4784 Commits (bc8dead6104f3faf59ab8575cfa2553702109836)

Author	SHA1	Message	Date
Kevin Harwell	8681fc9db7	ARI event type filtering ... Event type filtering is now enabled, and configurable per application. An app is now able to specify which events are sent to the application by configuring an allowed and/or disallowed list(s). This can be done by issuing the following: PUT /applications/{applicationName}/eventFilter And then enumerating the allowed/disallowed event types as a body parameter. ASTERISK-28106 Change-Id: I9671ba1fcdb3b6c830b553d4c5365aed5d588d5b	8 years ago
Torrey Searle	8ea9608efb	res/res_rtp_asterisk: clear smoother when local bridging ... p2p_write updates txformat but doesn't require a smoother. If a smoother was created by another bridge type the smoother could fall out of date causing one way audio issues. To prevent this the smoother is now destroyed on the start of native bridge. ASTERISK-28284 #close Change-Id: I84e67f144963787fff9b4d79ac500514fb40cdc6	8 years ago
Friendly Automation	d4a78063eb	Merge "res_odbc: Add basic query logging."	8 years ago
Kevin Harwell	61a8f79a29	res_pjsip_registrar: lock transport monitor when setting 'removing' flag ... A previous patch attempt to mitigate blocked threads on transport shutdown for a given contact. It was thought that a second lock could be avoided by checking the 'removing' flag on the transport monitor twice (once before and once after the normal named aor locking). However as with usual threading issues if the timing was right the original problem still occured. This patch adds locking around the first 'removing' flag check and set, thus nullifying the secondary check, so it was removed. ASTERISK-28213 Change-Id: Iaa8e36e5311789549b76d8de42dfcea96013b2ed	8 years ago
Joshua Colp	54a912b26d	res_odbc: Add basic query logging. ... When Asterisk is connected and used with a database the response time of the database can cause problems in Asterisk if it is long. Normally the only way to see this problem would be to retrieve a backtrace from Asterisk and examine where things are blocked, or examine the database to see if there is any indication of a problem. This change adds some basic query logging to make it easier to investigate such a problem. When logging is enabled res_odbc will now keep track of the number of queries executed, as well as the query that has taken the longest time to execute. There is also an option which will cause a WARNING message to be output if a query takes longer than a configurable amount of time to execute. This makes it easier and clearer for users that their database may be experiencing a problem that could impact Asterisk. ASTERISK-28277 Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6	8 years ago
Friendly Automation	99186c4b15	Merge "pjsip/config_global: regcontext context not created"	8 years ago
George Joseph	00d7b6df49	Merge "res_stasis: Auto-create context and extens on Stasis app launch."	8 years ago
George Joseph	4787b47da0	Merge "Added ARI resource /ari/asterisk/ping"	8 years ago
Ben Ford	3f9c5fba95	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	8 years ago
sungtae kim	ac90968afd	Added ARI resource /ari/asterisk/ping ... Added ARI resource. GET /ari/asterisk/ping : It returns "pong" message with timestamp and asterisk id. It would be useful for simple heath check. Change-Id: I8d24e1dcc96f60f73437c68d9463ed746f688b29	8 years ago
Kevin Harwell	f668db9ba0	pjsip/config_global: regcontext context not created ... The context specified by 'regcontext' was not being created, so when Asterisk attempted to later dynamically add an extension it would fail. This patch now creates the context if a 'regcontext' is specified. ASTERISK-28238 Change-Id: I0f36cf4ab0a93ff4b1cc5548d617ecfd45e09265	8 years ago
George Joseph	7071e9d64c	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. "sounds" is no longer a valid target for the "module reload" command. 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	8 years ago
George Joseph	423a51ca11	Merge "res_http_websocket: ensure control frames do not interfere with data"	8 years ago
Joshua C. Colp	5e3d7ffdca	Merge "res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown"	8 years ago
Jeremy Lainé	69e9fd63e1	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	8 years ago
Joshua C. Colp	884aaa5f72	Merge "stasis / manager / ari: Better filter messages."	8 years ago
Joshua C. Colp	3655d304af	Merge "bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix"	8 years ago
Joshua C. Colp	4e41be0a59	Merge "pjsip_transport_management: Shutdown transport immediately on disconnect"	8 years ago
Kevin Harwell	b82d2856b4	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	8 years ago
Xiemin Chen	a526676836	bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix ... To avoid the stream name collide if there're more than one video track in one client. If client has multi video tracks, the name of ast_stream which represents each video track may be the same. Use the MSID:LABEL here because it's identifiable. ASTERISK-28196 #close Reported-by: xiemchen Change-Id: Ib62b2886e8d3a30e481d94616b0ceaeab68a870b	8 years ago
Jeremy Lainé	0b8867f7d6	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	8 years ago
Sean Bright	20f672539e	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	8 years ago
Joshua C. Colp	1323730f6c	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	8 years ago
Sean Bright	2b8602e8cf	res_pjsip_transport_websocket: Don't assert on 0 length payloads ... When --enable-dev-mode is used, pjsip_tpmgr_receive_packet() will assert if passed a payload length of 0, so treat empty frames as if we didn't receive them. Change-Id: I9c5fdccd89cc8d2f3ed7e3ee405ef0fc78178f48	8 years ago
Joshua C. Colp	b19210214f	Merge "res_pjsip: add option to enable ContactStatus event when contact is updated"	8 years ago
Joshua C. Colp	a3f1f82272	Merge "res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled."	8 years ago
Alexei Gradinari	f0546d1d87	res_pjsip: add option to enable ContactStatus event when contact is updated ... The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out the ContactStatus AMI event when a contact is updated. Thist change broke things which rely on old behavior. This patch adds a new PJSIP global configuration option 'send_contact_status_on_update_registration' to be able to preserve old ContactStatus behavior. By default new behavior, i.e. the ContactStatus event will not be sent when a device refreshes its registration. Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46	8 years ago
Joshua Colp	18e206381a	res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled. ... For video streams it was possible for the abs-send-time information to be placed into RTP streams even if not negotiated. Depending on the endpoint in use this could cause video to not flow. We now only enable abs-send-time for negotiation if WebRTC is enabled. ASTERISK-28230 Change-Id: I0eb682302f8da3a4ea3c42e839208d55f825ed0c	8 years ago
Alexei Gradinari	f662a26ea0	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	8 years ago
Friendly Automation	31bacc2354	Merge "res/res_ari: Add additional hangup reasons"	8 years ago
George Joseph	3464093f85	Merge "res_pjsip: Patch for res_pjsip_* module load/reload crash"	8 years ago
George Joseph	2e21910ca1	Merge "res_rtp_asterisk: Remove some unused structure fields."	8 years ago
Sean Bright	357219dfb3	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	8 years ago
Sean Bright	5b12dfa6dd	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	8 years ago
Joshua C. Colp	a28f0382e8	Merge "Use non-blocking socket() and pipe() wrappers"	8 years ago
George Joseph	5d4d723844	Merge "Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit""	8 years ago
Sean Bright	42ff856216	Use non-blocking socket() and pipe() wrappers ... Change-Id: I050ceffe5a133d5add2dab46687209813d58f597	8 years ago
George Joseph	d1598dbc7d	Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit" ... This reverts commit 3f53041267. Pending resolution of ASTERISK_28200 Change-Id: Iad4f3614cac95b00fdbb2b799aab8ae6285ec988	8 years ago
Sebastian Damm	a24bb1c4b6	res/res_ari: Add additional hangup reasons ... The ARI DELETE /channels command takes a "reason" parameter Previously, there were only five reasons implemented This patch adds more reasons to choose from for more complex setups ASTERISK-28198 #close Change-Id: I85996f1076c9946d65c778413f040a845a90fecc	8 years ago
Sungtae Kim	8644511cbf	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	8 years ago
George Joseph	3667c5e1d2	bridges: Remove reliance on stasis caching ... * The bridging core no longer uses the stasis cache for bridge snapshots. The latest bridge snapshot is now stored on the ast_bridge structure itself. * The following APIs are no longer available since the stasis cache is no longer used: ast_bridge_topic_cached() ast_bridge_topic_all_cached() * A topic pool is now used for individual bridge topics. * The ast_bridge_cache() function was removed since there's no longer a separate container of snapshots. * A new function "ast_bridges()" was created to retrieve the container of all bridges. Users formerly calling ast_bridge_cache() can use the new function to iterate over bridges and retrieve the latest snapshot directly from the bridge. * The ast_bridge_snapshot_get_latest() function was renamed to ast_bridge_get_snapshot_by_uniqueid(). * A new function "ast_bridge_get_snapshot()" was created to retrieve the bridge snapshot directly from the bridge structure. * The ast_bridge_topic_all() function now returns a normal topic not a cached one so you can't use stasis cache functions on it either. * The ast_bridge_snapshot_type() stasis message now has the ast_bridge_snapshot_update structure as it's data. It contains the last snapshot and the new one. * cdr, cel, manager and ari have been updated to use the new arrangement. Change-Id: I7049b80efa88676ce5c4666f818fa18ad1985369	8 years ago
Jenkins2	4ca709768d	Merge "stasis: Segment channel snapshot to reduce creation cost."	8 years ago
Joshua Colp	b80c9071e3	Merge "RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit"	8 years ago
Joshua Colp	50ac85cb40	stasis: Segment channel snapshot to reduce creation cost. ... When a channel snapshot was created it used to be done from scratch, copying all data (many strings). This incurs a cost when doing so. This change segments the channel snapshot into different components which can be reused if unchanged from the previous snapshot creation, reducing the cost. In normal cases this results in some pointers being copied with reference count being bumped, some integers being set, and a string or two copied. The other benefit is that it is now possible to determine if a channel snapshot update is redundant and thus stop it before a message is published to stasis. The specific segments in the channel snapshot were split up based on whether they are changed together, how often they are changed, and their general grouping. In practice only 1 (or 0) of the segments actually get changed in normal operation. Invalidation is done by setting a flag on the channel when the segment source is changed, forcing creation of a new segment when the channel snapshot is created. ASTERISK-28119 Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423	8 years ago
Joshua Colp	d0ccbb3377	stasis: Use an implementation specific channel snapshot cache. ... Channels no longer use the Stasis cache for channel snapshots. Instead they are stored in a hash table in stasis_channels which reduces the number of Stasis messages created and allows better storage. As a result the following APIs are no longer available since the stasis cache is no longer used: ast_channel_topic_cached() ast_channel_topic_all_cached() The ast_channel_cache_all() and ast_channel_cache_by_name() functions now return an ao2_container of ast_channel_snapshots rather than a container of stasis_messages therefore you can't (and don't need to) call stasis_cache functions on it. The ast_channel_topic_all() function now returns a normal topic not a cached one so you can't use stasis cache functions on it either. The ast_channel_snapshot_type() stasis message now has the ast_channel_snapshot_update structure as it's data. It contains the last snapshot and the new one. ast_channel_snapshot_get_latest() still returns the latest snapshot. The latest snapshot is now stored on the channel itself to eliminate cache hits when Stasis messages that have the snapshot as a payload are created. ASTERISK-28102 Change-Id: I9334febff60a82d7c39703e49059fa3a68825786	8 years ago
Alexei Gradinari	3f53041267	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	8 years ago
Corey Farrell	021ce938ca	astobj2: Remove legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. Remove ao2_container_alloc macro. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	8 years ago
Joshua Colp	9f996ca1f0	Merge "res/res_ari: Fix null endpoint handle"	8 years ago
Joshua Colp	b285333673	Merge "res_pjsip_caller_id: Use static pj_str_t for fromto header names."	8 years ago
Joshua Colp	3077ad0c24	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	8 years ago