization header and presumably succeed. As a result though, that first security alert is actually a false alarm. To address this, a new feature has been added to pjsip_distributor that keeps track of unidentified requests and only sends the security alert if a configurable number of unidentified requests come from the same IP in a configurable amout of time. Those configuration options have been added to the global config object. This feature is only used when auth_username is enabled. Finally, default_realm was added to the globals object to replace the hard coded "asterisk" used when an endpoint is not yet identified. The testsuite tests all pass but new tests are forthcoming for this new feature. ASTERISK-25835 #close Reported-by: Ross Beer Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d	9 years ago
Joshua Colp	a16aa467f5	Merge "config: Fix ast_config_text_file_save2 writability check for missing files" into 13	9 years ago
Joshua Colp	e0e03cd2c8	Merge "res_pjsip: disable multi domain to improve realtime performace" into 13	9 years ago
Mark Michelson	677d5b5151	func_odbc: Check connection status before executing queries. A recent change to func_odbc made it so that a single connection was maintained per DSN. The problem was that the code was optimistic about the health of the connection after initially opening it and did nothing to re-connect in case the connection had died. This change adds a check before executing a query to ensure that the connection to the database is still up and running. ASTERISK-25963 #close Reported by Ross Beer Change-Id: Id33c86eb04ff48ca088bb2e3086c27b3b683491d	9 years ago
Alexei Gradinari	df3639700a	res_pjsip: disable multi domain to improve realtime performace This patch added new global pjsip option 'disable_multi_domain'. Disabling Multi Domain can improve Realtime performance by reducing number of database requests. ASTERISK-25930 #close Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7	9 years ago
Joshua Colp	e5c5fac8f1	Merge "res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c)" into 13	9 years ago
zuul	b4ddcc0e94	Merge "chan_sip: Give more time for TCP/TLS threads to stop." into 13	9 years ago
Joshua Colp	949bf6b282	chan_sip: Give more time for TCP/TLS threads to stop. The unload process currently tells each TCP/TLS to terminate but does not wait for them to do so. This introduces a race condition where the container holding the threads may be destroyed before the threads are able to remove themselves from it. When they finally do the container is invalid and can't be used causing a crash. A previous change existed which waited a bit to wait for any stranglers to finish. This change extends this and waits longer. ASTERISK-25961 #close Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6	9 years ago
Joshua Colp	6959f5484b	app_queue: Fix crash when unloading module. When unloading the app_queue module the members in each queue are destroyed and as part of this they are removed from the pending members container. Unfortunately a crash would occur as the container was destroyed before the members were removed. This change tweaks ordering so the container destruction occurs after the members are destroyed. ASTERISK-16115 Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b	9 years ago
Joshua Colp	9d8f59aaf2	Merge changes from topic 'system_stress_patches' into 13 * changes: test_message.c: Wait longer in case dialplan also processes the test message. Manager: Short circuit AMI message processing. manager.c: Eliminate most RAII_VAR usage. manager_channels.c: Fix allocation failure crash.	9 years ago
Joshua Colp	4efc6b4315	Merge changes from topic 'system_stress_patches' into 13 * changes: Bridge system: Fix memory leaks and double frees on impart failure. bridge_softmix.c: Fix crash if channel fails to join mixing tech.	9 years ago
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

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