asterisk

Commit Graph

Author	SHA1	Message	Date
Elazar Broad	29694eb2aa	core/logging: Fix logging to more than one syslog channel Currently, Asterisk will log to the last configured syslog channel in logger.conf. This is due to the fact that the final call to openlog() supersedes all of the previous calls. This commit removes the call to openlog() and passes the facility to ast_log_vsyslog(), along with utilizing the LOG_MAKEPRI macro to ensure that the message is routed to the correct facility and with the correct priority. ASTERISK-25407 #close Reported by: Elazar Broad Tested by: Elazar Broad Change-Id: Ie2a2416bc00cce1b04e99ef40917c2011953ddd2	10 years ago
Matt Jordan	764323fcc6	Merge "app_record: RECORDED_FILE variable not being populated" into 11	10 years ago
Joshua Colp	b5c38ff744	Merge "pbx: Update device and presence state when changing a hint extension." into 11	10 years ago
Kevin Harwell	455a31476b	app_record: RECORDED_FILE variable not being populated The RECORDED_FILE variable is empty unless a '%d' is specified in the filename. This patch makes it so the variable is always set to the filename. ASTERISK-25410 #close Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653	10 years ago
Joshua Colp	51013b052d	pbx: Update device and presence state when changing a hint extension. When changing a hint extension without removing the hint first the device state and presence state is not updated. This causes the state of the hint to be that of the previous extension and not the current one. This state is kept until a state change occurs as a result of something (presence state change, device state change). This change updates the hint with the current device and presence state of the new extension when it is changed. Any state callbacks which may have been added before the hint extension is changed are also informed of the new device and presence state if either have changed. ASTERISK-25394 #close Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f	10 years ago
Walter Doekes	2fcf45e9cf	chan_sip: Fix From header truncation for extremely long CALLERID(name). The CALLERID(num) and CALLERID(name) and other info are placed into the `char from[256]` in initreqprep. If the name was too long, the addr-spec and params wouldn't fit. Code is moved around so the addr-spec with params is placed there first, and then fitting in as much of the display-name as possible. ASTERISK-25396 #close Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260	10 years ago
Mark Michelson	3cf0f29310	scheduler: Use queue for allocating sched IDs. It has been observed that on long-running busy systems, a scheduler context can eventually hit INT_MAX for its assigned IDs and end up overflowing into a very low negative number. When this occurs, this can result in odd behaviors, because a negative return is interpreted by callers as being a failure. However, the item actually was successfully scheduled. The result may be that a freed item remains in the scheduler, resulting in a crash at some point in the future. The scheduler can overflow because every time that an item is added to the scheduler, a counter is bumped and that counter's current value is assigned as the new item's ID. This patch introduces a new method for assigning scheduler IDs. Instead of assigning from a counter, a queue of available IDs is maintained. When assigning a new ID, an ID is pulled from the queue. When a scheduler item is released, its ID is pushed back onto the queue. This way, IDs may be reused when they become available, and the growth of ID numbers is directly related to concurrent activity within a scheduler context rather than the uptime of the system. Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2	10 years ago
Matt Jordan	0a92cb361e	Merge "chan_ooh323: Add ProgressIndicator IE with inband info available" into 11	10 years ago
Alexander Anikin	198a1cab8e	chan_ooh323: Add ProgressIndicator IE with inband info available Add ProgressIndicator IE with inband info present to Progress and Alerting Q.931 message ASTERISK-25227 #close Reported by: Alexandr Dranchuk Change-Id: I326ad13cb1db9a72b3fd902bafed3c28a3684203	10 years ago
Matt Jordan	7de5a820ae	Merge "res_rtp_asterisk: Add more ICE debugging" into 11	10 years ago
David M. Lee	819760baec	res_rtp_asterisk: Add more ICE debugging In working through a recent ICE negotiation bug, I found the debug logging in res_rtp_asterisk to be lacking. This patch adds a number of debug and warning statements that were helpful. Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80	10 years ago
Guido Falsi	ffa26a0c2e	Core/General: Add #ifdef needed on FreeBSD. pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED on FreeBSD too. ASTERISK-25310 #close Reported by: Guido Falsi Change-Id: Iae6befac9028b5b9795f86986a4a08a1ae6ab7c4	10 years ago
Alexander Anikin	07b25a2312	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
David M. Lee	94b764c8f3	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
Joshua Colp	59636e82b2	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
Mark Michelson	60fccb7d3c	Merge "app_queue.c: Extract some functions for simpler code." into 11	10 years ago
Scott Griepentrog	b4535b0e59	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
Richard Mudgett	6a364807f4	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. NOTE: This may fix a consistency issue with realtime ringinuse because the ordering of things was backported from v13. It is similar to how set_member_paused() treats realtime for paused. Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306	10 years ago
Richard Mudgett	a56da797d9	app_queue.c: Fix error checking in QUEUE_MEMBER() read. Change-Id: I7294e13d27875851c2f4ef6818adba507509d224	10 years ago
Richard Mudgett	43150cc58d	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. NOTE: This may fix a consistency issue with the realtime paused setting since how the value is set is controlled by set_member_paused() which treats realtime a little better. * 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
Mark Michelson	6446061614	Merge "chan_sip.c: wrong peer searched in sip_report_security_event" into 11	10 years ago
Kevin Harwell	430db4333e	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	84e16bcafd	Merge "res_http_websocket: Forcefully terminate on write errors." into 11	10 years ago
Richard Mudgett	c777c9565d	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	f43ea74e9e	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
Joshua Colp	b9bd3c1435	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
David M. Lee	06b464ab1b	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
Joshua Colp	c7a1dca4ba	res_rtp_asterisk: Don't leak temporary key when enabling PFS. A change recently went in which enabled perfect forward secrecy for DTLS in res_rtp_asterisk. This was accomplished two different ways depending on the availability of a feature in OpenSSL. The fallback method created a temporary instance of a key but did not free it. This change fixes that. ASTERISK-25265 Change-Id: Iadc031b67a91410bbefb17ffb4218d615d051396	10 years ago
Mark Duncan	2d2e741905	res/res_rtp_asterisk: Add ECDH support This will add ECDH support to Asterisk. It will detect auto ECDH support in OpenSSL (1.0.2b and above) during ./configure. If this is available, it will use it, otherwise it will fall back to prime256v1 (this behavior is consistent with other projects such as Apache and nginx). This fixes WebRTC being broken in Firefox 38+ due to Firefox now only supporting ciphers with perfect forward secrecy. ASTERISK-25265 #close Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b	10 years ago
Richard Mudgett	5311d18101	chan_sip.c: Move NULL check to where it will do some good. v11 only fix. Change-Id: I340512f86cfd3a6f7703971fa8acfffc7d47132b	10 years ago
Richard Mudgett	75185c5d8f	rtp_engine.c: Fix off nominal ref leak and some minor tweaks. v11 only fix. Change-Id: I97885946ebc7eda19f1c18d08698117cf6a7f14f	10 years ago
Richard Mudgett	1b51b5efb6	rtp_engine.c: Tweak glue->update_peer() parameter nil value. Change glue->update_peer() parameter from 0 to NULL to better indicate it is a pointer. Change-Id: Iefbf9d4a708f2b64b7ad2b4e6c33bfaa12ccfa9d	10 years ago
Richard Mudgett	f5cd1fa0df	chan_sip.c: Tweak glue->update_peer() parameter nil value. Change glue->update_peer() parameter from 0 to NULL to better indicate it is a pointer. Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd	10 years ago
Mark Michelson	f2089dce3d	res_http_websocket: Properly encode 64 bit payload A test agent was continuously failing all ARI tests when run against Asterisk 13. As it turns out, the reason for this is that on those test runs, for some reason we decided to use the super extended 64 bit payload length for websocket text frames instead of the extended 16 bit payload length. For 64-bit payloads, the expected byte order over the network is 7, 6, 5, 4, 3, 2, 1, 0 However, we were sending the payload as 3, 2, 1, 0, 7, 6, 5, 4 This meant that we were saying to expect an absolutely MASSIVE payload to arrive. Since we did not follow through on this expected payload size, the client would sit patiently waiting for the rest of the payload to arrive until the test would time out. With this change, we use the htobe64() function instead of htonl() so that a 64-bit byte-swap is performed instead of a 32 bit byte-swap. Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a	10 years ago
Mark Michelson	7e8916908d	Local channels: Alternate solution to ringback problem. Commit `54b25c80c8` solved an issue where a specific scenario involving local channels and a native local RTP bridge could result in ringback still being heard on a calling channel even after the call is bridged. That commit caused many tests in the testsuite to fail with alarming consequences, such as not sending DialBegin and DialEnd events, and giving incorrect hangup causes during calls. This commit reverts the previous commit and implements and alternate solution. This new solution involves only passing AST_CONTROL_RINGING frames across local channels if the local channel is in AST_STATE_RING. Otherwise, the frame does not traverse the local channels. By doing this, we can ensure that a playtones generator does not get started on the calling channel but rather is started on the local channel on which the ringing frame was initially indicated. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39	10 years ago
Mark Michelson	534ed6744a	Merge "audiohook: Read the correct number of samples based on audiohook format." into 11	10 years ago
Mark Michelson	525bbf7d4e	Local channels: Do not block control -1 payloads. Control frames with a -1 payload are used as a special signal to stop playtones generators on channels. This indication is sent both by app_dial as well as by ast_answer() when a call is answered in case any tones were being generated on a calling channel. This control frame type was made to stop traversing local channel pairs as an optimization, because it was thought that it was unnecessary to send these indications, and allowing such unnecessary control frames to traverse the local channels would cause the local channels to optimize away less quickly. As it turns out, through some special magic dialplan code, it is possible to have a tones being played on a non-local channel, and it is important for the local channel to convey that the tones should be stopped. The result of having tones continue to be played on the non-local channel is that the tones play even once the channel has been bridged. By not blocking the -1 control frame type, we can ensure that this situation does not happen. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I40227e4249d6d61dc6a9b08bbe9ee3aa18be7e30	10 years ago
Joshua Colp	5606da754a	audiohook: Read the correct number of samples based on audiohook format. Due to changes in audiohooks to support different sample rates the underlying storage of samples is in the format of the audiohook itself and not of the format being requested. This means that if a channel is using G722 the samples stored will be at 16kHz. If something subsequently reads from the audiohook at a format which is not the same sample rate as the audiohook the number of samples needs to be adjusted. Given the following example: 1. Channel writing into audiohook at 16kHz (as it is using G722). 2. Chanspy reading from audiohook at 8kHz. The original code would read 160 samples from the audiohook for each 20ms of audio. This is incorrect. Since the audio in the audiohook is at 16kHz the actual number needing to be read is 320. Failure to read this much would cause the audiohook to reset itself constantly as the buffer became full. This change adjusts the requested number of samples by determining the duration of audio requested and then calculating how many samples that would be in the audiohook format. ASTERISK-25247 #close Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d	10 years ago
Rusty Newton	43a52df20d	Documentation: chan_sip doesn't support WS or WSS in outbound register. * In sip.conf.sample fix sentence where we said that WS or WSS are supported transports for use in an outbound register definition. They are not supported in that case. ASTERISK-24853 #close Reported by: PSDK Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca	10 years ago
Michael Cargile	bd6fd55ea4	res/res_musiconhold: Add a warning when MOH does not exist Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b	10 years ago
Patric Marschall	0433f08903	sig_pri.h: force_restart_unavailable_chans in wrong scope In channels/sig_pri.h, struct sig_pri_span, the field force_restart_unavailable_chans is only defined if #if defined(HAVE_PRI_MCID) is true. All other occurences of force_restart_unavailable_chans are outside of the #if defined(HAVE_PRI_MCID) endif scope. ASTERISK-25257 #close Reported by: Patric Marschall Change-Id: I071de89cc2cd0d85927a013036e235851f672549	10 years ago
Matt Jordan	b36d670dc1	apps/app_dictate: Fix typo in attribution Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian (GameGamer43) for pointing that out. Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106	10 years ago
Matt Jordan	883f6f5d21	tests/test_devicestate: Add additional tests for the device state API This patch adds more tests that exercise the device state API. This includes: * Tests that cover adding a device state provider, as well as deleting a device state provider. This also verifies that you cannot add an already added device state provider, and cannot delete an already deleted device state provider. * A test that covers changing device state and receiving said updates from a device state subscriber. This also covers hitting both the device state cache as well as a custom device state provider. * A test that covers converting device state to channel state and device state values to a string representation and back. * A test that covers obtaining device state from an active channel and a channel driver that provides its own device state. Change-Id: I2adca67ffb405cd8625a5d6df1e3f9b3d945c08d	10 years ago
Matt Jordan	e545c05e35	main/devicestate: Prevent duplicate registration of device state providers Currently, the device state provider API will allow you to register a device state provider with the same case insensitive name more than once. This could cause strange issues, as the duplicate device state providers will not be queried when a device's state has to be polled. This patch updates the API such that a device state provider with the same name as one that has already registered will be rejected. Change-Id: I4a418a12280b7b6e4960bd44f302e27cd036ceb2	10 years ago
Joshua Colp	47ebab959e	res_rtp_asterisk: Ensure DTLS timeout timer is -1 if DTLS is not used. This change fixes a bug where the DTLS timeout timer would be initialized to 0 if DTLS was not used for an RTP session. ASTERISK-25103 Change-Id: If8d26bb054f1d300838850da5b8db9044c2fe2ac	10 years ago
Joshua Colp	1ad827327a	res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context. This change moves logic for setting up the DTLS SSL contexts to when the SDP is done being processed instead of when ICE negotiation completes. It also stops handshakes from being initiated when we are acting as a server. Manipulating the SSL context when ICE negotiation has completed is problematic as the SSL context is not protected and if acting as a client the remote side may have started DTLS negotiation already. The retransmission timeout timer code has also been split up and simplified some. Both RTP and RTCP now have their own timers and the points at which the timer is stopped and started is now more specific. When a packet is sent the timer is started. When a response is received but before it is processed the timer is stopped. This provides a guarantee that the timeout is not occurring while the response is processed. ASTERISK-22805 #close ASTERISK-24550 #close ASTERISK-24651 #close ASTERISK-24832 #close ASTERISK-25103 #close ASTERISK-25127 #close Change-Id: Ib75ea2546f29d6efc3d2d37c58df6986c7bd9b91	10 years ago
Joshua Colp	895dbe532c	Merge "chan_sip: Fix early call pickup caused deadlock." into 11	10 years ago
Joshua Colp	6d9115767f	Merge "chan_vpb.cc: Fix compiler warning Jenkins found." into 11	10 years ago
Joshua Colp	89b0aaa652	Merge "res_timing: Don't close FD 0 when out of open files." into 11	10 years ago
Joshua Colp	077bfe7455	Merge "rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format." into 11	10 years ago

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