26977 Commits (fcea6910f60760527d874465f65cb617bc6f88a2)

 

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Author	SHA1	Message	Date
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
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	a79527ca64	Merge "endpoint snapshot: avoid second cleanup on alloc failure"	10 years ago
Matt Jordan	b16c7ef0ed	Merge "channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id"	10 years ago
Joshua Colp	e1c43223ab	Merge "res_pjsip: Change default from user value."	10 years ago
Joshua Colp	bf74956371	Merge "Fix when remote candidates exceed PJ_ICE_MAX_CAND"	10 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	10 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	10 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	10 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	10 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	10 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	10 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	10 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	10 years ago
Matt Jordan	e23a01acbb	Merge "taskprocessor: Fix race condition between unreferencing and finding."	10 years ago
Matt Jordan	fdc287ea3a	Merge "res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items."	10 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	10 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	10 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	10 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	10 years ago
Joshua Colp	388e628120	Merge "res_pjsip: Add common ast_sip_get_host_ip API."	10 years ago
Mark Michelson	4a540721d1	Merge "Chaos: make hangup NULL tolerant"	10 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	10 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	10 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	10 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	10 years ago
Mark Michelson	6b8734fe68	Merge "res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced"	10 years ago
Mark Michelson	d36c8aa006	Merge "bridge: Kick channel from bridge if hung up during action."	10 years ago
Joshua Colp	3108820000	Merge "res_pjsip/pjsip_configuration: Disregard empty auth values"	10 years ago
Joshua Colp	98d089fb9a	bridge: Kick channel from bridge if hung up during action. ... When executing an action in a bridge it is possible for the channel to be hung up without the bridge becoming aware of it. This is most easily reproducible by hanging up when the bridge is streaming DTMF due to a feature timeout. This change makes it so after action execution the channel is checked to determine if it has been hung up and if it has it is kicked from the bridge. ASTERISK-25341 #close Change-Id: I6dd8b0c3f5888da1c57afed9e8a802ae0a053062	10 years ago
Joshua Colp	a408369bac	res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced ... When recreating a subscription it is possible for a freed sub_tree to be referenced when the initial NOTIFY fails to be created. Change-Id: I681c215309aad01b21d611c2de47b3b0a6022788	10 years ago
Matt Jordan	3af34441eb	res_pjsip/pjsip_configuration: Disregard empty auth values ... When an endpoint is backed by a non-static conf file backend (such as the AstDB or Realtime), the 'auth' object may be returned as being an empty string. Currently, res_pjsip will interpret that as being a valid auth object, and will attempt to authenticate inbound requests. This isn't desired; is an auth value is empty (which the name of an auth object cannot be), we should instead interpret that as being an invalid auth object and skip it. ASTERISK-25339 #close Change-Id: Ic32b0c6eb5575107d5164a8c40099e687cd722c7	10 years ago
Rodrigo Ramírez Norambuena	89003ea320	README*: Remove trailing whitespace ... Change-Id: I18b7d75187548a9ed55b4f258d21aaaf29d08874	10 years ago
Richard Mudgett	857923d9c7	chan_sip.c: Set preferred rx payload type mapping on incoming offers. ... ASTERISK-25166 Reported by: Kevin Harwell ASTERISK-17410 Reported by: Boris Fox Change-Id: I7f04d5c8bee1126fee5fe6afbc39e45104469f4e	10 years ago
Richard Mudgett	d643b206c6	res_pjsip_sdp_rtp.c: Set preferred rx payload type mapping on incoming offers. ... ASTERISK-25166 Reported by: Kevin Harwell ASTERISK-17410 Reported by: Boris Fox Change-Id: I97ecebc1ab9b5654fb918bf1f4c98c956b852369	10 years ago
Richard Mudgett	f7df3e1a01	rtp_engine.c: Get current or create a needed rx payload type mapping. ... * Make ast_rtp_codecs_payload_code() get the current mapping or create a rx payload type mapping. ASTERISK-25166 Reported by: Kevin Harwell ASTERISK-17410 Reported by: Boris Fox Change-Id: Ia4b2d45877a8f004f6ce3840e3d8afe533384e56	10 years ago
Richard Mudgett	38854a9f7b	rtp_engine.c: Extract rtp_codecs_payload_replace_rx(). ... ASTERISK-25166 Reported by: Kevin Harwell ASTERISK-17410 Reported by: Boris Fox Change-Id: I34e23bf5b084c8570f9c3e6ccd19b95fe85af239	10 years ago
Richard Mudgett	1a549ed134	rtp_engine.c: Initial split of payload types into rx and tx mappings. ... There are numerous problems with the current implementation of the RTP payload type mapping in Asterisk. It uses only one mapping structure to associate payload types to codecs. The single mapping is overkill if all of the payload type values are well known values. Dynamic payload type mappings do not work as well with the single mapping because RFC3264 allows each side of the link to negotiate different dynamic mappings for what they want to receive. Not only could you have the same codec mapped for sending and receiving on different payload types you could wind up with the same payload type mapped to different codecs for each direction. 1) An independent payload type mapping is needed for sending and receiving. 2) The receive mapping needs to keep track of previous mappings because of the slack to when negotiation happens and current packets in flight using the old mapping arrive. 3) The transmit mapping only needs to keep track of the current negotiated values since we are sending the packets and know when the switchover takes place. * Needed to create ast_rtp_codecs_payload_code_tx() and make some callers use the new function because ast_rtp_codecs_payload_code() was used for mappings in both directions. * Needed to create ast_rtp_codecs_payloads_xover() for cases where we need to pass preferred codec mappings to the peer channel for early media bridging or when we need to prefer the offered mapping that RFC3264 says we SHOULD use. * ast_rtp_codecs_payloads_xover() and ast_rtp_codecs_payload_code_tx() are the only new public functions created. All the others were only used for the tx or rx mapping direction so the function doxygen now reflects which direction the function operates. * chan_mgcp.c: Removed call to ast_rtp_codecs_payloads_clear() as doing that makes no sense when processing an incoming SDP. We would be wiping out any mappings that we set for the possible outgoing SDP we sent earlier. ASTERISK-25166 Reported by: Kevin Harwell ASTERISK-17410 Reported by: Boris Fox Change-Id: Iaf6c227bca68cb7c414cf2fd4108a8ac98bd45ac	10 years ago
Mark Michelson	aacb46b56a	Merge "res_ari_events: Fix shutdown ref leak."	10 years ago
Mark Michelson	26fca72837	Merge "app_queue.c: Extract some functions for simpler code."	10 years ago
Mark Michelson	192693c2c1	Merge "res_http_websocket.c: Add missing unref on an off nominal path."	10 years ago
Mark Michelson	5d6b93a006	Merge "ari/ari_websockets.c: Fix ast_debug parameter type mismatch."	10 years ago
Richard Mudgett	21d419e4fc	ari/ari_websockets.c: Fix ast_debug parameter type mismatch. ... This is a type mismatch fix of the debugging commit c63316eec1 made to find out why a testsuite test was failing only on one of the continuous integration build agents. Change-Id: Iba34f6e87cec331f6ac80e4daff6476ea6f00a75	10 years ago
Scott Griepentrog	53e2a6a829	contrib: script install_prereq should install sqlite3 ... Asterisk needs the sqlite 3 library, which is package sqlite-devel in CentOS. By adding this package to the script, a problem with configure failing is resolved. ASTERISK-25331 #close Reported by: Kevin Harwell Change-Id: I90efaf6a01914fea03f21e5cdbd91c348f44b0ec	10 years ago
Matt Jordan	a260cee5cb	Merge "res_http_websocket.c: Fix some off nominal path cleanup."	10 years ago
Matt Jordan	2d20379f8e	Merge "app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'."	10 years ago
Richard Mudgett	03eb6cbc10	res_ari_events: Fix shutdown ref leak. ... ASTERISK-25308 #close Reported by: Joshua Colp Change-Id: I592785bf70ff4b63d00e535b482f40da8e82a082	10 years ago
Richard Mudgett	e1e7e205bc	res_http_websocket.c: Add missing unref on an off nominal path. ... Change-Id: I228df6adecd4cb450d03e09e9a38c86bb566e811	10 years ago
Richard Mudgett	59253a2262	res_http_websocket.c: Fix some off nominal path cleanup. ... * Remove extraneous unlock on off-nominal path. * Add missing HTTP error reply. Change-Id: I1f402bfe448fba8696b507477cab5f060ccd9b2b	10 years ago
Richard Mudgett	1f0a9f8a76	res_ari.c: Add missing off nominal unlock and remove a RAII_VAR(). ... Change-Id: I0c5e7b34057f26dadb39489c4dac3015c52f5dbf	10 years ago