28164 Commits (05cfe1a76e7c7e0162b3196c8ff16c064726f53f)

 

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Author	SHA1	Message	Date
Joshua Colp	58aef11dc1	Merge "pjproject_bundled: Move to pjproject 2.5"	9 years ago
Richard Mudgett	40d19f2e55	logging,cdr,cel: Fix stringfield memory leak. ... The stringfields refactor to allow adding stringfields to the end of a structure (f6f4cf459f) exposed some incomplete cleanup code by some stringfield users. The most noticeable leaker is the logging system where there is a leak for every log message generated. ASTERISK-26078 #close Reported by: Etienne Lessard Patches: jira_asterisk_26078_v13.patch (license #5621) patch uploaded by Richard Mudgett Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782	9 years ago
Joshua Colp	608e0267e8	Merge "Expand the scope of Dial Events"	9 years ago
Richard Mudgett	aec7916595	pjsip_distributor.c: Use correct rdata info access method (Part 2). ... The pjproject doxygen for rdata->msg_info.info says to call pjsip_rx_data_get_info() instead of accessing the struct member directly. You need to call the function mostly because the function will generate the struct member value if it is not already setup. Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799	9 years ago
Joshua Colp	ad31e5bb1c	Merge "followme: allow disabling callee prompt"	9 years ago
zuul	84a93b0d67	Merge "ARI: Re-implement the ARI dial command, allowing for early bridging."	9 years ago
zuul	dffbb2d7c3	Merge "res_pjsip_mwi_body_generator: Re-order the body items"	9 years ago
Mark Michelson	205a31f86c	Expand the scope of Dial Events ... Dial events up to this point have come in two flavors * A Dial event with no status to indicate that dialing has begun * A Dial event with a status to indicate that dialing has ended With this change, Dial events have been expanded to also give intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS". This is especially useful for ARI dialing, as it gives the application writer the opportunity to place a channel into an early bridge when early media is detected. AMI handles these in-progress dial events by sending a new event called "DialState" that simply indicates that dial state has changed but has not ended. ARI never distinguished between DialBegin and DialEnd, so no change was made to the event itself. Another change here relates to dial forwards. A forward-related event was previously only sent when a channel was successfully able to forward a call to a new channel. With this set of changes, if forwarding is blocked, we send a Dial event with a forwarding destination but no forwarding channel, since we were prevented from creating one. This is again useful for ARI since application writers can now handle call forward attempts from within their own application. ASTERISK-25925 #close Reported by Mark Michelson Change-Id: I42cbec7730d84640a434d143a0d172a740995543	9 years ago
Joshua Colp	11ea121cc8	Merge "res_pjsip: add "via_addr", "via_port", "call_id" to contact"	9 years ago
zuul	853350b5ea	Merge "res_pjsip: Add clarifying documentation to PJSIP_HEADER help text"	9 years ago
zuul	6db496c788	Merge "multicast RTP: Add dialing options"	9 years ago
zuul	47c4f52df6	Merge "res_pjsip: chatty verbose messages"	9 years ago
George Joseph	8a6a14590d	res_pjsip_mwi_body_generator: Re-order the body items ... Re-ordered the body items so Message-Account is second. Messages-Waiting: no Message-Account: sip:1571@<IP Removed>:5060 Voice-Message: 0/0 (0/0) ASTERISK-26065 #close Reported-by: Ross Beer Change-Id: If5d35a64656eac98c2dd5e490cc0b2807bed80c3	9 years ago
George Joseph	7fa5766752	pjproject_bundled: Move to pjproject 2.5 ... Although all the patches we had against 2.4.5 were applied by Teluu, a new bug was introduced preventing re-use of tcp and tls transports This patch removes all the previous patches against 2.4.5, updates the version to 2.5, and adds a new patch to correct the transport re-use problem. Change-Id: I0dc6c438c3910f7887418a5832ca186aea23d068	9 years ago
Rusty Newton	b56f611856	res_pjsip: Add clarifying documentation to PJSIP_HEADER help text ... Added notes about when you can read or write headers. Specifically about being able to read on the inbound channel and write on an outbound channel. ASTERISK-26063 #close Reported by: Private Name Tested by: Rusty Newton Change-Id: Ibeb64af17d1f6451028b3c29855a3f151a01d8c5	9 years ago
Mark Michelson	bb0f4a6310	multicast RTP: Add dialing options ... This adds a new parameter to the end of a multicast RTP dialing string. This parameter defines the following options: * i: Set the interface from which multicast RTP is sent * l: Set whether multicast packets are looped back to the sender * t: Set the TTL for multicast packets * c: Set the codec to use for RTP ASTERISK-26068 #close Reported by Mark Michelson Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219	9 years ago
Mark Michelson	88d997913f	ARI: Re-implement the ARI dial command, allowing for early bridging. ... ARI dial had been implemented using the Dial API. This made great sense when dialing was 100% separate from bridging. However, if a channel were to be added to a bridge during the dial attempt, there would be a conflict between the dialing thread and the bridging thread. Each would be attempting to read frames from the dialed channel and act on them. The initial attempt to make the two play nice was to have the Dial API suspend the channel in the bridge and stay in charge of the channel until the dial was complete. The problem with this was that it was riddled with potential race conditions. It also was not well-suited for the case where the channel changed which bridge it was in during the dial. This new approach removes the use of the Dial API altogether. Instead, the channel we are dialing is placed into an invisible ARI dialing bridge. The bridge channel thread handles incoming frames from the channel. If the channel is added to a real bridge, it is departed from the invisible bridge and then added to the real bridge. Similarly, if the channel is removed from the real bridge, it is automatically added back to the invisible bridge if the dial attempt is still active. This approach keeps the threading simple by always having the channel being handled by bridge channel threads. ASTERISK-25925 Change-Id: I7750359ddf45fcd45eaec749c5b3822de4a8ddbb	9 years ago
zuul	de29888904	Merge "app_voicemail: fix bugs, imap mm_status log change to debug"	9 years ago
Alexei Gradinari	31f17abe44	res_pjsip: add "via_addr", "via_port", "call_id" to contact ... As res_pjsip_nat rewrites contact's address, only the last Via header can contain the source address of registered endpoint. Also Call-Id header may contain the source address of registered endpoint. Added "via_addr", "via_port", "call_id" to contact. Added new fields ViaAddress, CallID to AMI event ContactStatus. ASTERISK-26011 Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576	9 years ago
Alexei Gradinari	574c9e77eb	res_pjsip: chatty verbose messages ... There are a lot of verbose messages about Endpoint and Contact status changes if there are many dynamic endpoints. The patch sets verbose level 2 for Endpoint status changes and verbose level 3 for Contact status changes. ASTERISK-26055 #close Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7	9 years ago
Alexei Gradinari	b3142e99e4	app_voicemail: fix bugs, imap mm_status log change to debug ... Fixed some bugs: - create dirpath when save downloading message from IMAP storage. - create IMAP folder if not exists when saving to IMAP storage - check if file successfully opened before write to it - some IMAP checks - remove non-standard flag 'Unseen' etc Change to debug IMAP mm_status log instead of verbose. Remove unused X-Asterisk-VM-Caller-channel message header for security reason. The clients should not know name of peer/endpoint. ASTERISK-26045 #close Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b	9 years ago
Richard Mudgett	7d44d12816	pjsip_distributor.c: Use correct rdata info access method. ... The pjproject doxygen for rdata->msg_info.info says to call pjsip_rx_data_get_info() instead of accessing the struct member directly. You need to call the function mostly because the function will generate the struct member value if it is not already setup. Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2	9 years ago
Tzafrir Cohen	1d60bfcdf1	followme: allow disabling callee prompt ... Add the option 'enable_callee_prompt' to followme.conf. Enabled by default. If disabled, a callee is not prompted to accept or reject the forwarded call. ASTERISK-26064 #close Change-Id: I0a8b19d4cf95c86a07c992813babb9e4a4acfff5 Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	9 years ago
zuul	a6b16d7029	Merge "res_pjsip_outbound_publish: Ensure publish is valid when explicitly destroying."	9 years ago
Joshua Colp	b0e4ea96de	Merge "Bridging: introduce "invisible" bridges."	9 years ago
zuul	d1ab0936ab	Merge "res_pjsip: Only check transaction on transaction state events."	9 years ago
Corey Farrell	80ff2c2540	threadpool: Fix potential data race. ... worker_start checked for ZOMBIE status without holding a lock. All other read/write of worker status are performed with a lock, so this check should do the same. ASTERISK-25777 #close Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781	9 years ago
Joshua Colp	56f24345f7	Merge "func_odbc: single database connection should be optional"	9 years ago
Joshua Colp	070eab6ed2	res_pjsip_outbound_publish: Ensure publish is valid when explicitly destroying. ... Recent changes to res_pjsip_outbound_publish have introduced a race condition at shutdown where an outbound publish may be shutdown twice. In this case the first succeeds as a result of the unpublish. In the second invocation since it's been unpublished a task is queued to just destroy the client. This task holds no ref to the publish and as a result the publish may be destroyed before the task is run, causing a crash. This explicit destruction task now holds a reference to the publish to ensure it remains valid. ASTERISK-26053 #close Change-Id: I10789b98add3e50292ee3b33a55a1d9061cec94b	9 years ago
Joshua Colp	cab97fd905	Merge "ARI: Add the ability to download the media associated with a stored recording"	9 years ago
Joshua Colp	c20e560516	Merge "chan_rtp.c: Cleanup ast_request() parameter parsing."	9 years ago
zuul	c7320aea6e	Merge "Makefile: remove OSARCH check for init install"	9 years ago
Mark Michelson	f6c33771f6	Bridging: introduce "invisible" bridges. ... Invisible bridges function the same as normal bridges, but they have the following restrictions: * They never show up in CLI, AMI, or ARI queries. * They do not have Stasis messages published about them. Invisible bridges' main use is for when use of the bridging system is desired, but the bridge should not be known to users of the Asterisk system. ASTERISK-25925 Change-Id: I804a209d3181d7c54e3d61a60eb462e7ce0e3670	9 years ago
Joshua Colp	56d5af4584	Merge "func_curl: Don't trim response text on non-ASCII characters"	9 years ago
Joshua Colp	0eb293d2c4	Merge "parking.h: Update ast_parking_park_call() doxygen to reality."	9 years ago
Joshua Colp	85d0272e76	res_pjsip: Only check transaction on transaction state events. ... The send request callback function currently assumes that it will only ever be called on transaction state changes. This is not always true. If our own timer callback occurs we will call the callback with a timer event instead of a transaction state change event. In this case the transaction on the event is invalid and accessing it will result in a crash. ASTERISK-26049 #close Change-Id: I623211c8533eb73056b0250b4580b49ad4174dfc	9 years ago
Ivan Poddubny	31897d2d99	func_curl: Don't trim response text on non-ASCII characters ... The characters 0x80-0xFF were trimmed as well as 0x00-0x20 because of a signed comparison. ASTERISK-25669 #close Reported by: Jesper patches: strings.curl.trim.patch submitted by Jesper (License 5518) Change-Id: Ia51e169f24e3252a7ebbaab3728630138ec6f60a	9 years ago
Richard Mudgett	2a77af9ed0	chan_rtp.c: Cleanup ast_request() parameter parsing. ... * Fixed NULL crash potential if parameters are missing. * Reordered some operations so further diagnostic messages can be more helpful. Change-Id: Ibbdc67a2496508cbfbfef0cf19c35177ae2fbd70	9 years ago
Richard Mudgett	ade5275a3e	parking.h: Update ast_parking_park_call() doxygen to reality. ... ASTERISK-26029 Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260	9 years ago
Alexei Gradinari	c378b00a83	func_odbc: single database connection should be optional ... func_odbc was changed in Asterisk 13.9.0 to make func_odbc use a single database connection per DSN because of reported bug ASTERISK-25938 with MySQL/MariaDB LAST_INSERT_ID(). This is drawback in performance when func_odbc is used very often in dialplan. Single database connection should be optional. ASTERISK-26010 Change-Id: I7091783a7150252de8eeb455115bd00514dfe843	9 years ago
Mark Michelson	1c02b19b79	res_pjsip: Match dialogs on responses better. ... When receiving an incoming response to a dialog-starting INVITE, we were not matching the response to the INVITE dialog. Since we had not recorded the to-tag to the dialog structure, the PJSIP-provided method to find the dialog did not match. Most of the time, this was not a problem, because there is a fall-back that makes the response get routed to the same serializer that the request was sent on. However, in cases where an asynchronous DNS lookup occurs in the PJSIP core, the thread that sends the INVITE is not actually a threadpool serializer thread. This means we are unable to record a serializer to handle the incoming response. Now, imagine what happens when an INVITE is sent on a non-serialized thread, and an error response (such as a 486) arrives. The 486 ends up getting put on some random threadpool thread. Eventually, a hangup task gets queued on the INVITE dialog serializer. Since the 486 is being handled on a different thread, the hangup task can execute at the same time that the 486 is being handled. The hangup task assumes that it is the sole owner of the INVITE session and channel, so it ends up potentially freeing the channel and NULLing the session's channel pointer. The thread handling the 486 can crash as a result. This change has the incoming response match the INVITE transaction, and then get the dialog from that transaction. It's the same method we had been using for matching incoming CANCEL requests. By doing this, we get the INVITE dialog and can ensure that the 486 response ends up being handled by the same thread as the hangup, ensuring that the hangup runs after the 486 has been completely handled. ASTERISK-25941 #close Reported by Javier Riveros Change-Id: I0d4cc5d07e2a8d03e9db704d34bdef2ba60794a0	9 years ago
Matt Jordan	e773e3a9bb	ARI: Add the ability to download the media associated with a stored recording ... This patch adds a new feature to ARI that allows a client to download the media associated with a stored recording. The new route is /recordings/stored/{name}/file, and transmits the underlying binary file using Asterisk's HTTP server's underlying file transfer facilities. Because this REST route returns non-JSON, a few small enhancements had to be made to the Python Swagger generation code, as well as the mustache templates that generate the ARI bindings. ASTERISK-26042 #close Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181	9 years ago
Joshua Colp	40cb032009	res_sorcery_astdb: Filter fields to only the registered ones. ... This change introduces the same filtering that is done in res_sorcery_realtime to the res_sorcery_astdb module. This allows persisted sorcery objects that may contain unknown fields to still be read in from the AstDB and used. This is particularly useful when switching between different versions of Asterisk that may have introduced additional fields. ASTERISK-26014 #close Change-Id: Ib655130485a3ccfd635b7ed5546010ca14690fb2	9 years ago
Joshua Colp	ff3cbc0046	Merge "res_pjsip_empty_info: Respond to empty SIP INFO packets"	9 years ago
Joshua Colp	e205eb55a4	Merge "res_pjsip: Endpoint IP Access Controls"	9 years ago
snuffy	9766a12b4c	res_pjsip_empty_info: Respond to empty SIP INFO packets ... Some SBCs require responses to empty SIP INFO packets after establishing call via INVITE, if not responded to they may drop your call after unspecified timeout of X minutes. They are identified by having no Content-Type, check for this and respond with 200 - OK message. ASTERISK-24986 #close Reported-by: Ilya Trikoz, Federico Santulli Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0	9 years ago
Joshua Colp	acbaa1b0cf	Merge "udptl: Don't eat sequence numbers until OK is received"	9 years ago
Joshua Colp	5acb25722c	Merge "logger: Support JSON logging with Verbose messages"	9 years ago
Joshua Colp	b57032c364	Merge "res_hep: Provide an option to pick the UUID type"	9 years ago
Joshua Colp	1f36270b21	Merge "res/res_hep_pjsip: Fix reported local IP address when bound to 'any'"	9 years ago