asterisk

Commit Graph

Author	SHA1	Message	Date
Alexander Traud	3f720155b7	chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers. Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those codecs, which the caller did not request/support. That fix was not complete because on the second Session Timer all codecs were sent again. Some VoIP/SIP clients interpreted that complete codec-list as a change in the SIP session. Because of that, Asterisk did not send the RTP audio via NAT anymore which created a non-audio scenario after the second Session Timer fired. ASTERISK-24543 #close Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66	9 years ago
Gianluca Merlo	894071ea2c	config: fix flags in uint option handler The configuration unsigned integer option handler sets flags for the parser as if the option should be a signed integer (PARSE_INT32), leading to errors on "out of range" values. Fix flags (PARSE_UINT32). A fix to res_pjsip is also present which stops invalid flags from being passed when registering sorcery object fields for qualify status. ASTERISK-25612 #close Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e	9 years ago
Walter Doekes	13cdf3e8a1	musiconhold: Only warn if music class is not found in memory and database. The log message when a MusicOnHold music class was not found was changed from debug level to WARNING level in Asterisk 11.19 and 13.5. For those using realtime musiconhold, this message is wrong because it warns before checking the database. This changeset delays the warning until after the database has been checked. Reported-by: Conrad de Wet ASTERISK-25444 #close Change-Id: I6cfb2db2f9cfbd2bb3d30566ecae361c4abf6dbf	9 years ago
Walter Doekes	87c9ab97ea	core/logging: Fix broken syslog levels on older glibc. The fix to ASTERISK-25407 introduced the usage of LOG_MAKEPRI. However this macro is broken in older glibc (< 2.17); it would left-shift the facility a second time, causing the resultant priority to become invalid. The syslog manpage mentions nothing about LOG_MAKEPRI and suggests this: The priority argument is formed by ORing the facility and the level values [...]. ASTERISK-25510 #close Reported by: Michael Newton Change-Id: Ia89debe7fac5ad090c7ef595c0707f31bb1e3d03	9 years ago
Joshua Colp	a72f3b5bb4	tests/test_http_media_cache: Fix file descriptor leak in test. Change-Id: Ie8a9ae3d13bdeaacafc8d28271adc6707f633a5f	9 years ago
Joshua Colp	d7ee89b499	Merge "main/app: Only look to end of file if ':end' is specified, and not just ':'"	9 years ago
Joshua Colp	b1c688acab	Merge "main/file: Add the ability to play media in the media cache"	9 years ago
zuul	d82e3a1f28	Merge "tests/test_http_media_cache: Add unit tests for res_http_media_cache"	9 years ago
zuul	58c4d48e11	Merge "res/res_http_media_cache: Add an HTTP(S) backend for the core media cache"	9 years ago
zuul	88cc68d9ed	Merge "main/media_cache: Provide an extension on the local file associated with a URI"	9 years ago
zuul	c530840dbd	Merge "funcs/func_curl: Add the ability for CURL to download and store files"	9 years ago
Matt Jordan	13efea24f7	main/app: Only look to end of file if ':end' is specified, and not just ':' There is a little known feature in app_controlplayback that will cause the specified offset to be used relative to the end of a file if a ':end' is detected within the filename. This feature is pretty bad, but okay. However, a bug exists in this code where a ':' detected in the filename will cause the end pointer to be non-NULL, even if the full ':end' isn't specified. This causes us to treat an unspecified offset (0) as being "start playing from the end of the file", resulting in no file playback occurring. This patch fixes this bug by resetting the end pointer if ':end' is not found in the filename. Change-Id: Ib4c7b1b45283e4effd622a970055c51146892f35	9 years ago
Matt Jordan	ca14b99e6e	main/file: Add the ability to play media in the media cache This patch allows applications/APIs that access media through the core file APIs to play media in the media cache. Prior to determining if a 'filename' exists, the filename is passed to the media cache's retrieve API call. If that call succeeds, the local file specified passed back by the API is opened for streaming. When used in this fashion, the 'filename' is actually a URI that the media cache process and understand. ASTERISK-25654 #close Change-Id: I73b6e2e90c3e91b8500581c45cdf9c0dc785f5f0	9 years ago
Matt Jordan	01962a3932	tests/test_http_media_cache: Add unit tests for res_http_media_cache This patch adds unit tests for res_http_media cache, that covers nominal creation and retrieval - and through them as well, staleness and deletion checks. In addition, this patch adds tests that covers the interaction of various HTTP headers, including Expires, Etag, and Cache-Control. ASTERISK-25654 Change-Id: I2db101e307c863857fe416d6f5bf4cace9ac7cf5	9 years ago
Matthew Jordan	22e2340813	res/res_http_media_cache: Add an HTTP(S) backend for the core media cache This patch adds a bucket backend for the core media cache that interfaces to a remote HTTP server. When a media item is requested in the cache, the cache will query its bucket backends to see if they can provide the media item. If that media item has a scheme of HTTP or HTTPS, this backend will be invoked. The backend provides callbacks for the following: * create - this will always retrieve the URI specified by the provided bucket_file, and store it in the file specified by the object. * retrieve - this will pull the URI specified and store it in a temporary file. It is then up to the media cache to move/rename this file if desired. * delete - destroys the file associated with the bucket_file. * stale - if the bucket_file has expired, based on received HTTP headers from the remote server, or if the ETag on the server no longer matches the ETag stored on the bucket_file, the resource is determined to be stale. Note that the backend respects the ETag, Expires, and Cache-Control headers provided by the HTTP server it is querying. ASTERISK-25654 Change-Id: Ie201c2b34cafc0c90a7ee18d7c8359afaccc5250	9 years ago
Matt Jordan	791b4c9f81	main/media_cache: Provide an extension on the local file associated with a URI This patch does the following: First, it addresses file extension handling in the media cache. The media core in Asterisk is a bit interesting in that it wants: * A file to have an extension on it. That extension is used to associate the file with a defined format module. * The filename passed to the core to not have an extension on it. This allows the core to match the available file formats with the format a channel is capable of handling. Unfortunately, this makes the current implementation a bit lacking in the media cache. By default, we do not store the extension of a retrieved URI on the local file that is created. As a result, the media core does not know what format the file is, and the file is ignored. Modifying the file outside of the media core is bad, as we would not be able to update the internal ast_bucket_file's path. At the same time, we do not want to pass the extension out in the file_path parameter in ast_media_cache_retrieve. This parameter is intended to be fed into the media core; if we passed the extension, all callers would have to strip it off. Thus, this patch does the following: * If there is an extension specified in the URL, we append it to the local file name (if a preferred file name isn't specified), and we store that in the local file path. * The extension, however, is stripped off of the file_path parameter passed back out of ast_media_cache_retrieve. Second, this patch causes stale items to be completely removed from the system. Prior to this patch, sound files could be orphaned due to the bucket referencing the file being deleted, but the file itself not being removed. This is now addressed by explicitly calling ast_bucket_file_delete on the bucket_file when it is deemed to be stale. Note that this only happen when we know we will attempt to retrieve the resource again. Finally, this patch changes the AO2 container holding media items to just use a regular mutex. The usage for this container already assumed it was a plain mutex, and - given that retrieval of an item can cause it to be replaced in the container - a mutex makes more sense than a read/write lock. Change-Id: I51667fff86ae8d2e4a663555dfa85b11e935fe0f	9 years ago
Matthew Jordan	6bbcfb34bd	funcs/func_curl: Add the ability for CURL to download and store files This patch adds a write option to the CURL dialplan function, allowing it to CURL files and store them locally. The value 'written' to the CURL URL specifies the location on disk to store the file. As an example: same => n,Set(CURL(http://1.1.1.1/foo.wav)=/tmp/foo.wav) Would retrieve the file foo.wav from the remote server and store it in the /tmp directory. Due to the potentially dangerous nature of this function call, APIs are forbidden from using the write functionality unless live_dangerously is set to True in asterisk.conf. ASTERISK-25652 #close Change-Id: I44f4ad823d7d20f04ceaad3698c5c7f653c41b0d	9 years ago
George Joseph	392341ba37	pjproject-bundled: Cleanups for reported issues PortAudio should no longer be required PJSIP_MAX_PKT_LEN is now 6000 Older autoconf issue fixed. (CentOS 6) Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd	9 years ago
Francesco Castellano	ac66999971	chan_sip.c: Space after port causes unnecessary resolution attempt check_via() already skips leading blanks where the sent-by address (with the optional port) should be placed. Since RFC 3261 allows for blanks between the port ant the Via parameters: > https://tools.ietf.org/html/rfc3261#section-20.42 (actually it allows a lot of blanks more ;-)). I just switched from ast_skip_blanks() to ast_strip() on the local copy of the string. ASTERISK-21301 #close Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06	9 years ago
zuul	c21cee80cc	Merge "func_aes: fix misuse of strlen on binary data"	9 years ago
George Joseph	1d3191b118	progdocs: Exclude ./third-party from documentation generation We don't need pjproject's documentation embedded in Asterisk's. Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17	9 years ago
Gianluca Merlo	8f94f947f5	func_aes: fix misuse of strlen on binary data The encryption code for AES_ENCRYPT evaluates the length of the data to be encoded in base64 using strlen. The data is binary, thus the length of it can be underestimated at the first NULL character. Reuse the write pointer offset to evaluate it, instead. ASTERISK-25857 #close Change-Id: If686b5d570473eb926693c73461177b35b13b186	9 years ago
Kevin Harwell	a3c9a74a02	chan_pjsip: ref leak when checking direct_media_glare Fix the reference leak introduced in the following commit: `c534bd5807` ASTERISK-25849 Change-Id: I5cfefd5ee6c1c3a1715c050330aaa10e4d2a5e85	9 years ago
zuul	4aaf8f2ace	Merge "chan_pjsip: transfers with direct media reinvite has wrong address/port"	9 years ago
Kevin Harwell	c534bd5807	chan_pjsip: transfers with direct media reinvite has wrong address/port During a transfer involving direct media a race occurs between when the transferer channel is swapped out, initiating rtp changes/updates, and the subsequent reinvites. When Alice, after speaking with Charlie (Bob is on hold), connects Bob and Charlie invites are sent to each in order to establish the call between them. Bob is taken off hold and Charlie is told to have his media flow through Asterisk. However, if before those invites go out the bridge updates Bob's and/or Charlie's rtp information with direct media data (i.e. address, port) then the invite(s) will contain the remote data in the SDP instead of the Asterisk data. The race occurs in the native bridge glue code when updating the peer. The direct_media_address can get set twice before sending out the first invite during call connection. This can happen because the checking/setting of the direct_media_address happened in one thread while the sending of the invite(s) happened in another thread. This fix removes the race condition by moving the checking/setting of the direct_media_address to be in the same thread as the sending of the invites(s). This serializes the checking/setting and sending so they can no longer happen out of order. ASTERISK-25849 #close Change-Id: Idfea590175e74f401929a601dba0c91ca1a7f873	9 years ago
Sergio Medina Toledo	bdccb81157	res_pjsip_refer.c: Fix seg fault in process of Refer-to header. The "Refer-to" header of an incoming REFER request is parsed by pjsip_parse_uri(). That function requires the URI parameter to be NULL terminated. Unfortunately, the previous code added the NULL terminator by overwriting memory that may not be safe. The overwritten memory results could be benign, memory corruption, or a segmentation fault. Now the URI is NULL terminated safely by copying the URI to a new chunk of memory with the correct size to be NULL terminated. ASTERISK-25814 #close Change-Id: I32565496684a5a49c3278fce06474b8c94b37342	9 years ago
Joshua Colp	f557843323	Merge "Add initial support to build Docker images"	9 years ago
Joshua Colp	7d63b4499e	Merge "chan_sip.c: Made sip_reinvite_retry() call sip_pvt_lock_full()."	9 years ago
Leif Madsen	0da36fca6b	Add initial support to build Docker images This work-in-progress is the first step to being able to reliably build Asterisk containers from the Asterisk source. I'm submitting this based on feedback gained at AstriDevCon 2015. Information about how to use this is provided in contrib/docker/README.md and will result in a local Asterisk container being built right from your source. I believe this can eventually be automated via hub.docker.com. Change-Id: Ifa070706d40e56755797097b6ed72c1e243bd0d1	9 years ago
Richard Mudgett	810f92c9dc	chan_sip.c: Fix mwi resub deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 #close Change-Id: I96d429c57a48861fd8bde63dd93db4e92dc3adb6	9 years ago
Richard Mudgett	72c444ba37	chan_sip.c: Fix registration timeout and expire deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I2e40de89efc8ae6e8850771d089ca44bc604b508	9 years ago
Richard Mudgett	7ea1e181dc	chan_sip.c: Fix waitid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Made always run check_pendings() under the scheduler thread so scheduler ids can be checked safely. ASTERISK-25023 Change-Id: Ia834d6edd5bdb47c163e4ecf884428a4a8b17d52	9 years ago
Richard Mudgett	fbf8e04aed	chan_sip.c: Fix t38id deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: If595e4456cd059d7171880c7f354e844c21b5f5f	9 years ago
Richard Mudgett	02458cc6fd	chan_sip.c: Fix session timers deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I6d65269151ba95e0d8fe4e9e611881cde2ab4900	9 years ago
Richard Mudgett	c7fdff2e37	chan_sip.c: Fix reinviteid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I9c11b9d597468f63916c99e1dabff9f4a46f84c1	9 years ago
Richard Mudgett	69810b306d	chan_sip.c: Fix autokillid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Fix clearing autokillid in __sip_autodestruct() even though we could reschedule. ASTERISK-25023 Change-Id: I450580dbf26e2e3952ee6628c735b001565c368f	9 years ago
Richard Mudgett	f484ddbdfe	chan_sip.c: Fix packet retransid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Fix retrans_pkt() to call check_pendings() with both the owner channel and the private objects locked as required. * Refactor dialog retransmission packet list to safely remove packet nodes. The list nodes are now ao2 objects. The list has a ref and the scheduled entry has a ref. ASTERISK-25023 Change-Id: I50926d81be53f4cd3d572a3292cd25f563f59641	9 years ago
Richard Mudgett	67c79c326d	chan_sip.c: Fix provisional_keepalive_sched_id deadlock. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I98a694fd42bc81436c83aa92de03226e6e4e3f48	9 years ago
Richard Mudgett	76be7093cd	chan_sip.c: Adjust how dialog_unlink_all() stops scheduled events. This patch is part of a series to resolve deadlocks in chan_sip.c. * Make dialog_unlink_all() unschedule all items at once in the sched thread. ASTERISK-25023 Change-Id: I7743072fb228836e8228b72f6dc46c8cc50b3fb4	9 years ago
Richard Mudgett	52f0932e4c	chan_sip.c: Clear scheduled immediate events on unload. This patch is part of a series to resolve deadlocks in chan_sip.c. The reordering of chan_sip's shutdown is to handle any immediate events that get put onto the scheduler so resources aren't leaked. The typical immediate events at this time are going to be concerned with stopping other scheduled events. ASTERISK-25023 Change-Id: I3f6540717634f6f2e84d8531a054976f2bbb9d20	9 years ago
Richard Mudgett	0987a11cce	sip/dialplan_functions.c: Fix /channels/chan_sip/test_sip_rtpqos crash. This patch is part of a series to resolve deadlocks in chan_sip.c. Delaying destruction of the chan_sip sip_pvt structures caused the /channels/chan_sip/test_sip_rtpqos unit test to crash. That test registers a special test ast_rtp_engine with the rtp engine module. When the unit test completes it cleans up by unregistering the test ast_rtp_engine and exits. Since the delayed destruction of the sip_pvt happens after the unit test returns, the destructor tries to call the rtp engine destroy callback of the test ast_rtp_engine auto variable which no longer exists on the stack. * Change the test ast_rtp_engine auto variable to a static variable. Now the variable can still exist after the unit test exits so the delayed sip_pvt destruction can complete successfully. ASTERISK-25023 Change-Id: I61e34a12d425189ef7e96fc69ae14993f82f3f13	9 years ago
Richard Mudgett	9a7cfa2b61	sched.c: Ensure oldest expiring entry runs first. This patch is part of a series to resolve deadlocks in chan_sip.c. * Updated sched unit test to check new behavior. ASTERISK-25023 Change-Id: Ib69437327b3cda5e14c4238d9ff91b2531b34ef3	9 years ago
zuul	20d67cc6f5	Merge "app_stasis: Don't hang up if app is not registered"	9 years ago
zuul	3fa6076215	Merge "chan_sip.c: Simplify sip_pvt destructor call levels."	9 years ago
Andrew Nagy	7964e260d3	app_stasis: Don't hang up if app is not registered This prevents pbx_core from hanging up the channel if the app isn't registered. ASTERISK-25846 #close Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce	9 years ago
zuul	b567181402	Merge "pjproject: Pass (dont_)optimize flags to pjproject and fix pjsua"	9 years ago
Joshua Colp	d108bee11b	Merge "build_system: Split COMPILE_DOUBLE from DONT_OPTIMIZE"	9 years ago
zuul	d804b904c5	Merge "build: Add configure check for proto field of PJSIP TLS transport setting."	9 years ago
Joshua Colp	abe893725b	Merge "res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100"	9 years ago
Richard Mudgett	cb97198ca6	chan_sip.c: Simplify sip_pvt destructor call levels. Remove destructor calling destroy_it calling really_destroy_it for no benefit. Just make the destructor the really_destroy_it function. Change-Id: Idea0d47b27dd74f2488db75bcc7f353d8fdc614a	9 years ago

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