4438 Commits (33ebaae23f5f6de339c4cd3e8727a1d8a5536f25)

Author	SHA1	Message	Date
George Joseph	33ebaae23f	Merge "res_stasis: Auto-create context and extens on Stasis app launch." into 13	7 years ago
Joshua C. Colp	41da4ed8b2	Merge "media_index.c: Refactored so it doesn't cache the index" into 13	7 years ago
Friendly Automation	762f40fca0	Merge "res/res_pjsip: Fix crash due to misuse of session->media between threads." into 13	7 years ago
Ben Ford	26a04477f4	res_stasis: Auto-create context and extens on Stasis app launch. ... At AstriCon, there was a strong desire for the ability to completely bypass dialplan when using ARI. This is possible through the automatic creation of a context and a couple of extensions whenever an application is started. For example, if you have an application named 'ari-example', a context named 'stasis-ari-example' will be automatically created whenever this application is started as long as one does not already exist. Two extensions (a match-all extension for Stasis and a 'h' extension) are created within this context. Any endpoint that registers to Asterisk within this context will send all calls to the corresponding Stasis application. When the application is destroyed, the context is removed. ASTERISK-28104 #close Change-Id: Ie35bd93075e05b05e3ae129a83c9426931b7ebac	7 years ago
George Joseph	66982824bf	media_index.c: Refactored so it doesn't cache the index ... Testing revealed that the cache added no benefit but that it could consume excessive memory. Two new index related functions were created: ast_sounds_get_index_for_file() and ast_media_index_update_for_file() which restrict index updating to specific sound files. The original ast_sounds_get_index() and ast_media_index_update() calls are still available but since they no longer cache the results internally, developers should re-use an index they may already have instead of calling ast_sounds_get_index() repeatedly. If information for only a single file is needed, ast_sounds_get_index_for_file() should be called instead of ast_sounds_get_index(). The media_index directory scan code was elimininated in favor of using the existing ast_file_read_dirs() function. Since there's no more cache, ast_sounds_index_init now only registers the sounds cli commands instead of generating the initial index and subscribing to stasis format register/unregister messages. ast_sounds_reindex() is now a no-op but left for backwards compatibility. loader.c no longer registers "sounds" as a special reload target. Both the sounds cli commands and the sounds ari resources were refactored to only call ast_sounds_get_index() once per invocation and to use ast_sounds_get_index_for_file() when a specific sound file is requested. Change-Id: I1cef327ba1b0648d85d218b70ce469ad07f4aa8d	7 years ago
George Joseph	c496fc2e28	Merge "res_http_websocket: ensure control frames do not interfere with data" into 13	7 years ago
Joshua C. Colp	b13051705d	Merge "res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown" into 13	7 years ago
Paulo Vicentini	c01d2f66ee	res/res_pjsip: Fix crash due to misuse of session->media between threads. ... This patch makes sure that thread running ast_taskprocessor_execute cannot suddenly dispose the session->media object making the other threads (running pbx_thread / bridge_channel_ind_thread) crash when they try to access the pointer to invalid memory. We were experiencing a crash due to a misuse of session->media container between threads running (bridge_channel_ind_thread/pbx_thread) and the thread running ast_taskprocessor_execute. Depending on the SIP flow (during a disconnection) and the threads' code path, the session->media container was being destroyed (and set to NULL) by the thread running ast_taskprocessor_execute while the thread running t38_framehook_read was still referring to it. Now res_pjsip_t38 is referring a session_media in a datastore. ASTERISK-28156 Change-Id: Ia92e2389b8d804bf205473e92ec06217e87ce237	7 years ago
Jeremy Lainé	59ae83d07e	res_http_websocket: ensure control frames do not interfere with data ... Control frames (PING / PONG / CLOSE) can be received in the middle of a fragmented message. In order to ensure they do not interfere with the reassembly buffer, we exit early and do not return the payload to the caller. ASTERISK-28257 #close Change-Id: Ia5367144fe08ac6141bba3309517a48ec7f013bc	7 years ago
Joshua C. Colp	2c1ac524b8	Merge "stasis / manager / ari: Better filter messages." into 13	7 years ago
Friendly Automation	06df43f6dd	Merge "pjsip_transport_management: Shutdown transport immediately on disconnect" into 13	7 years ago
Kevin Harwell	2e1cbcde7d	res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown ... When a reliable transport is shutdown it's possible for the pjsip registrar resource shutdown handler to get called multiple times. If this happens and one of the threads is taking "too long" (slow database call for instance) then the others get blocked waiting to delete. Since it only takes one to delete the contact then the other threads should be able to continue on if one of the threads is currently "deleting". This patch makes it so now when a thread enters the shutdown handler it checks to see if a thread is currently already "deleting". If so, then the thread does not attempt to get the lock, and instead continues on thus avoiding the blockage. ASTERISK-28213 #close Change-Id: I7563ca596312b1dff4f3ab41483e89fe2862328a	7 years ago
Jeremy Lainé	da50162fa4	res_http_websocket: respond to CLOSE opcode ... This ensures that Asterisk responds properly to frames received from a client with opcode 8 (CLOSE) by echoing back the status code in its own CLOSE frame. Handling of the CLOSE opcode is moved up with the rest of the opcodes so that unmasking gets applied. The payload is no longer returned to the caller, but neither ARI nor the chan_sip nor pjsip made use of the payload, which is a good thing since it was masked. ASTERISK-28231 #close Change-Id: Icb1b60205fc77ee970ddc91d1f545671781344cf	7 years ago
Sean Bright	9cd6161083	pjsip_transport_management: Shutdown transport immediately on disconnect ... The transport management code that checks for idle connections keeps a reference to PJSIP's transport for IDLE_TIMEOUT milliseconds (32000 by default). Because of this, if the transport is closed before this timeout, the idle checking code will keep the transport from actually being shutdown until the timeout expires. Rather than passing the AO2 object to the scheduler task, we just pass its key and look it up when it is time to potentially close the idle connection. The other transport management code handles cleaning up everything else for us. Additionally, because we use the address of the transport when generating its name, we concatenate an incrementing ID to the end of the name to guarantee uniqueness. Related to ASTERISK~28231 Change-Id: I02ee9f4073b6abca9169d30c47aa69b5e8ae9afb	7 years ago
Joshua C. Colp	42922af7b8	stasis / manager / ari: Better filter messages. ... Previously both AMI and ARI used a default route on their stasis message router to handle some of the messages for publishing out their respective connection. This caused messages to be given to their subscription that could not be formatted into AMI or JSON. This change adds an API call to the stasis message router which allows a default route to be set as well as formatters that the default route is expecting. This allows both AMI and ARI to specify that their default route only wants messages of their given formatter. By doing so stasis can more intelligently filter at publishing time so that they do not receive messages which will not be turned into AMI or JSON. ASTERISK-28244 Change-Id: I65272819a53ce99f869181d1d370da559a7d1703	7 years ago
Sean Bright	d87fc1c591	res_pjsip_transport_websocket: Don't assert on 0 length payloads ... If an external pjproject is used and built with assertions enabled, we will assert if passed a payload length of 0, so treat empty frames as if we didn't receive them. Change-Id: I9c5fdccd89cc8d2f3ed7e3ee405ef0fc78178f48	7 years ago
Friendly Automation	5254ebf09f	Merge "res_pjsip: add option to disable ContactStatus event when contact is updated" into 13	7 years ago
Alexei Gradinari	4a8564cafa	res_pjsip: add option to disable ContactStatus event when contact is updated ... This patch adds a new PJSIP global configuration option 'send_contact_status_on_update_registration' to be able to have the same performance benefits as version 16. By default old behavior, i.e. the ContactStatus event will be sent when a device refreshes its registration. Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46	7 years ago
Alexei Gradinari	f9face3612	RTP: reset DTMF last seqno/timestamp on RTP renegotiation ... The remote side may start a new stream when renegotiating RTP. Need to reset the DTMF last sequence number and the timestamp of the last END packet on RTP renegotiation. If the new time stamp is lower then the timestamp of the last DTMF END packet the asterisk drops all DTMF frames as out of order. This bug was caught using Cisco ip-phone SPA5XX and codec g722. On SIP session update the SPA50X resets stream and a new timestamp is twice smaller then the previous. ASTERISK-28162 #close Change-Id: Ic72b4497e74d801b27a635559c1cf29c16c95254	7 years ago
George Joseph	1036959a49	Merge "res_rtp_asterisk: Remove some unused structure fields." into 13	7 years ago
Sean Bright	a2336adfb9	res_rtp_asterisk: Remove some unused structure fields. ... All of the fields that were removed were no longer referenced except for 'lastrxts' and 'rxseqno' which were only ever written to. Change-Id: I5a5d31eb33e97663843698f58d0d97f22a76627c	7 years ago
Sean Bright	560747dca4	res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set ... The profile-iop octet (the 2nd) of profile-level-id can be zero according to RFC 6184 Section 8.1. So we ignore its value when deciding to include profile-level-id in the outgoing SDP. ASTERISK-27959 #close Reported by: David Kuehling Change-Id: Id28cd916a3d7748058fe9609b585d07d9e243f73	7 years ago
Friendly Automation	212f2a3aa3	Merge "Use non-blocking socket() and pipe() wrappers" into 13	7 years ago
Friendly Automation	53bda83f5e	Merge "Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit"" into 13	7 years ago
Sean Bright	b84fd37ef0	Use non-blocking socket() and pipe() wrappers ... Change-Id: I050ceffe5a133d5add2dab46687209813d58f597	7 years ago
George Joseph	1ef993911f	Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit" ... This reverts commit cf620ce0f6. Pending resolution of ASTERISK_28200 Change-Id: If18ab1166db0d634ce8a099bd7460251e9c682e1	7 years ago
Corey Farrell	ec02d7fd4c	res_pjsip: Patch for res_pjsip_* module load/reload crash ... The session_supplements for the pjsip makes crashes when the module load/unload. ASTERISK-28157 Change-Id: I5b82be3a75d702cf1933d8d1417f44aa10ad1029	7 years ago
Jenkins2	9f0459d2a0	Merge "RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit" into 13	7 years ago
Corey Farrell	0a9904e1c6	astobj2: Eliminate usage of legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. ao2_container_alloc is now restricted to modules only and is being removed from Asterisk 17. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Joshua Colp	b9692fb909	Merge "res_pjsip_caller_id: Use static pj_str_t for fromto header names." into 13	7 years ago
Joshua Colp	d748ed4147	stasis: Add internal filtering of messages. ... This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190	7 years ago
George Joseph	ba068d3576	Merge "res/res_pjsip_nat: Fix logic for REINVITES" into 13	7 years ago
Corey Farrell	580bc5e2c6	res_pjsip_caller_id: Use static pj_str_t for fromto header names. ... PJSIP assumes that these header names are not allocated, does not clone the name strings when reusing headers. Block unload of res_pjsip_caller_id until shutdown to ensure static memory stays valid. It was previously unsafe to unload while any sessions are active. Change-Id: I190854dea943d6e441cf03733f8a0da661aea27f	7 years ago
Torrey Searle	da4879443b	res/res_pjsip_nat: Fix logic for REINVITES ... The presence of Record-Route in re-invites is optional, thus it is important to make sure the dialog doesn't have a routset before rewriting the contact header. ASTERISK-28129 #close Change-Id: Ic8ceb54ccfc93f7e315e476c514a2c777f2da7dc	7 years ago
Alexei Gradinari	cf620ce0f6	RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit ... The marker bit set on the voice packet indicates the start of a new stream and a new time stamp. Need to reset the DTMF last sequence number and the timestamp of the last END packet. If the new time stamp is lower then the timestamp of the last DTMF END packet the asterisk drops all DTMF frames as out of order. This bug was caught using Cisco ip-phone SPA50X and codec g722. On SIP session update the SPA50X resets stream indicating it with market bit and a new timestamp is twice smaller then the previous. ASTERISK-28162 #close Change-Id: If9c5742158fa836ad549713a9814d46a5d2b1620	7 years ago
Joshua Colp	6aea312a55	Merge "res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue" into 13	7 years ago
Chris-Savinovich	bc2420ee0e	res_pjsip: Send a 503 response when overload state if reliable transport. ... When Asterisk's taskprocessors get overloaded we need to reduce the work load. res_pjsip currently ignores new SIP requests and relies on SIP retransmissions in the hope that the overload condition will clear soon enough to handle the retransmitted SIP request. This change adds the following code after ast_taskprocessor_alert_get() has returned TRUE: 1- identifies transport type. If non-udp then send a 503 response 2- if transport type is udp/udp6 then ignore, as before. Change-Id: I1c230b40d43a254ea0f226b7acf9ee480a5d3836	7 years ago
Kevin Harwell	214d0d118a	res_pjsip: formatting error in documentation ... The use of a '|' in the "global/debug" synopsis documentation caused the generated html table on the wiki to add an extra column that included the text after the pipe. This patch replaces the pipe with a comma. ASTERISK-28150 Change-Id: I3d79a6ca6d733d9cb290e779438114884b98a719	7 years ago
Alexei Gradinari	158214c1a0	res_pjsip.c: Make taskprocessor scheduling algorithm pick the shortest queue ... The current round-robin method does not take the current taskprocessor load into consideration when distributing requests. Using the least-size method the request goes to the taskprocessor that is servicing the least number of active tasks at the current time. Longer running tasks with the round-robin method can delay processing tasks. * Change the algorithm from round-robin to least-size for picking the PJSIP taskprocessor from the default serializer pool. Change-Id: I7b8d8cc2c2490494f579374b6af0a4868e3a37cd	7 years ago
George Joseph	5605928db0	Merge "res_pjsip: Add XML documentation for "use_callerid_contact"" into 13	7 years ago
Joshua Colp	9946bcc557	res_pjsip: Add XML documentation for "use_callerid_contact" ... ASTERISK-28087 Change-Id: I69d48813ec514f5ef06c6de994cba52630e0a3b4	7 years ago
Alexei Gradinari	bfe3821800	pjsip: new endpoint's options to control Connected Line updates ... This patch adds new options 'trust_connected_line' and 'send_connected_line' to the endpoint. The option 'trust_connected_line' is to control if connected line updates are accepted from this endpoint. The option 'send_connected_line' is to control if connected line updates can be sent to this endpoint. The default value is 'yes' for both options. Change-Id: I16af967815efd904597ec2f033337e4333d097cd	7 years ago
George Joseph	0b695d1e38	Merge "res_pjsip_notify: improve realtime performance on CLI completion on the endpoint" into 13	7 years ago
Alexei Gradinari	82f0a86e39	res_pjsip_notify: improve realtime performance on CLI completion on the endpoint ... The module 'res_pjsip_notify' inefficiently makes a lot of DB requests on CLI completion on the endpoint. For example if there are 10k endpoints the module makes 10k requests of these 10k records. Even if a part of the endpoint entered the module makes the same 10k requests and then filtered them by itself. This patch gathers endpoints container by prefix and adds all gathered endpoints to completion at once. ASTERISK-28137 #close Change-Id: Ic20024912cc77bf4d3e476c4cd853293c52b254b	7 years ago
Torrey Searle	bbbec2e95e	res_pjsip_session: add new flag use_callerid_contact ... Add a new global flag to res_pjsip to allow the callerid to be used as the username in the contact header. This allows chan_pjsip to have the same behavour as chan_sip ASTERISK-28087 #close Change-Id: I9a720e058323f6862a91c62f8a8c1a4b5c087b95	7 years ago
George Joseph	1c33cf5cf1	Merge "astobj2: Eliminate usage of legacy container allocation macros." into 13	7 years ago
Corey Farrell	54a1fbe428	astobj2: Eliminate usage of legacy container allocation macros. ... These macros have been documented as legacy for a long time but are still used in new code because they exist. Remove all references to: * ao2_container_alloc_options * ao2_t_container_alloc_options * ao2_t_container_alloc These macro's are still available for use but only in modules. Only ao2_container_alloc remains due to it's use in over 100 places. Change-Id: I1a26258b5bf3deb081aaeed11a0baa175c933c7a	7 years ago
Richard Mudgett	d8b9e3413c	Fix 'statement' typo throughout code. ... Most were in comments. A couple were in warning messages. Pointed out by Jonathan H on the Asterisk users mailing list. Change-Id: I6286939dff5d0a27a2758140570106f1cb351855	7 years ago
George Joseph	6296b5c7be	Merge "res_pjsip: set callerid_tag to empty string" into 13	7 years ago
Alexei Gradinari	6b1a981ee1	res_pjsip: set callerid_tag to empty string ... This patch sets the callerid_tag to empty string by default. If the callerid_tag is set to NULL then the tag does not become part of a connected line update. For example: Alice's tag is "Alice". Bob's tag is empty. Charlie's tag is "Charlie". Alice calls Bob and then does attended transfer to Charlie. When Alice hangs up the CONNECTEDLINE(tag) is "Alice" on the interception routine on the Charlie's channel, but should be empty. Ths patch also fix memory leaks if there are more then one options "callerid", "callerid_tag", "voicemail_extension" and "contact_user" in the pjsip.conf endpoint definition. Change-Id: I86ba455c4677ca8d516d9a04ce7fb4d24dd576e4	7 years ago