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Author	SHA1	Message	Date
Richard Mudgett	9b290dfe2f	res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response. ... We should not try to send a SIP response message because we may be restoring a persistent subscription where we are not responding to a SIP request. Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec	10 years ago
Richard Mudgett	73eb132012	res_pjsip_pubsub.c: Fix off-nominal memory leak. ... Fix off-nominal visited vector leak in build_resource_tree(). Change-Id: If0399c7941c9c0b1038bcfb7b9a371760977831c	10 years ago
Richard Mudgett	2b30fc2b2d	res_pjsip_pubsub.c: Fix one byte buffer overrun error. ... ast_sip_pubsub_register_body_generator() did not account for the null terminator set by sprintf() in the allocated output buffer. Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2	10 years ago
Richard Mudgett	08a182c8e6	res_pjsip_pubsub.c: Use ast_alloca() instead of alloca(). ... Change-Id: Ia396096b4fedc2874649ca11137612c3f55e83e3	10 years ago
Richard Mudgett	61f30db877	res_pjsip_pubsub.c: Add missing error return in load_module(). ... Change-Id: I15debd0f717f16ee2f78e7f56151c3b3b97b72fc	10 years ago
Richard Mudgett	b8f07527b2	res_pjsip/location.c: Use the builtin ao2_callback() match function instead. ... Change-Id: I364906d6d2bad3472929986704a0286b9a2cbe3f	10 years ago
Mark Michelson	f1a2e82d49	res_pjsip: Copy default_from_user to avoid crash. ... The default_from_user retrieval function was pulling the default_from_user from the global configuration struct in an unsafe way. If using a database as a backend configuration store, the global configuration struct is short-lived, so grabbing a pointer from it results in referencing freed memory. The fix here is to copy the default_from_user value out of the global configuration struct. Thanks go to John Hardin for discovering this problem and proposing the patch on which this fix is based. ASTERISK-25390 #close Reported by Mark Michelson Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c	10 years ago
Matt Jordan	bd71dcd1da	res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route ... We will only rewrite the Contact header if there is no Record-Route header in the received request. If a malfunctioning proxy places a Record-Route header into a REGISTER request, we will decide that we shouldn't update the IP/port in the Contact header, and we will end up storing a contact with an AoR that contains the NAT'd IP address. While it is nice to have the proxy *not* send a Record-Route in a REGISTER request, it's also a good idea to not process the header in a non-dialog message. This patch updates the code to explicitly ignore the Record-Route header in REGISTER requests. ASTERISK-25387 #close Change-Id: I4bd3bcccc4003d460cc354d986b0dea2e433ef3f	10 years ago
Joshua Colp	be3f52a122	Merge "ParkAndAnnounce: Add variable inheritance"	10 years ago
Matt Jordan	0cc8d6f99d	Merge "chan_ooh323: Add ProgressIndicator IE with inband info available"	10 years ago
Joshua Colp	8e269a467d	Merge "pjsip: avoid possible crash req_caps allocation failure"	10 years ago
Alexander Anikin	5bd363010e	chan_ooh323: Add ProgressIndicator IE with inband info available ... Add ProgressIndicator IE with inband info present to Progress and Alerting Q.931 message ASTERISK-25227 #close Reported by: Alexandr Dranchuk Change-Id: I326ad13cb1db9a72b3fd902bafed3c28a3684203	10 years ago
Scott Griepentrog	fcea6910f6	pjsip: avoid possible crash req_caps allocation failure ... Make certain that the pjsip session has not failed to allocate the format capabilities structure, which can otherwise cause a crash when referenced. ASTERISK-25323 Change-Id: I602790ba12714741165e441cc64a3ecde4cb5750	10 years ago
Joshua Colp	647cdcd6a8	Merge "res_pjsip: Use hash for contact object identity instead of Contact URI."	10 years ago
Matt Jordan	0b63c2969f	Merge "res_rtp_asterisk: Add more ICE debugging"	10 years ago
Joshua Colp	fdf77633ed	Merge "Core/General: Add #ifdef needed on FreeBSD."	10 years ago
David M. Lee	8e5ed27a16	res_rtp_asterisk: Add more ICE debugging ... In working through a recent ICE negotiation bug, I found the debug logging in res_rtp_asterisk to be lacking. This patch adds a number of debug and warning statements that were helpful. Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80	10 years ago
Joshua Colp	820b9dc138	Merge "res/res_pjsip: Purge contacts when an AoR is deleted"	10 years ago
Joshua Colp	3628e380b8	res_pjsip: Use hash for contact object identity instead of Contact URI. ... In the wild it is possible for Contact URIs to be quite long as parameters can exist on them. This can present a problem when storing them in the AstDB as the URI is used as part of the object name and there is a fixed length limit for the AstDB. This will cause the contact to not get stored. This change uses the MD5 hash of the Contact URI as part of the object name instead. This has a fixed length which is guaranteed to not exceed the AstDB length limit. ASTERISK-25295 #close Change-Id: Ie8252a75331ca00b41b9f308f42cc1fbdf701a02	10 years ago
Alexander Anikin	d2106c0b21	chan_ooh323: call ast_rtp_instance_stop on ooh323_destroy ... Call ast_rtp_instance_stop on ooh323_destroy to free resources allocated by rtp instance ASTERISK-25299 #close Report by: Alexandr Dranchuk Change-Id: I455096bd7da016b871afe90af86067c2c7c9f33f	10 years ago
Matt Jordan	ef3358d0c0	res/res_pjsip: Purge contacts when an AoR is deleted ... When an AoR is deleted by an external mechanism, such as through ARI, we currently do not remove dynamic contacts that were created for that AoR as a result of a received REGISTER request. As a result, re-creating the AoR will cause the dynamic contact to be interpreted as a persistent contact, leading to some rather strange state being created for the contacts/endpoints. This patch adds a sorcery observer for the 'aor' object. When a delete is issued on the underlying sorcery object, the observer is called, and all contacts created and persisted in sorcery for that AoR are also removed. Note that we don't want to perform this action when an AO2 object that is an AoR is destroyed, as the AoR can still exist in the backing storage (and we would thus be removing valid contacts from an AoR that still "exists".) ASTERISK-25381 #close Change-Id: I6697e51ef6b2858b5d63401f35dc378bb0f90328	10 years ago
Matt Jordan	a79527ca64	Merge "endpoint snapshot: avoid second cleanup on alloc failure"	11 years ago
Matt Jordan	b16c7ef0ed	Merge "channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id"	11 years ago
Joshua Colp	e1c43223ab	Merge "res_pjsip: Change default from user value."	11 years ago
Joshua Colp	bf74956371	Merge "Fix when remote candidates exceed PJ_ICE_MAX_CAND"	11 years ago
Matt Jordan	86b02228f5	channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id ... This patch adds a new option to the CHANNEL function that allows for the extraction of the SIP call-id. It is used in conjunction with the 'pjsip' option, and will return the Call-ID of the INVITE request that established the PJSIP channel. ASTERISK-25352 Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a	11 years ago
David M. Lee	27c89053b0	Fix when remote candidates exceed PJ_ICE_MAX_CAND ... We were passing the wrong count into pj_ice_sess_create_check_list(), causing the create to fail if we ever received more than PJ_ICE_MAX_CAND candidates. Change-Id: I0303d8e1ecb20a8de9fe629a3209d216c4028378	11 years ago
Mark Michelson	993ae9a669	res_pjsip: Change default from user value. ... When Asterisk sends an outbound SIP request, if there is no direct reason to place a specific value for the username in the From header, Asterisk would generate a UUID. For example, this would happen when sending outbound OPTIONS requests when qualifying or when sending outbound INVITE requests when originating (if no explicit caller ID were provided). The issue is that some SIP providers reject these sorts of requests with a "Name too long" error response. This patch aims to fix this by changing the default outbound username in From headers to "asterisk". This value can be overridden by changing the default_from_user option in the global options if desired. ASTERISK-25377 #close Reported by Mark Michelson Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190	11 years ago
Jonathan Rose	7d981b787c	ParkAndAnnounce: Add variable inheritance ... In Asterisk 11, the announcer channel would receive channel variables from the channel being parked by means of normal channel inheritance. This functionality was lost during the big res_parking project in Asterisk 12. This patch restores that functionality. ASTERISK-25369 #close Review: https://gerrit.asterisk.org/#/c/1180/ Change-Id: Ie47e618330114ad2ea91e2edcef1cb6f341eed6e	11 years ago
Scott Griepentrog	7691035312	endpoint snapshot: avoid second cleanup on alloc failure ... In ast_endpoint_snapshot_create(), a failure to init the string fields results in two attempts to ao2_cleanup the same pointer. Removed RAII_VAR to eliminate problem. ASTERISK-25375 #close Reported by: Scott Griepentrog Change-Id: If4d9dfb1bbe3836b623642ec690b6d49b25e8979	11 years ago
Martin Tomec	be31747db8	res/pjsip: Mark WSS transport as secure ... Pjsip is refusing to use unsecure transport with "sips" in url. WSS should be considered as secure transport. ASTERISK-24602 #comment Partially fixed by setting WSS as secure Change-Id: Iddac406c6deba6240c41a603b8859dfefe1a5353	11 years ago
Guido Falsi	fbdb42c9fc	Core/General: Add #ifdef needed on FreeBSD. ... pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED on FreeBSD too. ASTERISK-25310 #close Reported by: Guido Falsi Change-Id: Iae6befac9028b5b9795f86986a4a08a1ae6ab7c4	11 years ago
Mark Michelson	c15d8cc0ed	res_pjsip: Fix contact refleak on stateful responses. ... When sending a stateful response, creation of the transaction can fail, most commonly because we are trying to create a transaction from a retransmitted request. When creation of the transaction fails, we end up leaking a reference to a contact that was bumped when the response was created. This patch adds the missing deref and fixes the reference leak. Change-Id: I2f97ad512aeb1b17e87ca29ae0abacb4d6395f07	11 years ago
Joshua Colp	b51cf1e712	pbx: Fix crash when issuing "core show hints" with long pattern match. ... When issuing the "core show hints" CLI command a combination of both the hint extension and context is created. This uses a fixed size buffer expecting that the extension will not exceed maximum extension length. When the extension is actually a pattern match this constraint does not hold true, and the extension may exceed the maximum extension length. In this case extra characters are written past the end of the fixed size buffer. This change makes it so the construction of the combined hint extension and context can not exceed the size of the buffer. ASTERISK-25367 #close Change-Id: Idfa1b95d0d4dc38e675be7c1de8900b3f981f499	11 years ago
Mark Michelson	beb568e51c	res_pjsip_pubsub: re-re-fix persistent subscription storage. ... A recent change to res_pjsip_pubsub switched to using pjsip_msg_print as a means of writing an appropriate packet to persistent storage. While this partially solved the issue, it had its own problems. pjsip_msg_print will always add a Content-Length header to the message it prints. Frequent restarts of Asterisk can result in persistent subscriptions being written with five or more Content-Length headers. In addition, sometimes some apparent corruption of individual headers could be seen. This aims to fix the problem by not running a parsed message through an interpreter but rather by taking the raw message and saving it. The logic for what to save is going to be different depending on whether a SUBSCRIBE was received from the wire or if it was pulled from persistence. When receiving a packet from the wire, when using a streaming transport, the rdata->pkt_info.packet may contain multiple SIP messages or fragments. However, the rdata->msg_info.msg_buf will always contain the current SIP message to be processed. When pulling from persistence, though, the rdata->msg_info.msg_buf will be NULL since no transport actually handled the packet. However, since we know that we will always ever pull one SIP message from persistence, we are free to save directly from rdata->pkt_info.packet instead. ASTERISK-25365 #close Reported by Mark Michelson Change-Id: I33153b10d0b4dc8e3801aaaee2f48173b867855b	11 years ago
Matt Jordan	e23a01acbb	Merge "taskprocessor: Fix race condition between unreferencing and finding."	11 years ago
Matt Jordan	fdc287ea3a	Merge "res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items."	11 years ago
Joshua Colp	fc4d4f5379	taskprocessor: Fix race condition between unreferencing and finding. ... When unreferencing a taskprocessor its reference count is checked to determine if it should be unlinked from the taskprocessors container and its listener shut down. In between the time when the reference count is checked and unlinking it is possible for another thread to jump in, find it, and get a reference to it. If the thread then uses the taskprocessor it may find that it is not in the state it expects. This change locks the taskprocessors container during almost the entire unreference operation to ensure that any other thread which may attempt to find the taskprocessor has to wait. ASTERISK-25295 Change-Id: Icb842db82fe1cf238da55df92e95938a4419377c	11 years ago
Joshua Colp	bb38010c67	res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items. ... The keepalive support in res_pjsip_sdp_rtp currently assumes that a stream will only be negotiated once. This is false. If the stream is replaced and later added back it can be negotiated again causing multiple keepalive scheduled items to exist. This change explicitly deletes the existing keepalive scheduled item before adding the new one. The res_pjsip_sdp_rtp module also does not stop RTP keepalives or timeout timer if the stream has been replaced. This change adds a callback to the session media interface to allow a media stream to be stopped without the resources being destroyed. This allows the scheduled items and RTP to be stopped when the stream no longer exists. ASTERISK-25356 #close Change-Id: Ibe6a7cc0927c87326fd5f1c0d4ad889dbfbea1de	11 years ago
Joshua Colp	c036e50fbe	sched: ast_sched_del may return prematurely due to spurious wakeup ... When deleting a scheduled item if the item in question is currently executing the ast_sched_del function waits until it has completed. This is accomplished using ast_cond_wait. Unfortunately the ast_cond_wait function can suffer from spurious wakeups so the predicate needs to be checked after it returns to make sure it has really woken up as a result of being signaled. This change adds a loop around the ast_cond_wait to make sure that it only exits when the executing task has really completed. ASTERISK-25355 #close Change-Id: I51198270eb0b637c956c61aa409f46283432be61	11 years ago
Joshua Colp	229b95d253	res_pjsip_session: Don't invoke session supplements twice for BYE requests. ... When a BYE request is received the PJSIP invite session implementation creates and sends a 200 OK response before we are aware of it. This causes the INVITE session state callback to be called into and ultimately the session supplements run on the BYE request. Once this response has been sent the normal transaction state callback is invoked which invokes the session supplements on the BYE request again. This can be problematic in particular with res_pjsip_rfc3326 as it may attempt to update the hangup cause code on the channel while it is in the process of being hung up. This change makes it so the session supplements are only invoked once by the INVITE session state callback. ASTERISK-25318 #close Change-Id: I69c17df55ccbb61ef779ac38cc8c6b411376c19a	11 years ago
Joshua Colp	388e628120	Merge "res_pjsip: Add common ast_sip_get_host_ip API."	11 years ago
Mark Michelson	4a540721d1	Merge "Chaos: make hangup NULL tolerant"	11 years ago
Scott Griepentrog	6bfa14bdad	Chaos: handle failed allocation in get_media_encryption_type ... If the ast_strndup() call fails to allocate a copy of the transport string for parsing, fail gracefully. ASTERISK-25323 Reported by: Scott Griepentrog Change-Id: Ia4b905ce6d03da53fea526224455c1044b1a5a28	11 years ago
Scott Griepentrog	490db8ba94	Chaos: make hangup NULL tolerant ... In chan_pjsip_new, if allocation of the pvt structure fails, ast_hangup is called. But it was written to assume pvt was valid, and this change corrects that. ASTERISK-25323 Reported by: Scott Griepentrog Change-Id: I5f47860fe9cee4cd56abd3f79b108678ab72cc87	11 years ago
Joshua Colp	d03d09aad3	chan_sip: Allow call pickup to set the hangup cause. ... The call pickup implementation in chan_sip currently sets the channel hangup cause to "normal clearing" if call pickup is successfully performed. This action overwrites the "answered elsewhere" hangup cause set by the call pickup code and can result in the SIP device in question showing a missed call when it should not. This change sets the hangup cause to "normal clearing" as a default initially but allows the call pickup to change it as needed. ASTERISK-25346 #close Change-Id: I00ac2c269cee9e29586ee2c65e83c70e52a02cff	11 years ago
Joshua Colp	d013ecf748	res_pjsip: Add common ast_sip_get_host_ip API. ... Modules commonly used the pj_gethostip function for retrieving the IP address of the host. This function does not cache the result and may result in a DNS lookup occurring, or additional work. If the DNS server is unreachable or network issues arise this can cause the pj_gethostip function to block for a period of time. This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string function which does the same thing but caches the host IP address at module load time. This results in no additional work being done each time the local host IP address is needed. ASTERISK-25342 #close Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e	11 years ago
Mark Michelson	6b8734fe68	Merge "res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced"	11 years ago
Mark Michelson	d36c8aa006	Merge "bridge: Kick channel from bridge if hung up during action."	11 years ago
Joshua Colp	3108820000	Merge "res_pjsip/pjsip_configuration: Disregard empty auth values"	11 years ago