29392 Commits (1a1c86239d9aa64dc7e9301ea03326dd5436931c)

 

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Author	SHA1	Message	Date
George Joseph	1a1c86239d	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	3dae4279be	Merge "func_cdr: Allow empty value for CDR dialplan function."	9 years ago
Joshua Colp	c62b5721b3	Merge "stream: ast_stream_clone() cannot copy the opaque user data."	9 years ago
Joshua Colp	d96f755682	Merge "netsock2.c: Made get/set addr port avoid potential uninitialized memory."	9 years ago
Joshua Colp	552e6d81ef	Merge "bridge: Fix returning to dialplan when executing Bridge() from AMI."	9 years ago
Richard Mudgett	56c5c51076	stream: ast_stream_clone() cannot copy the opaque user data. ... ast_stream_clone() cannot copy the opaque user data stored on a stream. We don't know how to clone the data so it isn't copied into the clone. Change-Id: Ia51321bf38ecbfdcc53787ca77ea5fd2cabdf367	9 years ago
Richard Mudgett	924628812b	netsock2.c: Made get/set addr port avoid potential uninitialized memory. ... Change-Id: I532052bd7cd95a4b3565485fc01e2a1ea07ee647	9 years ago
Joshua Colp	4146facfec	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	0001834157	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
Jenkins2	a20db27c56	Merge "SDP: Replace SDP telephone_event option with dtmf option"	9 years ago
Jenkins2	4262a5aa6a	Merge "res_rtp_asterisk: Clearing the remote RTCP address causes RTCP failures"	9 years ago
Joshua Colp	c90d81ef51	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	7b0e3b92fd	bridge_simple: Added support for streams ... This patch is the first cut at adding stream support to the bridging framework. Changes were made to the framework that allows mapping of stream topologies to a bridge's supported media types. The first channel to enter a bridge initially defines the media types for a bridge (i.e. a one to one mapping is created between the bridge and the first channel). Subsequently added channels merge their media types into the bridge's adding to it when necessary. This allows channels with different sized topologies to map correctly to each other according to media type. The bridge drops any frame that does not have a matching index into a given write stream. For now though, bridge_simple will align its two channels according to size or first to join. Once both channels join the bridge the one with the most streams will indicate to the other channel to update its streams to be the same as that of the other. If both channels have the same number of streams then the first channel to join is chosen as the stream base. A topology change source was also added to a channel when a stream toplogy change request is made. This allows subsystems to know whether or not they initiated a change request. Thus avoiding potential recursive situations. ASTERISK-26966 #close Change-Id: I1eb5987921dd80c3cdcf52accc136393ca2d4163	9 years ago
Kevin Harwell	008e25def9	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
Joshua Colp	bdec0852b9	Merge "channels/chan_sip.c: use binding IP address for outgoing TCP SIP connections"	9 years ago
Sean Bright	675e058e77	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
Richard Mudgett	cd272da7a8	SDP: Replace SDP telephone_event option with dtmf option ... The telephone_event option was used as a flag and a bit mapped value in different places when it is a boolean. It is also inadequate to configure the DTMF operation of the RTP instance created for the stream. Change-Id: Ib1addeaf0ce86f07039f2f979cab29405dc5239b	9 years ago
Jenkins2	dc948163ca	Merge "res_pjsip_t38.c: Fix deadlock in T.38 framehook."	9 years ago
Joshua Colp	d1944c1892	Merge "res_sdp_translator_pjmedia.c: Add TODO notes."	9 years ago
Joshua Colp	1d6429b269	Merge "SDP: Make SDP translation to/from internal representation more const."	9 years ago
Joshua Colp	090c6b702e	Merge "stream: Make ast_stream_topology_create_from_format_cap() allow NULL cap."	9 years ago
Jenkins2	9af53d3563	Merge "SDP: Make ast_sdp_state_set_remote_sdp() return error."	9 years ago
Jenkins2	b67423c8a3	Merge "res_pjsip_outbound_authenticator_digest: Add context to log messages"	9 years ago
Jenkins2	74134a03bc	Merge "SDP: Misc cleanups (Mostly memory leaks)"	9 years ago
Jenkins2	94b97e0835	Merge "SDP API: Add SSRC-level attributes"	9 years ago
Richard Mudgett	52e4f02b1a	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	8170793be6	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	09cde039a3	Merge "chan_vpb.cc: Fix compile error."	9 years ago
George Joseph	07164d04a6	Merge "chan_sip: Trigger reinvite if the SDP answer is included in the SIP ACK"	9 years ago
Richard Mudgett	48566b8c66	res_sdp_translator_pjmedia.c: Add TODO notes. ... Change-Id: If27ca61f79accc882c3376d2e876d2b44aa1347b	9 years ago
Richard Mudgett	ede90e4aa5	SDP: Make SDP translation to/from internal representation more const. ... Change-Id: I473a174b869728604b37c60853896b0c458bc504	9 years ago
Richard Mudgett	5c1851cbc0	stream: Make ast_stream_topology_create_from_format_cap() allow NULL cap. ... Change-Id: Ie29760c49c25d7022ba2124698283181a0dd5d08	9 years ago
Richard Mudgett	d71c6e3bfd	SDP: Make ast_sdp_state_set_remote_sdp() return error. ... Change-Id: I7707c9d872c476d897ff459008652b35142a35e1	9 years ago
Richard Mudgett	176123e76c	SDP: Misc cleanups (Mostly memory leaks) ... Change-Id: I74431b385da333f2c5f5a6d7c55e70b69a4f05d2	9 years ago
Richard Mudgett	bad091b317	chan_vpb.cc: Fix compile error. ... Change-Id: I6d9edd34d8b2474222c86f44e379ead61e57a54f	9 years ago
Jenkins2	528e238447	Merge "channel: Add ability to request an outgoing channel with stream topology."	9 years ago
Jenkins2	16089ae1c9	Merge "frame: Better handle interpolated frames."	9 years ago
Jenkins2	066659a383	Merge "res_pjsip_session: Add cleanup to ast_sip_session_terminate"	9 years ago
Jenkins2	175297fe34	Merge "res_pjsip/res_pjsip_callerid: NULL check on caller id name string"	9 years ago
Jenkins2	49b2d1bde5	Merge "vector: defaults and indexes"	9 years ago
Mark Michelson	d6535c0080	SDP API: Add SSRC-level attributes ... RFC 5576 defines how SSRC-level attributes may be added to SDP media descriptions. In general, this is useful for grouping related SSRCes, indicating SSRC-level format attributes, and resolving collisions in RTP SSRC values. These attributes are used widely by browsers during WebRTC communications, including attributes defined by documents outside of RFC 5576. This commit introduces the addition of SSRC-level attributes into SDPs generated by Asterisk. Since Asterisk does not tend to use multiple SSRCs on a media stream, the initial support is minimal. Asterisk includes an SSRC-level CNAME attribute if configured to do so. This at least gives browsers (and possibly others) the ability to resolve SSRC collisions at offer-answer time. In order to facilitate this, the RTP engine API has been enhanced to be able to retrieve the SSRC and CNAME on a given RTP instance. res_rtp_asterisk currently does not provide meaningful CNAME values in its RTCP SDES items, and therefore it currently will always return an empty string as the CNAME value. A task in the near future will result in res_rtp_asterisk generating more meaningful CNAMEs. Change-Id: I29e7f23e7db77524f82a3b6e8531b1195ff57789	9 years ago
Joshua Colp	413de95ab2	Merge "cleanup: Fix fread() and fwrite() error handling"	9 years ago
Jenkins2	623bbcbed2	Merge "pjproject_bundled: Add --disable-libwebrtc to configure"	9 years ago
George Joseph	d6b2a58736	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	54e27cad3c	Merge "res_rtp_asterisk.c: Fix crash in RTCP DTLS operation."	9 years ago
Joshua Colp	2b22c3c84b	channel: Add ability to request an outgoing channel with stream topology. ... This change extends the ast_request functionality by adding another function and callback to create an outgoing channel with a requested stream topology. Fallback is provided by either converting the requested stream topology into a format capabilities structure if the channel driver does not support streams or by converting the requested format capabilities into a stream topology if the channel driver does support streams. The Dial application has also been updated to request an outgoing channel with the stream topology of the calling channel. ASTERISK-26959 Change-Id: Ifa9037a672ac21d42dd7125aa09816dc879a70e6	9 years ago
Joshua Colp	78eb08e7ba	Merge "sdp: Add support for T.38"	9 years ago
Joshua Colp	ed69471f94	Merge "SDP: Ensure SDPs "merge" properly."	9 years ago
Kevin Harwell	c6b757fa05	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	cf3429b934	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