29658 Commits (14)

 

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Author	SHA1	Message	Date
Jenkins2	040487b90e	Merge "res_srtp: lower log level of auth failures" into 14	8 years ago
Rodrigo Ramírez Norambuena	339676b77d	res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1 ... In PostgreSQL 9.1 the backslash are string literals and not the escape of characters. In previous issue ASTERISK_26057 was fixed the use of escape LIKE but the support for old version of Postgresql than 9.1 was dropped. The sentence before make was "ESCAPE '\'" but in version before than 9.1 need it to be as follow "ESCAPE '\\'". ASTERISK-27283 Change-Id: I96d9ee1ed7693ab17503cb36a9cd72847165f949	8 years ago
Joshua Colp	8ee7ed020f	Merge "chan_sip: Expose read-only access to the full SIP INVITE Request-URI" into 14	8 years ago
Jenkins2	e5f911519d	Merge "bridge : Fix one-way direct-media when early bridging with native_rtp" into 14	8 years ago
Jean Aunis	d884871ca7	bridge : Fix one-way direct-media when early bridging with native_rtp ... When two channels were early bridged in a native_rtp bridge, the RTP description on one side was not updated when the other side answered. This patch forbids non-answered channels to enter a native_rtp bridge, and triggers a bridge reconfiguration when an ANSWER frame is received. ASTERISK-27257 Change-Id: If1aaee1b4ed9658a1aa91ab715ee0a6413b878df	8 years ago
George Joseph	c97eb95193	res_pjsip_pubsub: Check for Content-Type header in rx_notify_request ... pubsub_on_rx_notify_request wasn't checking for a null Content-Type header before checking that it was application/simple-message-summary. ASTERISK-27279 Reported by: Ross Beer Change-Id: Iec2a6c4d2e74af37ff779ecc9fd35644c5c4ea52	8 years ago
David J. Pryke	060cea2cca	chan_sip: Expose read-only access to the full SIP INVITE Request-URI ... Provide a way to get the contents of the the Request URI from the initial SIP INVITE in dial plan function call. (In this case "${CHANNEL(ruri)}") ASTERISK-27278 Reported by: David J. Pryke Tested by: David J. Pryke Change-Id: I1dd4d6988eed1b6c98a9701e0e833a15ef0dac3e	8 years ago
Joshua Colp	3274c13b12	Merge "AST-2017-008: Improve RTP and RTCP packet processing." into 14	8 years ago
Joshua Colp	864e76374c	Merge "res_calendar: On reload, update all configuration" into 14	8 years ago
Joshua Colp	05bb57c56f	Merge "cdr_mysql.c: Apply cdrzone to start and answer" into 14	8 years ago
Alexander Traud	393d813789	tcptls: Fixed a white space error. ... ASTERISK-26606 Change-Id: I81a7268ef7ba012d4d80d44c70b6276d48e397fa	8 years ago
Alexander Traud	73da3df7b6	res_srtp: lower log level of auth failures ... Previously, sRTP authentication failures were reported on log level WARNING. When such failures happen, each RT(C)P packet is affected, spamming the log. Now, those failures are reported at log level VERBOSE 2. Furthermore, the amount is further reduced (previously all two seconds, now all three seconds). Additionally, the new log entry informs whether media (RTP) or statistics (RTCP) are affected. ASTERISK-16898 #close Change-Id: I6c98d46b711f56e08655abeb01c951ab8e8d7fa0	8 years ago
Richard Mudgett	4a57e20d52	AST-2017-008: Improve RTP and RTCP packet processing. ... Validate RTCP packets before processing them. * Validate that the received packet is of a minimum length and apply the RFC3550 RTCP packet validation checks. * Fixed potentially reading garbage beyond the received RTCP record data. * Fixed rtp->themssrc only being set once when the remote could change the SSRC. We would effectively stop handling the RTCP statistic records. * Fixed rtp->themssrc to not treat a zero value as special by adding rtp->themssrc_valid to indicate if rtp->themssrc is available. ASTERISK-27274 Make strict RTP learning more flexible. Direct media can cause strict RTP to attempt to learn a remote address again before it has had a chance to learn the remote address the first time. Because of the rapid relearn requests, strict RTP could latch onto the first remote address and fail to latch onto the direct media remote address. As a result, you have one way audio until the call is placed on and off hold. The new algorithm learns remote addresses for a set time (1.5 seconds) before locking the remote address. In addition, we must see a configured number of remote packets from the same address in a row before switching. * Fixed strict RTP learning from always accepting the first new address packet as the new stream. * Fixed strict RTP to initialize the expected sequence number with the last received sequence number instead of the last transmitted sequence number. * Fixed the predicted next sequence number calculation in rtp_learning_rtp_seq_update() to handle overflow. ASTERISK-27252 Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c	8 years ago
Jenkins2	a9160793a5	Merge "res_pjsip: Filter out non SIP(S) requests" into 14	8 years ago
Sean Bright	edd5b30387	res_calendar: On reload, update all configuration ... This changes the behavior of res_calendar to drop all existing calendars and re-create them whenever a reload is done. The Calendar API provides no way for configuration information to be pushed down to calendar 'techs' so updated settings would not take affect until a module unload/load was done or Asterisk was restarted. Asterisk 15+ already has a configuration option 'fetch_again_at_reload' that performs a similar function. Also fix a tiny memory leak in res_calendar_caldav while we're at it. ASTERISK-25524 #close Reported by: Jesper Change-Id: Ib0f8057642e9d471960f1a79fd42e5a3ce587d3b	8 years ago
Jenkins2	cdadbd3d22	Merge "res_calendar: Various fixes" into 14	8 years ago
George Joseph	cf8d1d8cc7	res_pjsip: Filter out non SIP(S) requests ... Incoming requests with non sip(s) URIs in the Request, To, From or Contact URIs are now rejected with PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in pjsip_message_filter (formerly pjsip_message_ip_updater) and is done at pjproject's "TRANSPORT" layer before a request can even reach the distributor. URIs read by res_pjsip_outbound_publish from pjsip.conf are now also checked for both length and sip(s) scheme. Those URIs read by outbound registration and aor were already being checked for scheme but their error messages needed to be updated to include scheme failure as well as length failure. Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460	8 years ago
Jenkins2	aa90b169df	Merge "chan_rtp: Use μ-law by default instead of signed linear" into 14	8 years ago
Sean Bright	263578f3f7	res_calendar: Various fixes ... * The way that we were looking at XML elements for CalDAV was extremely fragile, so use SAX2 for increased robustness. * Don't complain about a 'channel' not be specified if autoreminder is not set. Assume that if 'channel' is not set, we don't want to be notified. * Fix some truncated CLI output in 'calendar show calendar' and make the 'Autoreminder' description a bit more clear ASTERISK-24588 #close Reported by: Stefan Gofferje ASTERISK-25523 #close Reported by: Jesper Change-Id: I200d11afca6a47e7d97888f286977e2e69874b2c	8 years ago
alex	3e445c20d8	cdr_mysql.c: Apply cdrzone to start and answer ... Change-Id: I7de0a5adc89824a5f2b696fc22c80fc22dff36b0	8 years ago
Sean Bright	8ab2dcb4b0	chan_rtp: Use μ-law by default instead of signed linear ... Multicast/Unicast RTP do not use SDP so we need to use a format that cleanly maps to one of the static RTP payload types. Without this change, an Originate to a Multicast or Unicast channel without a format specified would produce no audio on the receiving device. ASTERISK-21399 #close Reported by: Tzafrir Cohen Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3	8 years ago
George Joseph	35cc916e63	res_pjsip: Add handling for incoming unsolicited MWI NOTIFY ... A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive unsolicited MWI NOTIFY requests and make them available to other modules via the stasis message bus. res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" that parses a simple-message-summary body and, if endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state with the voice-message counts from the message. Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c	8 years ago
Joshua Colp	cadc96c04b	Merge "alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode" into 14	8 years ago
Richard Mudgett	1232a40889	res_rtp_asterisk.c: Add doxygen to RTCP payload types. ... Change-Id: I3f20ce428777cc4ce9c13b2f808d29ff8c873998	8 years ago
Jenkins2	cd87533882	Merge "alembic: Add support for MS-SQL" into 14	8 years ago
George Joseph	5aa60aea6f	alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode ... The downgrade function was missing "_v2" at the end of the alter column type. Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e	8 years ago
Walter Doekes	b4faf94ea1	res/res_pjsip: Fix localnet checks in pjsip, part 2. ... In 45744fc53, I mistakenly broke SDP media address rewriting by misinterpreting which address was checked in the localnet comparison. Instead of checking the remote peer address to decide whether we need media address rewriting, we check our local media address: if it's local, then we rewrite. This feels awkward, but works and even made directmedia work properly if you set local_net. (For the record: for local peers, the SDP media rewrite code is not called, so the comparison does no harm there.) ASTERISK-27248 #close Change-Id: I566be1c33f4d0a689567d451ed46bab9c3861d4f	8 years ago
Florian Floimair	1badedfe08	alembic: Add support for MS-SQL ... MS-SQL has no native Enum-type support and therefore needs to work with constraints. Since these constraints need unique names the suggested approach referenced in the following alembic documentation has been applied: http://bit.ly/2x9r8pb ASTERISK-27255 #close Change-Id: I8b579750dae0c549f1103ee50172644afb9b2f95	8 years ago
Jenkins2	9042b8cecd	Merge "app_waitforsilence: Cleanup & don't treat missing frames as 'noise'" into 14	8 years ago
Jenkins2	2522e203ca	Merge "chan_sip: when getting sip pvt return failure if not found" into 14	8 years ago
Jenkins2	35c8c389d3	Merge "res_srtp: Add support for libsrtp2.1." into 14	8 years ago
Jenkins2	1405365f81	Merge "chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE" into 14	8 years ago
Joshua Colp	2d86620027	Merge "func_cdr: honour 'u' flag on dummy channel" into 14	8 years ago
Jenkins2	6829d11a25	Merge "stasis/control.c: Fix set_interval_hook() ref leak." into 14	8 years ago
Scott Griepentrog	74e4f7ecf1	chan_sip: when getting sip pvt return failure if not found ... In handle_request_invite, when processing a pickup, a call is made to get_sip_pvt_from_replaces to locate the pvt for the subscription. The pvt is assumed to be valid when zero is returned indicating no error, and is dereferenced which can cause a crash if it was not found. This change checks the not found case and returns -1 which allows the calling code to fail appropriately. ASTERISK-27217 #close Reported-by: Bryan Walters Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612	8 years ago
Sean Bright	33533109e4	app_waitforsilence: Cleanup & don't treat missing frames as 'noise' ... * WaitForSilence completes successfully if it receives no media in the specified timeout, but when acting as WaitForNoise that logic needs to be reversed. * Use standard argument parsing macros and add some error checking for invalid values. * The documentation indicated that the first argument to both WaitForSilence and WaitForNoise was required when it was not. Update the documentation to reflect that. * Wrap up some behavior in structs to avoid boolean checks all over the place. ASTERISK-24066 #close Reported by: M vd S Change-Id: I01d40adc5b63342bb5018a1bea2081a0aa191ef9	8 years ago
Richard Mudgett	4bed94b044	stasis/control.c: Fix set_interval_hook() ref leak. ... Change-Id: Ia0edb7dc0dbbb879c079ff7000f1b722d86ce7dc	8 years ago
George Joseph	983f3e1f0f	stasis/control: Fix possible deadlock with swap channel ... If an error occurs during a bridge impart it's possible that the "bridge_after" callback might try to run before control_swap_channel_in_bridge has been signalled to continue. Since control_swap_channel_in_bridge is holding the control lock and the callback needs it, a deadlock will occur. * control_swap_channel_in_bridge now only holds the control lock while it's actually modifying the control structure and releases it while the bridge impart is running. * bridge_after_cb is now tolerant of impart failures. Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3	8 years ago
Jenkins2	47e4cbc395	Merge "alembic: Fix enum creation for dtls_fingerprint" into 14	8 years ago
Jenkins2	458067f336	Merge "alembic: fix erroneous commit for add_prune_on_boot" into 14	8 years ago
Jenkins2	d742de0309	Merge "res/res_pjsip: Standardize/fix localnet checks across pjsip." into 14	8 years ago
Vitezslav Novy	89e7bc1883	chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE ... If directmedia=yes is configured, when call is answered, Asterisk sends reINVITE to both parties to set up media path directly between the endpoints. In this reINVITE msg SDP origin line (o=) contains IP address of endpoint instead of IP of asterisk. This behavior violates RFC3264, sec 8: "When issuing an offer that modifies the session, the "o=" line of the new SDP MUST be identical to that in the previous SDP, except that the version in the origin field MUST increment by one from the previous SDP." This patch assures IP address of Asterisk is always sent in SDP origin line. ASTERISK-17540 Reported by: saghul Change-Id: I533a047490c43dcff32eeca8378b2ba02345b64e	8 years ago
Jenkins2	63102d9632	Merge "res_rtp_asterisk.c: Check RTP packet version earlier." into 14	8 years ago
Joshua Colp	c5bc2558d7	Merge "formats: Restore previous fread() behavior" into 14	8 years ago
George Joseph	a7a56c53d5	alembic: Fix enum creation for dtls_fingerprint ... Change-Id: Ic061c5066a146616a68376881c7e4cf6d6e7e7db	8 years ago
Jenkins2	dd201746fa	Merge "res_pjsip_t38: Make t38_reinvite_response_cb tolerant of NULL channel" into 14	8 years ago
Florian Floimair	2c1b03bcb3	alembic: fix erroneous commit for add_prune_on_boot ... Added include for postgresql ENUM type and redefined values in the same way as in the other migration scripts. ASTERISK-27254 #close Change-Id: Id667304cdf3891b1c2f7d35fab3e2a84026159fa	8 years ago
Alexander Traud	a491fb67c1	res_srtp: Add support for libsrtp2.1. ... Asterisk is able to use libSRTP 2.0.x. However since libSRTP 2.1.x, the macro SRTP_AES_ICM got renamed to SRTP_AES_ICM_128. Beside to still compile with previous versions of libSRTP, this change allows libSRTP 2.1.x as well. ASTERISK-27253 #close Change-Id: I2e6eb3c3bc844fee8a624060a2eb6f182dc70315	8 years ago
Ben Ford	3dc7781e29	chan_pjsip: Suppress frame warnings. ... When rtp_keepalive is on for a PJSIP endpoint dialing to another Asterisk instance also using PJSIP, Asterisk will continue to print warning messages about not being able to send frames of a certain type. This suppresses that warning message. Change-Id: I0332a05519d7bda9cacfa26d433909ff1909be67	8 years ago
Richard Mudgett	6d8c40659f	res_rtp_asterisk.c: Check RTP packet version earlier. ... Change-Id: Ic6493a7d79683f3e5845dff1cee49445fd5a0adf	8 years ago