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Author	SHA1	Message	Date
Joshua Colp	3070dd0660	Merge "Fix when remote candidates exceed PJ_ICE_MAX_CAND" into 13	10 years ago
Matt Jordan	78d0b9d97e	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	61c6c6aa6c	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	ac62928d6b	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	6002472a62	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	d32e516c7c	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
Joshua Colp	2733638d0d	Merge "res_pjsip: Fix contact refleak on stateful responses." into 13	10 years ago
Mark Michelson	ad9cb6c2ce	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	cc1363209e	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	d58c8d73af	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
Joshua Colp	4e3b3f25f6	Merge "Fix deadlock on presence state changes." into 13	10 years ago
Mark Michelson	03fe79f29e	Fix deadlock on presence state changes. ... A deadlock was observed where three threads were competing for different locks: * One thread held the hints lock and was attempting to lock a specific hint. * One thread was holding the specific hint's lock and was attempting to lock the contexts lock * One thread was holding the contexts lock and attempting to lock the hints lock. Clearly the second thread was doing the wrong thing here. The fix for this is to make sure that the hint's lock is not held on presence state changes. Something similar is already done (and commented about) for device state changes. ASTERISK-25362 #close Reported by Mark Michelson Change-Id: I15ec2416b92978a4c0c08273b2d46cb21aff97e2	10 years ago
Matt Jordan	45b06a648b	Merge "taskprocessor: Fix race condition between unreferencing and finding." into 13	10 years ago
Matt Jordan	545795ae0d	Merge "res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items." into 13	10 years ago
Joshua Colp	a676ba2aad	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	1b1561f4c8	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	85e1cb51b2	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
Mark Michelson	b5801fe42c	Merge "res_pjsip_session: Don't invoke session supplements twice for BYE requests." into 13	10 years ago
Joshua Colp	c2df44ad3e	Merge "res_pjsip: Add common ast_sip_get_host_ip API." into 13	10 years ago
Mark Michelson	fdf096fff1	Merge "Chaos: make hangup NULL tolerant" into 13	10 years ago
Joshua Colp	c2c7319082	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
Scott Griepentrog	6862c2a167	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	f1cd636658	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	c01111223f	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	2a4eee0cd9	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	d238cf33a9	Merge "res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced" into 13	10 years ago
Mark Michelson	d0f911f326	Merge "bridge: Kick channel from bridge if hung up during action." into 13	10 years ago
Joshua Colp	5f781cb799	Merge "res_pjsip/pjsip_configuration: Disregard empty auth values" into 13	10 years ago
Joshua Colp	7c4d0c3506	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
Joshua Colp	6c2dab1e88	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
Matt Jordan	bc6fe07f5c	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
Joshua Colp	e0cf42f61e	Merge topic 'ASTERISK-25215' into 13 ... * changes: app_queue.c: Extract some functions for simpler code. app_queue.c: Fix error checking in QUEUE_MEMBER() read.	10 years ago
Mark Michelson	53c3dab8ed	Merge "contrib: script install_prereq should install sqlite3" into 13	10 years ago
Richard Mudgett	0582776f7f	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	504213f542	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	94d93e4d40	Merge "res_ari.c: Add missing off nominal unlock and remove a RAII_VAR()." into 13	10 years ago
Matt Jordan	1a5e087ecf	Merge "res_http_websocket.c: Fix some off nominal path cleanup." into 13	10 years ago
Richard Mudgett	77518d5434	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	c61547fee6	res_ari.c: Add missing off nominal unlock and remove a RAII_VAR(). ... Change-Id: I0c5e7b34057f26dadb39489c4dac3015c52f5dbf	10 years ago
Richard Mudgett	bd867cd078	app_queue.c: Extract some functions for simpler code. ... * Extract set_queue_member_pause() from set_member_paused() for simpler and more consistent code. * Extract set_queue_member_ringinuse() from set_member_ringinuse_help_members() for simpler code. Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306	10 years ago
Richard Mudgett	e5f5b9f384	app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'. ... Setting the 'paused' and 'ringinuse' options on a queue member using the dialplan function QUEUE_MEMBER did not behave the same way as the equivalent dialplan applications or AMI actions. * Made queue_function_mem_write() call the set_member_paused() and set_member_value() for the 'paused' and 'ringinuse' options respectively. A beneficial side effect is that the queue name is now optional and sets the value in all queues the interface is a member. * Update QUEUE_MEMBER XML documentation. * Fix error checking in QUEUE_MEMBER() write. ASTERISK-25215 #close Reported by: Lorne Gaetz Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb	10 years ago
Richard Mudgett	ded51e3d77	app_queue.c: Fix error checking in QUEUE_MEMBER() read. ... Change-Id: I7294e13d27875851c2f4ef6818adba507509d224	10 years ago
Scott Griepentrog	ab373f2cef	CHAOS: prevent sorcery object with null id ... When allocating a sorcery object, fail if the id value was not allocated. ASTERISK-25323 Reported by: Scott Griepentrog Change-Id: I152133fb7545a4efcf7a0080ada77332d038669e	10 years ago
Mark Michelson	b719f56c72	res_pjsip_sdp_rtp: Restore removed NULL check. ... When sending an RTP keepalive, we need to be sure we're not dealing with a NULL RTP instance. There had been a NULL check, but the commit that added the rtp_timeout and rtp_hold_timeout options removed the NULL check. Change-Id: I2d7dcd5022697cfc6bf3d9e19245419078e79b64	10 years ago
Richard Mudgett	cea5dc7b8a	audiohook.c: Simplify variable usage in audiohook_read_frame_both(). ... Change-Id: I58bed58631a94295b267991c5b61a3a93c167f0c	10 years ago
Richard Mudgett	b3a56bee83	audiohook.c: Fix MixMonitor crash when using the r() or t() options. ... The built frame format in audiohook_read_frame_both() is now set to a signed linear format before the rx and tx frames are duplicated instead of only for the mixed audio frame duplication. ASTERISK-25322 #close Reported by Sean Pimental Change-Id: I86f85b5c48c49e4e2d3b770797b9d484250a1538	10 years ago
Mark Michelson	8f3117ebaf	Merge "chan_sip.c: wrong peer searched in sip_report_security_event" into 13	10 years ago
Kevin Harwell	25af2d71c8	chan_sip.c: wrong peer searched in sip_report_security_event ... In chan_sip, after handling an incoming invite a security event is raised describing authorization (success, failure, etc...). However, it was doing a lookup of the peer by extension. This is fine for register messages, but in the case of an invite it may search and find the wrong peer, or a non existent one (for instance, in the case of call pickup). Also, if the peers are configured through realtime this may cause an unnecessary database lookup when caching is enabled. This patch makes it so that sip_report_security_event searches by IP address when looking for a peer instead of by extension after an invite is processed. ASTERISK-25320 #close Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4	10 years ago
Joshua Colp	e18c300550	res_http_websocket: When shutting down a session don't close closed socket ... Due to the use of ast_websocket_close in session termination it is possible for the underlying socket to already be closed when the session is terminated. This occurs when the close frame is attempted to be written out but fails. Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b	10 years ago
Joshua Colp	0bc6fe6f7d	Merge "res_http_websocket: Forcefully terminate on write errors." into 13	10 years ago