26945 Commits (89003ea320fdaec02354ac654d398e0220610877)

 

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Author	SHA1	Message	Date
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
Mark Michelson	cf45868984	Merge "res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message."	10 years ago
Joshua Colp	7e65be4ecd	res_http_websocket: Forcefully terminate on write errors. ... The res_http_websocket module will currently attempt to close the WebSocket connection if fatal cases occur, such as when attempting to write out data and being unable to. When the fatal cases occur the code attempts to write a WebSocket close frame out to have the remote side close the connection. If writing this fails then the connection is not terminated. This change forcefully terminates the connection if the WebSocket is to be closed but is unable to send the close frame. ASTERISK-25312 #close Change-Id: I10973086671cc192a76424060d9ec8e688602845	10 years ago
Matt Jordan	a87e2dd254	res/res_format_attr_silk: Expose format attributes to other modules ... This patch adds the .get callback to the format attribute module, such that the Asterisk core or other third party modules can query for the negotiated format attributes. Change-Id: Ia24f55cf9b661d651ce89b4f4b023d921380f19c	10 years ago
Richard Mudgett	87c92d2aee	chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF. ... Pressing DTMF digits on a phone to go out on a DAHDI channel can result in the digit not being recognized or even heard by the peer. Phone -> Asterisk -> DAHDI/channel Turns out the DAHDI behavior with DTMF generation (and any other generated tones) is exposed by the "buffers=" setting in chan_dahdi.conf. When Asterisk requests to start sending DTMF then DAHDI waits until its write buffer is empty before generating any samples for the DTMF tones. When Asterisk subsequently requests DAHDI to stop sending DTMF then DAHDI immediately stops generating the DTMF samples. As a result, the more samples there are in the DAHDI write buffer the shorter the time DTMF actually gets sent on the wire. If there are more samples in the write buffer than the time DTMF is supposed to be sent then no DTMF gets sent on the wire. With the "buffers=12,half" setting and each buffer representing 20 ms of samples then the DAHDI write buffer is going to contain around 120 ms of samples. For DTMF to be recognized by the peer the actual sent DTMF duration needs to be a minimum of 40 ms. Therefore, the intended duration needs to be a minimum of 160 ms for the peer to receive the minimum DTMF digit duration to recognize it. A simple and effective solution to work around the DAHDI behavior is for Asterisk to flush the DAHDI write buffer when sending DTMF so the full duration of DTMF is actually sent on the wire. When someone is going to send DTMF they are not likely to be talking before sending the tones so the flushed write samples are expected to just contain silence. * Made dahdi_digit_begin() flush the DAHDI write buffer after requesting to send a DTMF digit. ASTERISK-25315 #close Reported by John Hardin Change-Id: Ib56262c708cb7858082156bfc70ebd0a220efa6a	10 years ago
Richard Mudgett	b9b957d4e9	chan_dahdi.c: Lock private struct for ast_write(). ... There is a window of opportunity for DTMF to not go out if an audio frame is in the process of being written to DAHDI while another thread starts sending DTMF. The thread sending the audio frame could be past the currently dialing check before being preempted by another thread starting a DTMF generation request. When the thread sending the audio frame resumes it will then cause DAHDI to stop the DTMF tone generation. The result is no DTMF goes out. * Made dahdi_write() lock the private struct before writing to the DAHDI file descriptor. ASTERISK-25315 Reported by John Hardin Change-Id: Ib4e0264cf63305ed5da701188447668e72ec9abb	10 years ago
Richard Mudgett	f3f5b45d57	res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message. ... If the saved SUBSCRIBE message is not parseable for whatever reason then Asterisk could crash when libpjsip tries to parse the message and adds an error message to the parse error list. * Made ast_sip_create_rdata() initialize the parse error rdata list. The list is checked after parsing to see that it remains empty for the function to return successful. ASTERISK-25306 Reported by Mark Michelson Change-Id: Ie0677f69f707503b1a37df18723bd59418085256	10 years ago
Alexander Traud	991d4da1eb	chan_sip: Fix negotiation of iLBC 30. ... iLBC 20 was advertised in a SIP/SDP negotiation. However, only iLBC 30 is supported. Removes "a=fmtp:x mode=y" from SDP. Because of RFC 3952 section 5, only iLBC 30 is negotiated now. ASTERISK-25309 #close Change-Id: I92d724600a183eec3114da0ac607b994b1a793da	10 years ago
Matt Jordan	e188192ad1	main/format: Add an API call for retrieving format attributes ... Some codecs that may be a third party library to Asterisk need to have knowledge of the format attributes that were negotiated. Unfortunately, when the great format migration of Asterisk 13 occurred, that ability was lost. This patch adds an API call, ast_format_attribute_get, to the core format API, along with updates to the unit test to check the new API call. A new callback is also now available for format attribute modules, such that they can provide the format attribute values they manage. Note that the API returns a void *. This is done as the format attribute modules themselves may store format attributes in any particular manner they like. Care should be taken by consumers of the API to check the return value before casting and dereferencing. Consumers will obviously need to have a priori knowledge of the type of the format attribute as well. Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3	10 years ago
Joshua Colp	feffafbd55	Merge "Replace htobe64 with htonll"	10 years ago
Joshua Colp	ff36b5482b	Merge "Replaces clock_gettime() with ast_tsnow()"	10 years ago
Matt Jordan	35fe8f74fe	Merge "rtp_engine.c: Fix performance issue with several channel drivers that use RTP."	10 years ago
David M. Lee	d5f0c27122	Replace htobe64 with htonll ... We don't have a compatability function to fill in a missing htobe64; but we already have one for the identical htonll. Change-Id: Ic0a95db1c5b0041e14e6b127432fb533b97e4cac	10 years ago
David M. Lee	40caf0ad9b	Replaces clock_gettime() with ast_tsnow() ... clock_gettime() is, unfortunately, not portable. But I did like that over our usual `ts.tv_nsec = tv.tv_usec * 1000` copy/paste code we usually do when we want a timespec and all we have is ast_tvnow(). This patch adds ast_tsnow(), which mimics ast_tvnow(), but returns a timespec. If clock_gettime() is available, it will use that. Otherwise ast_tsnow() falls back to using ast_tvnow(). Change-Id: Ibb1ee67ccf4826b9b76d5a5eb62e90b29b6c456e	10 years ago
Scott Emidy	12e6f5ac01	ARI: Retrieve existing log channels ... An http request can be sent to get the existing Asterisk logs. The command "curl -v -u user:pass -X GET 'http://localhost:8088 /ari/asterisk/logging'" can be run in the terminal to access the newly implemented functionality. * Retrieve all existing log channels ASTERISK-25252 Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808	10 years ago
Scott Emidy	b91ca7ba49	ARI: Creating log channels ... An http request can be sent to create a log channel in Asterisk. The command "curl -v -u user:pass -X POST 'http://localhost:088/ari/asterisk/logging/mylog? configuration=notice,warning'" can be run in the terminal to access the newly implemented functionality for ARI. * Ability to create log channels using ARI ASTERISK-25252 Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782	10 years ago
Joshua Colp	ecd4cde521	Merge "ARI: Deleting log channels"	10 years ago
Joshua Colp	1b89cbb3b0	Merge "res_pjsip: Ensure sanitized XML is NULL terminated."	10 years ago
Joshua Colp	12d7c8a740	Merge "res_pjsip_pubsub: More accurately persist packet."	10 years ago