29332 Commits (178d2bc7ef17c506b9fb9c119e494642adc4b38c)

 

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Author	SHA1	Message	Date
Jenkins2	a0e3da1f71	Merge "logger: Added logger_queue_limit to the configuration options." into 14	9 years ago
Joshua Colp	d6a9beaab1	Merge "Prevent Undefined Capath Crash" into 14	9 years ago
Alexei Gradinari	d02020791b	res_pjsip: New endpoint option "refer_blind_progress" ... This option was added to turn off notifying the progress details on Blind Transfer. If this option is not set then the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted". Some SIP phones like Mitel/Aastra or Snom keep the line busy until receive "200 OK". ASTERISK-26333 #close Change-Id: Id606fbff2e02e967c02138457badc399144720f2	9 years ago
Joshua Colp	ecd371846e	tcptls: Improve error messages for TLS connections. ... This change uses the functions provided by OpenSSL to query and better construct error messages for situations where the connection encounters a problem. ASTERISK-26606 Change-Id: I7ae40ce88c0dc4e185c4df1ceb3a6ccc198f075b	9 years ago
Joshua Elson	ee019a5f77	Prevent Undefined Capath Crash ... It is possible to initialize a valid config without a capath or cafile definition. This will cause a crash on a reload. This fix ensures capath is always allocated. ASTERISK-26983 #close Change-Id: I63ff715d9d9023427543a5b8a4ba7b0d82533c12	9 years ago
George Joseph	247cb7c9f2	cel_odbc: Fix timestamp processing for microseconds ... When a column is of type timestamp, the fraction part of the event field's seconds was frequently parsed incorrectly especially if there were leading zeros. For instance "2017-05-23 23:55:03.023" would be parsed into an int as "23" then when the timestamp was formatted again to be inserted into the database column it'd be "2017-05-23 23:55:03.23" which is now 230 milliseconds instead of 23 milliseconds. "03.000001" would be transformed to "03.1", etc. * If the event field is 'eventtime' and the db column is timestamp, then existing processing has already correctly formatted the timestamp so now we simply use it rather than parsing it and re-printing it. This is the most common use case anyway. * If the event field is other than 'eventtime' and the db column is timestamp, we now parse the seconds, including the fractional part into a double rather than 2 ints. This preserves the magnitude and precision of the fractional part. When we print it, we now print it as a "%09.6lf" which correctly represents the input. To be honest, why we parse the string timestamp into components, test the components, then print the components back into a string timestamp is beyond me. We should use parse it, test it, then if it passes, use the original string representation in the database call. Maybe someone thought that some implementations wouldn't take a partial timestamp string like "2017-05-06" and decided to always produce a full timestamp string even if an abbreviated one was supplied. Anyway, I'm leaving it as it is. ASTERISK-25032 #close Reported-by: Etienne Lessard Change-Id: Id407e6221f79a5c1120e1a70bc7e893bbcaf1938	9 years ago
Joshua Colp	097f90220a	res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages. ... This change adds the required logic to allow the SIP Call-ID to be placed into the HEP RTCP traffic if the chan_sip module is used. In cases where the option is enabled but the channel is not either SIP or PJSIP then the code will fallback to the channel name as done previously. Based on the change on Nir's branch at: team/nirs/hep-chan-sip-support ASTERISK-26427 Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d	9 years ago
Joshua Colp	1e0213616c	Merge "func_cdr: Allow empty value for CDR dialplan function." into 14	9 years ago
George Joseph	944f9435bc	logger: Added logger_queue_limit to the configuration options. ... All log messages go to a queue serviced by a single thread which does all the IO. This setting controls how big that queue can get (and therefore how much memory is allocated) before new messages are discarded. The default is 1000. Should something go bezerk and log tons of messages in a tight loop, this will prevent memory escalation. When the limit is reached, a WARNING is logged to that effect and messages are discarded until the queue is empty again. At that time another WARNING will be logged with the count of discarded messages. There's no "low water mark" for this queue because the logger thread empties the entire queue and processes it in 1 batch before going back and waiting on the queue again. Implementing a low water mark would mean additional locking as the thread processes each message and it's not worth it. A "test" was added to test_logger.c but since the outcome is non-deterministic, it's really just a cli command, not a unit test. Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1	9 years ago
Joshua Colp	6b9ed2574b	Merge "netsock2.c: Made get/set addr port avoid potential uninitialized memory." into 14	9 years ago
Joshua Colp	eb38be715a	Merge "bridge: Fix returning to dialplan when executing Bridge() from AMI." into 14	9 years ago
Richard Mudgett	fe06758718	netsock2.c: Made get/set addr port avoid potential uninitialized memory. ... Change-Id: I532052bd7cd95a4b3565485fc01e2a1ea07ee647	9 years ago
Joshua Colp	fbfce63b30	func_cdr: Allow empty value for CDR dialplan function. ... A regression was introduced in 12 where passing an empty value to the CDR dialplan function was not longer allowed. This change returns to the behavior of 11 where it is permitted. ASTERISK-26173 Change-Id: I3f148203b54ec088007e29e30005a5de122e51c5	9 years ago
George Joseph	e9d563c4db	app_confbridge: Fix reference to cfg in menu_template_handler ... menu_template_handler wasn't properly accounting for the fact that it might be called both during a load/reload (which isn't really valid but not prevented) and by a dialplan function. In both cases it was attempting to use the "pending" config which wasn't valid in the latter case. aco_process_config is also partly to blame because it wasn't properly cleaning "pending" up when a reload was done and no changes were made. Both of these contributed to a crash if CONFBRIDGE(menu,template) was called in a dialplan after a reload. * aco_process_config now sets info->internal->pending to NULL after it unrefs it although this isn't strictly necessary in the context of this fix. * menu_template_handler now uses the "current" config and silently ignores any attempt to be called as a result of someone uses the "template" parameter in the conf file. Luckily there's no other place in the codebase where aco_pending_config is used outside of aco_process_config. ASTERISK-25506 #close Reported-by: Frederic LE FOLL Change-Id: Ib349a17d3d088f092480b19addd7122fcaac21a7	9 years ago
Joshua Colp	8cbdd0a5b8	bridge: Fix returning to dialplan when executing Bridge() from AMI. ... When using the Bridge AMI action on the same channel multiple times it was possible for the channel to return to the wrong location in the dialplan if the other party hung up. This happened because the priority of the channel was not preserved across each action invocation and it would fail to move on to the next priority in other cases. This change makes it so that the priority of a channel is preserved when taking control of it from another thread and it is incremented as appropriate such that the priority reflects where the channel should next be executed in the dialplan, not where it may or may not currently be. The Bridge AMI action was also changed to ensure that it too starts the channels at the next location in the dialplan. ASTERISK-24529 Change-Id: I52406669cf64208aef7252a65b63ade31fbf7a5a	9 years ago
Kevin Harwell	3a5683d66c	res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures ... When a call gets put on hold RTP is temporarily stopped and Asterisk was setting the remote RTCP address to NULL. Then when RTCP data was received from the remote endpoint, Asterisk would be missing this information when publishing the rtcp_message stasis event. Consequently, message subscribers (in this case res_hep_rtcp) trying to parse the "from" field output the following error: "ast_sockaddr_split_hostport: Port missing in (null)" This patch makes it so the remote RTCP address is no longer set to NULL when stopping RTP. There was only one place that appeared to check if the remote RTCP address was NULL as a way to tell if RTCP was running. This patch added an additional check on the RTCP schedid for that case to make sure RTCP was truly not running. ASTERISK-26860 #close Change-Id: I6be200fb20db647e48b5138ea4b81dfa7962974b	9 years ago
Jenkins2	24e5ae7e0a	Merge "channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections" into 14	9 years ago
Sean Bright	7f5ce428ed	cleanup: Change severity of fread short-read warning ... Many sound files don't have a full frame's worth of data at EOF, so the warning messages were a bit too noisy. So we demote them to debug messages. Change-Id: I6b617467d687658adca39170a81797a11cc766f6	9 years ago
Jenkins2	956f462431	Merge "res_pjsip_t38.c: Fix deadlock in T.38 framehook." into 14	9 years ago
Thierry Magnien	5cee143d9c	channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections ... For outgoing TCP connections, Asterisk uses the first IP address of the interface instead of the IP address we asked him to bind to. ASTERISK-26922 #close Reported-by: Ksenia Change-Id: I43c71ca89211dbf1838e5bcdb9be8d06d98e54eb	9 years ago
Richard Mudgett	d54ae43cc8	res_pjsip_t38.c: Fix deadlock in T.38 framehook. ... A deadlock can happen between a channel lock and a pjsip session media container lock. One thread is processing a reINVITE's SDP and walking through the session's media container when it waits for the channel lock to put the determined format capabilities onto the channel. The other thread is writing a frame to the channel and processing the T.38 frame hook. The T.38 frame hook then waits for the pjsip session's media container lock. The two threads are now deadlocked. * Made the T.38 frame hook release the channel lock before searching the session's media container. This fix has been done to several other frame hooks to fix deadlocks. ASTERISK-26974 #close Change-Id: Ie984a76ce00bef6ec9aa239010e51e8dd74c8186	9 years ago
George Joseph	d9549bc475	res_pjsip_outbound_authenticator_digest: Add context to log messages ... There was no context info in this module's log messages so it was impossible to toubleshoot. Added endpoint or host to all messages and added the realms in the challenge for the "No auth credentials for any realm" message. Change-Id: Ifeed2786f35fbea7d141237ae15625e472acff9b	9 years ago
Jenkins2	a5e79f71ee	Merge "frame: Better handle interpolated frames." into 14	9 years ago
Jenkins2	d2a135f121	Merge "res_pjsip_session: Add cleanup to ast_sip_session_terminate" into 14	9 years ago
Jenkins2	41c0ff26e1	Merge "res_pjsip/res_pjsip_callerid: NULL check on caller id name string" into 14	9 years ago
Jenkins2	416319852d	Merge "chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK" into 14	9 years ago
Jenkins2	3d82216b32	Merge "vector: defaults and indexes" into 14	9 years ago
Joshua Colp	efe84aede9	Merge "cleanup: Fix fread() and fwrite() error handling" into 14	9 years ago
Jenkins2	35c15c589d	Merge "pjproject_bundled: Add --disable-libwebrtc to configure" into 14	9 years ago
George Joseph	cd4824ca69	res_pjsip_session: Add cleanup to ast_sip_session_terminate ... If you use ast_request to create a PJSIP channel but then hang it up without causing a transaction to be sent, the session will never be destroyed. This is due ot the fact that it's pjproject that triggers the session cleanup when the transaction ends. app_chanisavail was doing this to get more granular channel state and it's also possible for this to happen via ARI. * ast_sip_session_terminate was modified to explicitly call the cleanup tasks and unreference session if the invite state is NULL AND invite_tsx is NULL (meaning we never sent a transaction). * chan_pjsip/hangup was modified to bump session before it calls ast_sip_session_terminate to insure that session stays valid while it does its own cleanup. * Added test events to session_destructor for a future testsuite test. ASTERISK-26908 #close Reported-by: Richard Mudgett Change-Id: I52daf6f757184e5544c261f64f6fe9602c4680a9	9 years ago
Jenkins2	31ede0f9d3	Merge "res_rtp_asterisk.c: Fix crash in RTCP DTLS operation." into 14	9 years ago
Kevin Harwell	6722ec6a6a	res_pjsip/res_pjsip_callerid: NULL check on caller id name string ... It's possible for a name in a party id structure to be marked as valid, but the name string itself be NULL (for instance this is possible to do by using the dialplan CALLERID function). There were a couple of places where the name was validated, but the string itself was not checked before passing it to functions like 'strlen'. This of course caused a crashed. This patch adds in a NULL check before attempting to pass it into a function that is not NULL tolerant. ASTERISK-25823 #close Change-Id: Iaa6ffe9d92f598fe9e3c8ae373fadbe3dfbf1d4a	9 years ago
Kevin Harwell	00001443dc	vector: defaults and indexes ... Added an pre-defined integer vector declaration. This makes integer vectors easier to declare and pass around. Also, added the ability to default a vector up to a given size with a default value. Lastly, added functionality that returns the "nth" index of a matching value. Also, updated a unit test to test these changes. Change-Id: Iaf4b51b2540eda57cb43f67aa59cf1d96cdbcaa5	9 years ago
Joshua Colp	4cdf937a2e	frame: Better handle interpolated frames. ... Interpolated frames are frames which contain a number of samples but have no actual data. Audiohooks did not handle this case when translating an incoming frame into signed linear. It assumed that a frame would always contain media when it may not. If this occurs audiohooks will now immediately return and not act on the frame. As well for users of ast_trans_frameout the function has been changed to be a bit more sane and ensure that the data pointer on a frame is set to NULL if no data is actually on the frame. This allows the various spots in Asterisk that check for an interpolated frame based on the presence of a data pointer to work as expected. ASTERISK-26926 Change-Id: I7fa22f631fa28d540722ed789ce28e84c7f8662b	9 years ago
Jean Aunis	87a24362a2	chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK ... Some equipments may send a re-INVITE containing an SDP in the final ACK request. If this happens in the context of direct media, the remote end should be updated with a re-INVITE. This patch queues an "update RTP peer" frame to trigger the re-INVITE, instead of the "source change" frame wich was used previously. ASTERISK-26951 Change-Id: I3644d2025f20e086ea9f8f62b486172c52b5b2e6	9 years ago
George Joseph	6b15c7fc05	pjproject_bundled: Add --disable-libwebrtc to configure ... Without the disable, pjproject tries to build it's internal webrtc implementation which requires sse2. This fails on platforms without sse2. ASTERISK-26930 #close Reported-by: abelbeck Change-Id: I07231f9160c35cfa42b194d3aad4e7d51fd9a410	9 years ago
Jenkins2	7b7d0571f7	Merge "alembic: Add table for 'resource_list' PJSIP RLS type." into 14	9 years ago
Jenkins2	3e791349af	Merge "res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions." into 14	9 years ago
George Joseph	89d10f40b7	Merge "res_hep: Add additional config initialization and validation" into 14	9 years ago
Sean Bright	4cb1458245	cleanup: Fix fread() and fwrite() error handling ... Cleaned up some of the incorrect uses of fread() and fwrite(), mostly in the format modules. Neither of these functions will ever return a value less than 0, which we were checking for in some cases. I've introduced a fair amount of duplication in the format modules, but I plan to change how format modules work internally in a subsequent patch set, so this is simply a stop-gap. Change-Id: I8ca1cd47c20b2c0b72088bd13b9046f6977aa872	9 years ago
George Joseph	64f759c3e8	Merge "res_pjsip_session.c: Restructure ast_sip_session_alloc()" into 14	9 years ago
Joshua Colp	1174b566ab	alembic: Add table for 'resource_list' PJSIP RLS type. ... This change adds an Alembic migration which adds a ps_resource_list table that can contain resource_list RLS configuration objects. ASTERISK-26929 Change-Id: I7c888fafc67b3e87012de974f71ca7a5b8b1ec05	9 years ago
Sean Bright	03fb67f027	res_hep: Add additional config initialization and validation ... * Initialize hepv3_runtime_data.sockfd to -1 so that our ao2 destructor does not close fd 0 * Add logging output when the required option - capture_address - is not specified. * Remove a no longer relevant #define and correct related documentation * Pass appropriate flags to aco_option_register so that capture_address cannot be the empty string. ASTERISK-26953 #close Change-Id: Ief08441bc6596d6f1718fa810e54a5048124f076	9 years ago
Sean Bright	d757a70156	core: Use eventfd for alert pipes on Linux when possible ... The primary win of switching to eventfd when possible is that it only uses a single file descriptor while pipe() will use two. This means for each bridge channel we're reducing the number of required file descriptors by 1, and - if you're using timerfd - we also now have 1 less file descriptor per Asterisk channel. The API is not ideal (passing int arrays), but this is the cleanest approach I could come up with to maintain API/ABI. I've also removed what I believe to be an erroneous code block that checked the non-blocking flag on the pipe ends for each read. If the file descriptor is 'losing' its non-blocking mode, it is because of a bug somewhere else in our code. In my testing I haven't seen any measurable difference in performance. Change-Id: Iff0fb1573e7f7a187d5211ddc60aa8f3da3edb1d	9 years ago
George Joseph	a40f2c8246	Merge "pbx: Use same thread if AST_OUTGOING_WAIT_COMPLETE specified" into 14	9 years ago
Richard Mudgett	85bdf33808	res_pjsip_session.c: Send 100 Trying out earlier to prevent retransmissions. ... If ICE is enabled and a STUN server does not respond then we will block until we give up on the STUN response. This will take nine seconds. In the mean time the peer that sent the INVITE will send retransmissions. * Restructure res_pjsip_session.c:new_invite() to send a 100 Trying out earlier to prevent these retransmissions. ASTERISK-26890 Change-Id: Ie3fc611e53a0eff6586ad55e4aacad81cf6319a8	9 years ago
Richard Mudgett	47bc32173f	res_pjsip_session.c: Restructure ast_sip_session_alloc() ... * Restructure ast_sip_session_alloc() to need less cleanup on off nominal error paths. * Made ast_sip_session_alloc() and ast_sip_session_create_outgoing() avoid unnecessary ref manipulation to return a session. This is faster than calling a function. That function may do logging of the ref changes with REF_DEBUG enabled. Change-Id: I2a0affc4be51013d3f0485782c96b8fee3ddb00a	9 years ago
George Joseph	f9b6e9e439	Merge "rtp_engine/res_rtp_asterisk: Fix RTP struct reentrancy crashes." into 14	9 years ago
George Joseph	d072e3211d	Merge "build: Update config.guess and config.sub" into 14	9 years ago
George Joseph	757491d315	Merge "make ari-stubs so doc periodic jobs can run" into 14	9 years ago