26963 Commits (e23a01acbbef147824be26189d67b454663bf60c)

 

All Branches   

Search

Author	SHA1	Message	Date
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
Richard Mudgett	9fb4a96e15	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	87b22969a4	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	5cf98e2459	app_queue.c: Fix error checking in QUEUE_MEMBER() read. ... Change-Id: I7294e13d27875851c2f4ef6818adba507509d224	10 years ago
Scott Griepentrog	178e1adffb	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	5a85711568	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	7c4cb8618d	audiohook.c: Simplify variable usage in audiohook_read_frame_both(). ... Change-Id: I58bed58631a94295b267991c5b61a3a93c167f0c	10 years ago
Richard Mudgett	bb37473234	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	f96ce0de87	Merge "chan_sip.c: wrong peer searched in sip_report_security_event"	10 years ago
Kevin Harwell	43bdddfc26	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	495dfb24b7	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	e1e37e47fd	Merge "res_http_websocket: Forcefully terminate on write errors."	10 years ago
Mark Michelson	58edd2dddc	Merge "chan_dahdi.c: Lock private struct for ast_write()."	10 years ago
Mark Michelson	318b97fd1e	Merge "chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF."	10 years ago
Joshua Colp	1aa722b3ce	Merge "chan_sip: Fix negotiation of iLBC 30."	10 years ago