asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	3cd36d0e10	res_pjsip_transport_websocket: Fix crash when the Contact header is not a URI. The code for changing the Contact header wrongly assumed that the Contact would always contain a URI. This is incorrect. ASTERISK-24271 Reported by: Dafi Ni ........ Merged revisions 422557 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422558 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	1b64f353f1	Resolve race condition where channels enter dialplan application before media has been negotiated. Testsuite tests will occasionally fail because on reception of a 200 OK SIP response, an AST_CONTROL_ANSWER frame is queued prior to when media has finished being negotiated. This is because session supplements are called into before PJSIP's inv_session code has told us that media has been updated. Sometimes the queued answer frame is handled by the PBX thread before the ensuing media negotiations occur, causing a test failure. As it turns out, there is another place that session supplements could be called into, which is after media has finished getting negotiated. What this commit introduces is a means for session supplements to indicate when they wish to be called into when handling an incoming SIP response. By default, all session supplements will be run at the same point that they were prior to this commit. However, session supplements may indicate that they wish to be handled earlier than normal on redirects, or they may indicate they wish to be handled after media has been negotiated. In this changeset, two session supplements have been updated to indicate a preference for when they should be run: res_pjsip_diversion executes before handling redirection in order to get information from the Diversion header, and chan_pjsip now handles responses to INVITEs after media negotiation to fix the race condition mentioned previously. ASTERISK-24212 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/3930 ........ Merged revisions 422536 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422542 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	df5dbbd878	res_stasis: Don't play MoH to channels by default when added to holding bridges When ARI manipulates a bridge, it generally doesn't care what the mixing technology is. Operations on a bridge initiated through ARI should perform their action in generally the same way, regardless of the bridge's mixing technology. While the mixing technology may determine how media flows to channels, the actual operations on a bridge themselves should be the same. Currently, this isn't the case with holding bridges. When a channel joins without a role, MoH is started on that channel automatically. Subsequent bridge operations that would stop MoH would fail (as there is no Announcer channel playing MoH to the bridge). Starting MoH on the bridge will also create two MoH streams: one from the MoH being played on the participant channel, and one from the announcer channel. From the perspective of ARI users, this is counter-intuitive - I would not expect MoH to be started for me. The mixing technology determines how media is shared between participants, not the application experience. This patch does the following: * The Stasis bridge class now inspects channels as they are going into a bridge. If the bridge has a holding capability, and the channel has no roles, we give it a participant role and mark the default behaviour to have no entertainment. This allows addChannel operations to continue to set a participant role with an entertainment option if it felt like it (or could do it). * The music on hold channel is now Stasis approved (tm) Review: https://reviewboard.asterisk.org/r/3929/ ASTERISK-24264 #close Reported by: Samuel Galarneau Tested by: Samuel Galarneau ........ Merged revisions 422503 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422504 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	327d67270f	Fix bug that did not allow for multiple batched RLS notifications to be sent. A misunderstanding of how the scheduler worked caused further batched notifications beyond the first not to get scheduled. Now we reset our scheduler ID to -1 after the batched notification is sent. This way, further notifications can be scheduled when they arise. ........ Merged revisions 422239 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	94e1b4a8a4	res/res_pjsip/pjsip_options.c: Eliminate excessive RAII_VAR usage. * Fix off nominal ref leak in find_or_create_contact_status(). * Add missing NULL check of status in update_contact_status() and init_start_time(). ........ Merged revisions 422214 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422215 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	bf85018107	CallerID: Fix parsing of malformed callerid This allows the callerid parsing function to handle malformed input strings and strings containing escaped and unescaped double quotes. This also adds a unittest to cover many of the cases where the parsing algorithm previously failed. Review: https://reviewboard.asterisk.org/r/3923/ Review: https://reviewboard.asterisk.org/r/3933/ ........ Merged revisions 422112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422113 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422114 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422154 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	fefa6fba82	res_musiconhold.c: Release any format refs before memset(). * Clear the channel music_state pointer before destroying the music_state object for safety. ........ Merged revisions 422037 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2b19d94a71	res_musiconhold: Fix MOH restarting where it left off from the last hold. Restore code removed by https://reviewboard.asterisk.org/r/3536/ that introduced a regression that prevents MOH from restarting were it left off the last time. ASTERISK-24019 #close Reported by: Jason Richards Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3928/ ........ Merged revisions 421976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421977 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421978 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421979 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	497a92d079	res_pjsip_transport_websocket: Attach the Websocket module on outgoing INVITEs. In order to alter the Contact header on in-dialog requests and responses the Websocket module must be attached on outgoing INVITEs. The Contact header is modified so that the PJSIP transport layer can find and use the existing Websocket connection based on the source IP address, port, and transport. ASTERISK-24143 #close Reported by: Aleksei Kulakov ........ Merged revisions 421955 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421956 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	477e2e6edb	res_pjsip_transport_websocket: Fix a progressive memory growth. The packet structure used to receive messages was using the transport pool. This meant that for each parsing the pool would grow accordingly. Since memory can not be reclaimed without resetting it this would cause the memory pool to grow and grow. This change uses a specific memory pool for the packet structure and resets it to a fresh state after the message has been received and handled. ........ Merged revisions 421939 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421945 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	2c0cbf8e64	res_pjsip_transport_websocket: Ensure secure Websocket clients can be called. This change enforces the transport in the Contact header for Websocket clients. Previously a client may provide a transport of 'ws' when it is actually using a transport of 'wss'. This would cause outgoing calls to fail as the existing connection could not be found. ........ Merged revisions 421931 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421932 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	33835e17a0	ARI: Fix a crash caused by hanging during playback to a channel in a bridge ASTERISK-24147 #close Reported by: Edvin Vidmar Review: https://reviewboard.asterisk.org/r/3908/ ........ Merged revisions 421879 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421880 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	f8c4fc1121	res_musiconhold.c: Remove obsolete REF_DEBUG code. Remove unneeded code that writes to the wrong file location in an obsolete format. ........ Merged revisions 421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421800 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421801 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421802 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	644e693645	Switch from hostname to an IP address in the SDP origin line. Using the hostname in the SDP origin line may not satisfy the requirement of RFC 4566 that we use a FQDN or IP address. This change has us use the same information from the SDP connection line if possible. If not possible, we'll use the configured media address. And if that's not possible, we use the result of a PJLIB call to get the IP address of ourself. ASTERISK-23994 #close Reported by Private Name Review: https://reviewboard.asterisk.org/r/3925 ........ Merged revisions 421796 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421797 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	56a1d4930a	Ensure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge. Because of the departable state of channels that enter Stasis bridges, Stasis has to take responsibility for directing the channel to its intended after-bridge destination if the channel moves from a Stasis bridge to a non-Stasis bridge. This change ensures that when such a move occurs, when the channel leaves the bridging system, any after bridge gotos are honored. Review: https://reviewboard.asterisk.org/r/3920 ........ Merged revisions 421792 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421794 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	4946981646	res_musiconhold: Fix reference leaks caused when reloading with REF_DEBUG set Due to a faulty function for debugging reference decrementing, it was possible to reduce the refcount on the wrong object if two moh classes of the same name were in the moh class container. (closes issue ASTERISK-22252) Reported by: Walter Doekes Patches: 18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license 6182) ........ Merged revisions 398937 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421777 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421779 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421788 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	12d34bb12f	Let's try checking the name and number, instead of the name twice. ........ Merged revisions 421789 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421790 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	2150daf748	Improve consistency of party ID privacy usage. Prior to this change, the Remote-Party-ID header took the position of "If caller name and number are not explicitly allowed, then they are private" and P-Asserted-Identity took the position of "Caller name and number are only private if marked explicitly so" Now both mechanisms of conveying party identification use the former approach. ........ Merged revisions 421778 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421783 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	085d5a2629	Clean up files that do not end with newlines Trivial patch to add new lines to several files missing them. This fixes warnings when compiling with gcc 4.1.2 on CentOS 5. ASTERISK-24245 #close Reported by: Shaun Ruffell patches: 0002-Trivial-addition-of-newlines-at-end-of-three-files.patch uploaded by Shaun Ruffell (License 5417) ........ Merged revisions 421677 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421678 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b7f98c3da4	chan_pjsip: Update media translation paths when new SDP negotiated. On a SIP reinvite that changes media strams, the PJSIP channel driver was flooding the log with "Asked to transmit frame type %s, while native formats is %s" warnings. * Fixes PJSIP not setting up translation paths when the formats change on a reinvite. AFS-63 was effectively reintroduced because of the media formats work. res_pjsip_sdp_rtp.c:set_caps() * Improved the unexpected frame format WARNING message to include more information. * Added protective locking while altering formats on a channel. Reworked set_format() to simplify and protect the formats under manipulation. * Restored some code that got lost in the media_formats work. (channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps()) AFS-137 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3906/ ........ Merged revisions 421645 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	49f8bd4ad4	Set the role for inbound subscriptions correctly. This was causing the AMI show_subscriptions test in the testsuite to fail since all subscriptions were being seen as subscribers instead of notifiers. ........ Merged revisions 421585 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	36f4bff943	Stasis: Add information to blind transfer event When a blind transfer occurs that is forced to create a local channel pair to satisfy the transfer request, information about the local channel pair is not published. This adds a field to describe that channel to the blind transfer message struct so that this information is conveyed properly to consumers of the blind transfer message. This also fixes a bug in which Stasis() was unable to properly identify the channel that was replacing an existing Stasis-controlled channel due to a blind transfer. Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3921/ ........ Merged revisions 421537 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421538 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	76290adf50	Alter documentation for callerid_privacy to use correct values. ........ Merged revisions 421485 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421488 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	28a89e7685	Fix compilation error on certain versions of GCC. ........ Merged revisions 421447 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421448 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421449 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	222b5cd036	ARI: Fix a bug where /channels/{channelID}/continue doesn't execute PBX If /channels/{channelID}/continue is called on a channel that was originated without a PBX (such as the ARI command POST channel with a stasis application argument), the channel will not start dialplan execution. This patch will now run the PBX out of the stasis execution if the channel doesn't currently have an active PBX upon continuing. ASTERISK-24043 #close Reported by: Krandon Bruse Review: https://reviewboard.asterisk.org/r/3917/ Patches: stasis-continue.diff submitted by Krandon Bruse (license 6631) ........ Merged revisions 421416 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421423 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	83a9b91da9	chan_pjsip: Fix attended transfer connected line name update. A calls B B answers B SIP attended transfers to C C answers, B and C can see each other's connected line information B completes the transfer A has number but no name connected line information about C while C has the full information about A I examined the incoming and outgoing party id information handling of chan_pjsip and found several issues: * Fixed ast_sip_session_create_outgoing() not setting up the configured endpoint id as the new channel's caller id. This is why party A got default connected line information. * Made update_initial_connected_line() use the channel's CALLERID(id) information. The core, app_dial, or predial routine may have filled in or changed the endpoint caller id information. * Fixed chan_pjsip_new() not setting the full party id information available on the caller id and ANI party id. This includes the configured callerid_tag string and other party id fields. * Fixed accessing channel party id information without the channel lock held. * Fixed using the effective connected line id without doing a deep copy outside of holding the channel lock. Shallow copy string pointers can become stale if the channel lock is not held. * Made queue_connected_line_update() also update the channel's CALLERID(id) information. Moving the channel to another bridge would need the information there for the new bridge peer. * Fixed off nominal memory leak in update_incoming_connected_line(). * Added pjsip.conf callerid_tag string to party id information from enabled trust_inbound endpoint in caller_id_incoming_request(). AFS-98 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3913/ ........ Merged revisions 421400 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	bb494067a5	Multiple revisions 421311-421312 ........ r421311 \| mjordan \| 2014-08-17 20:11:28 -0500 (Sun, 17 Aug 2014) \| 9 lines res/ari/resource_channels: Don't return allocation failure on failed function If a function fails to execute, it is most likely due to one of two reasons: (1) The function doesn't exist or can't be read from (2) The function is dangerous and is restricted based on the user's permissions Currently we return allocation failure, which is incorrect. This updates the reason code to more accurately reflect why the request failed. ASTERISK-24215 ........ r421312 \| mjordan \| 2014-08-17 20:13:41 -0500 (Sun, 17 Aug 2014) \| 4 lines res/ari/resource_channels: Fix compilation issue Forgot a parameter. Whoops. ........ Merged revisions 421311-421312 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421313 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ba5d5da60b	Improve call forwarding reporting, especially with regards to ARI. This patch addresses a few issues: 1) The order of Dial events have been changed when performing a call forward. The order has now been altered to 1) Dial begins dialing channel A. 2) When A forwards the call to B, we issue the dial end event to channel A, indicating the dial is being canceled due to a forward to B. 3) When the call to channel B occurs, we then issue a new dial begin to channel B. 2) Call forwards are now reported on the calling channel, not the peer channel. 3) AMI DialEnd events have been altered to display the extension the call is being forwarded to when relevant. 4) You can now get the values of channel variables for channels that are not currently in the Stasis application. This brings the retrieval of channel variables more in line with the rest of channel read operations since they may be performed on channels not in Stasis. ASTERISK-24134 #close Reported by Matt Jordan ASTERISK-24138 #close Reported by Matt Jordan Patches: forward-shenanigans.diff uploaded by Matt Jordan (License #6283) Review: https://reviewboard.asterisk.org/r/3899 ........ Merged revisions 420794 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	952da298ce	res_http_websocket: Include query parameters in client connection requests. Review: https://reviewboard.asterisk.org/r/3914/ ........ Merged revisions 421210 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	980e49614c	res/res_hep_rtcp: Remove dependency on PJSIP The res_hep_rtcp module was incorrectly including <pjsip.h>. This didn't need to be included, as the module does not using PJPROJECT any fashion. Unfortunately, because res_hep_rtcp did not include pjsip in its MODULEINFO as a dependency, this also meant that res_hep_rtcp will fail to compile on a system without PJPROJECT. This patch removes the include. Thanks to Damien Wedhorn for pointing this out in #asterisk-dev. ASTERISK-24236 #close Reported by: Damien Wedhorn, Matt Jordan Tested by: Damien Wedhorn ........ Merged revisions 421064 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421065 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	79c5c08db9	ARI: Originate to app local channel subscription code optimization. Reduce the scope of local_peer and only get it if the ARI originate is subscribing to the channels. Review: https://reviewboard.asterisk.org/r/3905/ ........ Merged revisions 421009 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421010 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421012 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	6aa510b41f	PJSIP: Prevent crash no-URI contacts This prevents a crash from occurring when a contact with no URI is used for the creation of an outbound out-of-dialog request with no associated endpoint. ........ Merged revisions 420949 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 420950 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420953 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	8526d967c9	res_stasis_snoop.c: Fix off nominial exit path leaving Snoop channel locked and not hungup. * Made use ast_copy_string() instead of strcpy() for snoop uniqueid for safety. There is no guarantee that the max channel uniqueid length will remain the same as the snoop uniqueid space. ........ Merged revisions 420879 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	aba07a0f6e	res/stasis/command.c: Fix recent commit using spaces instead of tabs. ........ Merged revisions 420836 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 420837 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420838 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	7a4691b425	Stasis: Use the correct return value Return the correct value instead of always returning 0 when setting internal status on unreal channels. Reported by: Richard Mudgett ........ Merged revisions 420802 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 420803 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	6f735288b0	Stasis: Allow internal channels directly into bridges The patch to catch channels being shoehorned into Stasis() via external mechanisms also happens to catch Announcer and Recorder channels because they aren't known to be stasis-controlled channels in the usual sense. This marks those channels as Stasis()-internal channels and allows them directly into bridges. Review: https://reviewboard.asterisk.org/r/3903/ ........ Merged revisions 420795 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 420796 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	db0a97f8ce	Fix crashing unit tests with regards to RLS. The unit tests require a sorcery.conf file that has been set up to store resource lists in memory rather than retrieving from configuration. With a setup that is not conducive to running the tests, a fault in sorcery currently causes Asterisk to crash when attempting to run any of the tests. To get around the crash, this adds a function that verifies the current environment and marks the tests as "not run" if the setup is not correct. ........ Merged revisions 420779 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420780 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	b4e33c81e3	Fix crash encountered by the testsuite. Running testsuite tests locally produced no errors, but when run using the continuous integration framework, crashes occurred. The crashes occurred due to a refcounting error that had been fixed for a similar situation. ........ Merged revisions 420758 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	becf7c7003	res_hep: Remove disabling of modules These modules were originally specified as being disabled, as they were introduced midstream in Asterisk 12. That makes it nicer for folks who are upgrading to a new release in the middle of Asterisk 12. That's not the case for Asterisk 13: it's a brand new release. There's no reason to have the modules disabled by default in that case. ........ Merged revisions 420742 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420743 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	965ba7c36f	Fix build in dev mode git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	99d0bccd35	Add support for RFC 4662 resource list subscriptions. This commit adds the ability for a user to configure a resource list in pjsip.conf. Subscribing to this list simultaneously subscribes the subscriber to all resources listed. This has the potential to reduce the amount of SIP traffic when loads of subscribers on a system attempt to subscribe to each others' states. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	0ac7f96057	Stasis: Convey transfer information to applications This fixes a class of issues where Stasis applications were not made aware that their channels were being manipulated or replaced by external entitiessuch as transfers, AMI commands, or dialplan applications such as Bridge(). Inconsistent information such as StasisEnd events with unknown channels as a result of masquerades has also been corrected. To accomplish these fixes, several new fields were added to blind and attended transfer messages as well as StasisStart and BridgeAttendedTransfer Stasis events. ASTERISK-23941 #close Review: https://reviewboard.asterisk.org/r/3865/ Review: https://reviewboard.asterisk.org/r/3857/ Review: https://reviewboard.asterisk.org/r/3852/ Review: https://reviewboard.asterisk.org/r/3816/ Review: https://reviewboard.asterisk.org/r/3731/ Review: https://reviewboard.asterisk.org/r/3729/ Review: https://reviewboard.asterisk.org/r/3728/ ........ Merged revisions 420325 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	a8829490b6	res_pjsip_publish_asterisk: Add support for exchanging device and mailbox state using SIP. This module uses the inbound and outbound PUBLISH support to exchange device and mailbox state between Asterisk instances. Each instance is configured to publish to the other and requires no intermediary server. The functionality provided is similar to the XMPP and Corosync support. Review: https://reviewboard.asterisk.org/r/3780/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ece78c6991	res_pjsip_outbound_publish: Add module which provides outbound PUBLISH support. This module implements the core parts required for doing outbound PUBLISH. It takes care of configuration, lifetime management, and authentication. Additional modules implement the specific events that are published. Review: https://reviewboard.asterisk.org/r/3780/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	f1036f40dc	Stasis: Allow message types to be blocked This introduces stasis.conf and a mechanism to prevent certain message types from being published. Internally, this works by preventing the chosen message types from being created which ensures that those message types can never be published. This patch also adjusts message publishers such that message payloads are not created if the related message type is not available. ASTERISK-23943 #close Review: https://reviewboard.asterisk.org/r/3823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ac5c75b45d	stasis: Fix compilation issue with ao2 tagged objects ........ Merged revisions 420099 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420100 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	47bf7efc4d	Multiple revisions 420089-420090,420097 ........ r420089 \| mjordan \| 2014-08-05 15:10:52 -0500 (Tue, 05 Aug 2014) \| 72 lines ARI: Add channel technology agnostic out of call text messaging This patch adds the ability to send and receive text messages from various technology stacks in Asterisk through ARI. This includes chan_sip (sip), res_pjsip_messaging (pjsip), and res_xmpp (xmpp). Messages are sent using the endpoints resource, and can be sent directly through that resource, or to a particular endpoint. For example, the following would send the message "Hello there" to PJSIP endpoint alice with a display URI of sip:asterisk@mycooldomain.org: ari/endpoints/sendMessage?to=pjsip:alice&from=sip:asterisk@mycooldomain.org&body=Hello+There This is equivalent to the following as well: ari/endpoints/PJSIP/alice/sendMessage?from=sip:asterisk@mycooldomain.org&body=Hello+There Both forms are available for message technologies that allow for arbitrary destinations, such as chan_sip. Inbound messages can now be received over ARI as well. An ARI application that subscribes to endpoints will receive messages from those endpoints: { "type": "TextMessageReceived", "timestamp": "2014-07-12T22:53:13.494-0500", "endpoint": { "technology": "PJSIP", "resource": "alice", "state": "online", "channel_ids": [] }, "message": { "from": "\"alice\" <sip:alice@127.0.0.1>", "to": "pjsip:asterisk@127.0.0.1", "body": "Watson, come here.", "variables": [] }, "application": "testsuite" } The above was made possible due to some rather major changes in the message core. This includes (but is not limited to): - Users of the message API can now register message handlers. A handler has two callbacks: one to determine if the handler has a destination for the message, and another to handle it. - All dialplan functionality of handling a message was moved into a message handler provided by the message API. - Messages can now have the technology/endpoint associated with them. Various other properties are also now more easily accessible. - A number of ao2 containers that weren't really needed were replaced with vectors. Iteration over ao2_containers is expensive and pointless when the lifetime of things is well defined and the number of things is very small. res_stasis now has a new file that makes up its structure, messaging. The messaging functionality implements a message handler, and passes received messages that match an interested endpoint over to the app for processing. Note that inadvertently while testing this, I reproduced ASTERISK-23969. res_pjsip_messaging was incorrectly parsing out the 'to' field, such that arbitrary SIP URIs mangled the endpoint lookup. This patch includes the fix for that as well. Review: https://reviewboard.asterisk.org/r/3726 ASTERISK-23692 #close Reported by: Matt Jordan ASTERISK-23969 #close Reported by: Andrew Nagy ........ r420090 \| mjordan \| 2014-08-05 15:16:37 -0500 (Tue, 05 Aug 2014) \| 2 lines Remove automerge properties :-( ........ r420097 \| mjordan \| 2014-08-05 16:36:25 -0500 (Tue, 05 Aug 2014) \| 2 lines test_message: Fix strict-aliasing compilation issue ........ Merged revisions 420089-420090,420097 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	48140b9808	Manager: Add PJSIPShowEndpoint[s] documentation This adds a large swath of response documentation for PJSIPShowEndpoint and PJSIPShowEndpoints AMI commands. It relies heavily on the existing text in the configInfo documentation via xi:include tags to avoid documentation duplication. Review: https://reviewboard.asterisk.org/r/3888/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	94b21f94d3	Add ContactStatusDetail to PJSIPShowEndpoint AMI output. Now when running PJSIPShowEndpoint, you will receive a ContactStatusDetail for each bound contact that Asterisk is qualifying. This information includes the URI of the contact, current reachability, and roundtrip time. AFS-91 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3797 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	b744adb8aa	PJSIP: Send Notify AMI and CLI commands can now send to URI instead of endpoint Review: https://reviewboard.asterisk.org/r/3817/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	bbeaeea1a3	res_hep_rtcp: Add module that sends RTCP information to a Homer Server This patch adds a new module to Asterisk, res_hep_rtcp. The module subscribes to the RTCP topics in Stasis and receives RTCP information back from the message bus. It encodes into HEPv3 packets and sends the information to the res_hep module for transmission. Using this, someone with a Homer server can get live call quality monitoring for all RTP-based channels in their Asterisk 12+ systems. In addition, there were a few bugs in the RTP engine, res_rtp_asterisk, and chan_pjsip that were uncovered by the tests written for the Asterisk Test Suite. This patch fixes the following: 1) chan_pjsip failed to set its channel unique ids on its RTP instance on outbound calls. It now does this in the appropriate location, in the serialized call callback. 2) The rtp_engine was overflowing some values when packed into JSON. Specifically, some longs and unsigned ints can't be be packed into integer values, for obvious reasons. Since libjansson only supports integers, floats, strings, booleans, and objects, we print these values into strings. 3) res_rtp_asterisk had a few problems: (a) it would emit a source IP address of 0.0.0.0 if bound to that IP address. We now use ast_find_ourip to get a better IP address, and properly marshal the result into an ast_strdupa'd string. (b) Reports can be generated with no report bodies. In particular, this occurs when a sender is transmitting information to a receiver (who will send no RTP back to the sender). As such, the sender has no report body for what it received. We now properly handle this case, and the sender will emit SR reports with no body. Likewise, if we receive an RTCP packet with no report body, we will still generate the appropriate events. ASTERISK-24119 #close ........ Merged revisions 419823 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	485d0379ae	manager: Add state list commands This patch adds three new AMI commands: * ExtensionStateList (pbx.c) - list all known extension state hints and their current statuses. Events emitted by the list action are equivalent to the ExtensionStatus events. * PresenceStateList (res_manager_presencestate) - list all known presence state values. Events emitted are generated by the stasis message type, and hence are PresenceStateChange events. * DeviceStateList (res_manager_devicestate) - list all known device state values. Events emitted are generated by the stasis message type, and hence are DeviceStateChange events. Patch-by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3799/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	e8a1e63498	res_pjsip_session: Fix race condition where redirecting information may not be set. Since the PJSIP INVITE session module is invoked before any session supplements it was possible for it to handle a redirect before the res_pjsip_diversion module interpreted and set redirecting information on the channel. This would cause the redirecting information to get lost. This patch ensures that session supplements are always invoked before a redirect occurs by explicitly calling them in the redirect handler. Review: https://reviewboard.asterisk.org/r/3850/ ........ Merged revisions 419764 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419766 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	e28f8936d3	res_pjsip_pidf_body_generator / res_pjsip_xpidf_body_generator: Ensure local entity is unquoted. The local entity as provided by PJSIP is quoted within '<' and '>'. As a result placing this value into XML will result in malformed XML being produced. This patch now unquotes the local entity so it can go safely into the XML. Review: https://reviewboard.asterisk.org/r/3851/ ........ Merged revisions 419750 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "module show" . ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	355dc3d2ad	Multiple revisions 419565-419566 ........ r419565 \| mjordan \| 2014-07-25 09:41:23 -0500 (Fri, 25 Jul 2014) \| 21 lines ARI: report duration values in LiveRecording objects This patch adds three new fields to the LiveRecording model: - total_duration: the total length of the live recording - talking_duration: optional. The duration of talking energy that was detected while the recording was made. - silence_duration: optional. The duration of silence that was detected while the recording was made. These values are reported in the RecordingFinished ARI event. When a DSP is enabled on the channel during the recording - which occurs when the recording is created with max_silence_seconds (indicating that the user actually cares about how much silence is in the file), we will report the talking_duration and silence_duration in addition to the total_duration. Review: https://reviewboard.asterisk.org/r/3770/ ASTERISK-24037 #close Reported by: Samuel Galarneau ........ r419566 \| mjordan \| 2014-07-25 09:46:15 -0500 (Fri, 25 Jul 2014) \| 1 line Update CHANGES for r419565 ........ Merged revisions 419565-419566 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ba9867fab0	module loader: Unload modules in reverse order of their start order When Asterisk starts a module (calling its load_module function), it re-orders the module list, sorting it alphabetically. Ostensibly, this was done so that the output of 'module show' listed modules in alphabetic order. This had the unfortunate side effect of making modules with complex usage patterns unloadable. A module that has a large number of modules that depend on it is typically abandoned during the unloading process. This results in its memory not being reclaimed during exit. Generally, this isn't harmful - when the process is destroyed, the operating system will reclaim all memory allocated by the process. Prior to Asterisk 12, we also didn't have many modules with complex dependencies. However, with the advent of ARI and PJSIP, this can make make unloading those modules successfully nearly impossible, and thus tracking memory leaks or ref debug leaks a real pain. While this patch is not a complete overhaul of the module loader - such an effort would be beyond the scope of what could be done for Asterisk 13 - this does make some marginal improvements to the loader such that modules like res_pjsip or res_stasis may be made properly un-loadable in the future. 1. The linked list of modules has been replaced with a doubly linked list. This allows traversal of the module list to occur backwards. The module shutdown routine now walks the global list backwards when it attempts to unload modules. 2. The alphabetic reorganization of the module list on startup has been removed. Instead, a started module is placed at the end of the module list. 3. The ast_update_module_list function - which is used by the CLI to display the modules - now does the sorting alphabetically itself. It creates its own linked list and inserts the modules into it in alphabetic order. This allows for the intent of the previous code to be maintained. This patch also contains a fix for res_calendar. Without calendar.conf, the calendar modules were improperly bumping the use count of res_calendar, then failing to load themselves. This patch makes it so that we detect whether or not calendaring is enabled before altering the use count. Review: https://reviewboard.asterisk.org/r/3777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419563 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a2ce95d9d2	accountcode: Slightly change accountcode propagation. The previous behavior was to simply set the accountcode of an outgoing channel to the accountcode of the channel initiating the call. It was done this way a long time ago to allow the accountcode set on the SIP/100 channel to be propagated to a local channel so the dialplan execution on the Local;2 channel would have the SIP/100 accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200 Propagating the SIP/100 accountcode to the local channels is very useful. Without any dialplan manipulation, all channels in this call would have the same accountcode. Using dialplan, you can set a different accountcode on the SIP/200 channel either by setting the accountcode on the Local;2 channel or by the Dial application's b(pre-dial), M(macro) or U(gosub) options, or by the FollowMe application's b(pre-dial) option, or by the Queue application's macro or gosub options. Before Asterisk v12, the altered accountcode on SIP/200 will remain until the local channels optimize out and the accountcode would change to the SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount support but ultimately had to punt on the support. The peeraccount support was rendered useless because of how the CDR code needed to unconditionally force the caller's accountcode onto the peer channel's accountcode. The CEL events were thus intentionally made to always use the channel's accountcode as the peeraccount value. With the arrival of Asterisk v12, the situation has improved somewhat so peeraccount support can be made to work. Using the indicated example, the the accountcode values become as follows when the peeraccount is set on SIP/100 before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already has an accountcode it can only change by the following explicit user actions: 1) A channel originate method that can specify an accountcode to use. 2) The calling channel propagating its non-empty peeraccount or its non-empty accountcode if the peeraccount was empty to the outgoing channel's accountcode before initiating the dial. e.g., Dial and FollowMe. The exception to this propagation method is Queue. Queue will only propagate peeraccounts this way only if the outgoing channel does not have an accountcode. 3) Dialplan using CHANNEL(accountcode). 4) Dialplan using CHANNEL(peeraccount) on the other end of a local channel pair. If a channel does not have an accountcode it can get one from the following places: 1) The channel driver's configuration at channel creation. 2) Explicit user action as already indicated. 3) Entering a basic or stasis-mixing bridge from a peer channel's peeraccount value. You can specify the accountcode for an outgoing channel by setting the CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue applications. Queue adds the wrinkle that it will not overwrite an existing accountcode on the outgoing channel with the calling channels values. Accountcode and peeraccount values propagate to an outgoing channel before dialing. Accountcodes also propagate when channels enter or leave a basic or stasis-mixing bridge. The peeraccount value only makes sense for mixing bridges with two channels; it is meaningless otherwise. * Made peeraccount functional by changing accountcode propagation as described above. * Fixed CEL extracting the wrong ie value for the peeraccount. This was done intentionally in Asterisk v1.8 when that version had to punt on peeraccount. * Fixed a few places dealing with accountcodes that were reading from channels without the lock held. AFS-65 #close Review: https://reviewboard.asterisk.org/r/3601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ef697de4a5	res_fax: unregister manager actions on unload * Unregister manager actions FAXSessions, FAXSession and FAXStats at unload. * Update ast_manager_register2 use ao2_t_alloc tagged with the action name. ASTERISK-24058 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3831/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	bb87796f67	ARI: Fix endpoint/channel subscription issues; allow for subscriptions to tech This patch serves two purposes: (1) It fixes some bugs with endpoint subscriptions not reporting all of the channel events (2) It serves as the preliminary work needed for ASTERISK-23692, which allows for sending/receiving arbitrary out of call text messages through ARI in a technology agnostic fashion. The messaging functionality described on ASTERISK-23692 requires two things: (1) The ability to send/receive messages associated with an endpoint. This is relatively straight forwards with the endpoint core in Asterisk now. (2) The ability to send/receive messages associated with a technology and an arbitrary technology defined URI. This is less straight forward, as endpoints are formed from a tech + resource pair. We don't have a mechanism to note that a technology that may have endpoints exists. This patch provides such a mechanism, and fixes a few bugs along the way. The first major bug this patch fixes is the forwarding of channel messages to their respective endpoints. Prior to this patch, there were two problems: (1) Channel caching messages weren't forwarded. Thus, the endpoints missed most of the interesting bits (such as channel creation, destruction, state changes, etc.) (2) Channels weren't associated with their endpoint until after creation. This resulted in endpoints missing the channel creation message, which limited the usefulness of the subscription in the first place (a major use case being 'tell me when this endpoint has a channel'). Unfortunately, this meant another parameter to ast_channel_alloc. Since not all channel technologies support an ast_endpoint, this patch makes such a call optional and opts for a new function, ast_channel_alloc_with_endpoint. When endpoints are created, they will implicitly create a technology endpoint for their technology (if one does not already exist). A technology endpoint is special in that it has no state, cannot have channels created for it, cannot be created explicitly, and cannot be destroyed except on shutdown. It does, however, have all messages from other endpoints in its technology forwarded to it. Combined with the bug fixes, we now have Stasis messages being properly forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar), where bar has a single channel associated with it and foo has two channels associated with it. The messages would be forwarded as follows: channel PJSIP/foo-1 -- \ --> endpoint PJSIP/foo -- / \ channel PJSIP/foo-2 -- \ ---- > endpoint PJSIP / channel PJSIP/bar-1 -----> endpoint PJSIP/bar -- ARI, through the applications resource, can: - subscribe to endpoint:PJSIP/foo and get notifications for channels PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo - subscribe to endpoint:PJSIP/bar and get notifications for channels PJSIP/bar-1 and endpoint PJSIP/bar - subscribe to endpoint:PJSIP and get notifications for channels PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar Note that since endpoint PJSIP never changes, it never has events itself. It merely provides an aggregation point for all other endpoints in its technology (which in turn aggregate all channel messages associated with that endpoint). This patch also adds endpoints to res_xmpp and chan_motif, because the actual messaging work will need it (messaging without XMPP is just sad). Review: https://reviewboard.asterisk.org/r/3760/ ASTERISK-23692 ........ Merged revisions 419196 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419203 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	9056c23bbd	Fix more dev-mode build issues ........ Merged revisions 419129 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 419162 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 419163 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	e04607f8a3	res_smdi: convert to astobj2 Remove functions: ast_smdi_interface_unref ast_smdi_md_message_putback ast_smdi_mwi_message_putback ast_smdi_md_message destructor ast_smdi_mwi_message destructor Includes for astobj.h are removed everywhere it's possible. ASTERISK-24066 #close Review: https://reviewboard.asterisk.org/r/3758/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a2c912e997	media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from and the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b299052e20	ari: Add a copy operation for stored recordings This patch adds a new operation for stored recordings, copy. It takes an existing stored recording and makes a copy of it in the same directory or a relative directory under the stored recording directory. /ari/recordings/stored/{recordingName}/copy?destinationRecordingName={copy_name} This is particularly useful for voicemail-esque applications, which may need to copy or move recordings around a directory structure. Review: https://reviewboard.asterisk.org/r/3768/ ASTERISK-24036 #close Reported by: Sam Galarneau Tested by: Sam Galarneau ........ Merged revisions 419021 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	88d8473746	Fix build in dev-mode git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	0a99e4099b	astobj2: assert on invalid ref and backtrace cleanup If a reference count goes negative, instead of just logging that fact, be more helpful with a backtrace and an assert that will DO_CRASH. This patch also removes the duplicate ao2_bt() function and cleans up extraneous usage of the ast_log_backtrace() call. Review: https://reviewboard.asterisk.org/r/3765/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	af4cd65143	Channels: Masquerades to automatically move frame/audio hooks Whenever possible, audiohooks and framehooks will now be copied over to the channel that the masquerading channel gets cloned into. This should occur for all audiohooks and most framehooks. As a result, in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now deprecated and its behavior is essentially the new default for all audiohooks, plus some additional audiohooks/framehooks. Review: https://reviewboard.asterisk.org/r/3721/ ........ Merged revisions 418914 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	5c988cc4e6	res_fax: Provide AMI equivalents for fax CLI commands Specifically the following equivalents were created: fax show session -> FAXSession fax show sessions -> FAXSessions fax show stats -> FAXStats Review: https://reviewboard.asterisk.org/r/3666/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fd94fea599	res_pjsip: Support setting a default accountcode on endpoints Most channel drivers let you specify a default accountcode to be set on channels associated with a particular peer/endpoint/object. Prior to this patch, chan_pjsip/res_pjsip did not support such a setting. This patch adds a new setting to the res_pjsip endpoint object, 'accountcode'. When a channel is created that is associated with an endpoint with this value set, the channel will automatically have its accountcode property set to the value configured for the endpoint. Review: https://reviewboard.asterisk.org/r/3724/ ASTERISK-24000 #close Reported by: Matt Jordan ........ Merged revisions 418756 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	03e9c598e5	cel_pgsql, cdr_pgsql, res_config_pgsql: Add PostgreSQL application_name support This patch adds support for the PostgreSQL application_name connection setting. When the appropriate PostgreSQL module's configuration is set with an application name, the name will be passed to PostgreSQL on connection and displayed in the database's pg_stat_activity view, as well as in CSV logs. This aids in managing which applications/servers are connected to a PostgreSQL database, as well as tracing the activity of those connections. Review: https://reviewboard.asterisk.org/r/3591 ASTERISK-23737 #close Reported by: Gergely Domodi patches: pgsql_application_name.patch uploaded by Gergely Domodi (License 6610) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418755 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	f962448eee	ARI: Make mixing bridges propagate linkedids and accountcodes. * Create a Stasis bridge sub-class to propagate linkedids and accountcodes. * Fixed the basic bridge sub-class to update peeraccount codes when the number of channels in the bridge drops back down to two parties. * Refactored ast_bridge_channel_update_accountcodes() to handle channels joining/leaving the bridge. * Fixed the basic bridge sub-class to not call the base bridge class pull method twice. AFS-105 #close ASTERISK-23852 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3720/ ........ Merged revisions 418225 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	3126d18c1b	res_rtp_asterisk: Fix undefined function when PJPROJECT is not installed The dtls_perform_handshake function was mistakenly placed under the guards for USE_PJPROJECT. If PJPROJECT was not installed, the function would not be defined, while other functions would attempt to still use it. This prevented res_rtp_asterisk from being loaded. ASTERISK-24001 #close Reported by: Don Fanning ........ Merged revisions 418172 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	534ffd8481	res_pjsip_dialog_info_body_generator: Add dialog-info+xml support for presence. This module implements dialog-info+xml for the purposes of presence. This means that phones such as Grandstreams can now subscribe to receive presence information for an extension. ASTERISK-21443 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3705/ ........ Merged revisions 418116 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d4b436d0ea	ARI/res_stasis: Subscribe to both Local channel halves when originating to app This patch fixes two bugs: 1. When originating a channel into a Stasis application, we already create a subscription for the channel that is going into our Stasis app. Unfortunately, when you create a Local channel and pass it off to a Stasis app, you really aren't creating just one channel: you're creating two. This patch snags the second half of the Local channel pair (assuming it is a Local channel pair, but luckily core_local is kind about such assumptions) and subscribes to it as well. 2. Subscriptions are a bit sticky right now. If a subscription is made, the 'interest' count gets bumped on the Stasis subscription - but unless something explicitly unsubscribes the channel, said subscription sticks around. This is not much of a problem is a user is creating the subscription - if they made it, they must want it. However, when we are creating implicit subscriptions, we need to make sure something clears them out. This patch takes a pessimistic approach: it watches the cache updates coming from Stasis and, if we notice that the cache just cleared out an object, we delete our subscription object. This keeps our ao2 container of Stasis forwards in an application from growing out of hand; it also is a bit more forgiving for end users who may not realize they were supposed to unsubscribe from that channel that just hung up. Review: https://reviewboard.asterisk.org/r/3710/ #ASTERISK-23939 #close ........ Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	edcaa54019	CEL: Fix incorrect/missing extra field information This corrects two issues with the extra field information in Asterisk 12+ in channel event logs. It is possible to inject custom values into the dialstatus provided by ast_channel_dial_type() Stasis messages that fall outside the enumeration allowed for the DIALSTATUS channel variable. CEL now filters for the allowed values and ignores other values. The "hangupsource" extra field key is always blank if the far end channel is a chan_pjsip channel. This is because the hangupsource is never set for the pjsip channel driver. This change sets the hangupsource whenever a hangup is queued for chan_pjsip channels. This corrects an issue with the pjsip channel driver where the hangupcause information was not being set properly. Review: https://reviewboard.asterisk.org/r/3690/ ........ Merged revisions 418071 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	97834718c2	Remove many deprecated modules Billing records are fair, To get paid is quite bright, You should really use ODBC; Good-bye cdr_sqlite. Microsoft did once push H.323, Hell, we all remember NetMeeting. But try to compile chan_h323 now And you will take quite a beating. The XMPP and SIP war was fierce, And in the distant fray Was birthed res_jabber/chan_jingle; But neither to stay. For everyone did care and chase what Google professed. "Free Internet Calling" was what devotees cried, But Google did change the specs so often That the developers were happy the day chan_gtalk died. And then there was that odd application Dedicated to the Polish tongue. app_saycountpl was subsumed by Say; One could say its bell was rung. To read and parse a file from the dialplan You could (I guess) use an application. app_readfile did fill that purpose, but I think A function is perhaps better in its creation. Barging is rude, I'm not sure why we do it. Inwardly, the caller will probably sigh. But if you really must do it, Don't use app_dahdibarge, use ChanSpy. We all despise the sound of tinny robots It makes our queues so cold. To control such an abomination It's better to not use Wait/SetMusicOnHold. It's often nice to know properties of a channel It makes our calls right We have a nice function called CHANNEL And so SIPCHANINFO is sent off into the night. And now things get odd; Apparently one could delimit with a colon Properties from the SIPPEER function! Commas are in; all others are done. Finally, a word on pipes and commas. We're sorry. We can't say it enough. But those compatibility options in asterisk.conf; To maintain them forever was just too tough. This patch removes: * cdr_sqlite * chan_gtalk * chan_jingle * chan_h323 * res_jabber * app_saycountpl * app_readfile * app_dahdibarge It removes the following applications/functions: * WaitMusicOnHold * SetMusicOnHold * SIPCHANINFO It removes the colon delimiter from the SIPPEER function. Finally, it also removes all compatibility options that were configurable from asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems. Review: https://reviewboard.asterisk.org/r/3698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	9b10813a2b	res_ari: Fix some off-nominal paths just dropping the HTTP connection. * Removed some incorrect newlines on ast_http_error() messages in manager.c. * Removed an incorrect newline in res_ari_channels.c. Addendum to ASTERISK-23552 ........ Merged revisions 417932 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	dbec5e0d8d	HTTP: Add persistent connection support. Persistent HTTP connection support is needed due to the increased usage of the Asterisk core HTTP transport and the frequency at which REST API calls are going to be issued. * Add http.conf session_keep_alive option to enable persistent connections. * Parse and discard optional chunked body extension information and trailing request headers. * Increased the maximum application/json and application/x-www-form-urlencoded body size allowed to 4k. The previous 1k was kind of small. * Removed a couple inlined versions of ast_http_manid_from_vars() by calling the function. manager.c:generic_http_callback() and res_http_post.c:http_post_callback() * Add missing va_end() in ast_ari_response_error(). * Eliminated unnecessary RAII_VAR() use in http.c:auth_create(). ASTERISK-23552 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/3691/ ........ Merged revisions 417880 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sam Galarneau	aa370d6105	ARI: Improvements to body parameters documentation The variables body parameter under the originate and originate with id operations of the channel resource showed invalid JSON in its description. The variables body parameter under the userEvent operation of the event resource made no mention that the custom key/value pairs should be wrapped in a variables key in order to be added to the custom user event. ASTERISK-23975 #close Review: https://reviewboard.asterisk.org/r/3692/ ........ Merged revisions 417878 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417879 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	a69e0cd32a	ARI: Remove unnecessary \briefs from automatically generated documentation Review: https://reviewboard.asterisk.org/r/3440/ ........ Merged revisions 412653 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	56a6cd0fa8	res_rtp_asterisk: Don't leak memory or reset state if DTLS configuration is set multiple times. ........ Merged revisions 417705 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	6e60f5d317	Recorded merge of revisions 417677 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP. This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation completes. Configuration options to chan_sip and chan_pjsip have also been added to allow behavior to be tweaked (such as forcing the AVP type media transports in SDP). ASTERISK-22961 #close Reported by: Jay Jideliov Review: https://reviewboard.asterisk.org/r/3679/ Review: https://reviewboard.asterisk.org/r/3686/ ........ Merged revisions 417678 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417679 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	688bb204dc	Reverse logic during subscription persistence recreation. In the abstraction effort, this bit of logic got messed up. We want to recreate the persistence if things go well, not if things fail. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	15dcaeef82	res_pjsip: Add ActionID to events created as a result of PJSIP AMI actions A number of various PJSIP AMI actions were failing to parse out and place the ActionID into their responses. This patch updates the various PJSIP actions such that the passed in ActionID is emitted on any event list complete events, as well as any intermediate events created as a result of the action. #ASTERISK-23947 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3675/ ........ Merged revisions 417460 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417461 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	435d20201c	res_http_websocket: Export symbol for ast_websocket_set_timeout Thanks to Sean Bright for pointing out that this was missed in #asterisk-dev. ........ Merged revisions 417419 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417420 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417421 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	365ae7523b	res_http_websocket: Close websocket correctly and use careful fwrite When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. Note that this version of the patch, unlike r417310 in Asterisk 11, exposes configuration options beyond just chan_sip's sip.conf. Configuration options to configure the write timeout have also been added to pjsip.conf and ari.conf. #ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ ........ Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	bc8c08c609	Abstract PJSIP-specific elements from the pubsub API. This helps to pave the way for RLS work that is to come. Since this is a self-contained change and subscription tests still pass, this work is being committed directly to trunk instead of a working branch. ASTERISK-23865 #close Review: https://reviewboard.asterisk.org/r/3628 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417233 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d5ca5b7f8f	res_rtp_asterisk: Return the length of data written when sending via ICE instead of 0. ASTERISK-23834 #close Reported by: Richard Kenner ........ Merged revisions 417141 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417142 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417143 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	a8cb4924cb	res_parking: Make manager commands register with module information Previously module information was not included due to an oversight. Review: https://reviewboard.asterisk.org/r/3626/ ........ Merged revisions 416849 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	d87f8c429e	pjsip cli: Change Identify to show CIDR notation instead of netmasks. * Added ast_sockaddr_cidr_bits() to count the 1 bits in an ast_sockaddr. * Added ast_ha_join_cidr() which uses ast_sockaddr_cidr_bits() for the netmask instead of ast_sockaddr_stringify_addr. * Changed res_pjsip_endpoint_identifier_ip to call ast_ha_join_cidr() instead of ast_ha_join() for the CLI output. This is a CLI change only. AMI was not affected. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3652/ ........ Merged revisions 416737 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	bd36288efa	Fix build warnings with TEST_FRAMEWORK enabled ........ Merged revisions 416732 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416733 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416734 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	351db2033c	Fix string growth algorithm for XML presence bodies. pjpidf_print() does not return < 0 if there is not enough room for the document to be printed. Rather, it returns 39, the length of the XML prolog. The algorithm also had a bug in that it would return if it attempted to grow the string larger. ........ Merged revisions 416442 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416444 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	163c734ae7	MoH: Don't restart stream on repeated start calls Currently, music on hold will stop and then start again from the beginning if ast_moh_start() is called multiple times. This can happen if a call is put on hold repeatedly (the channel receives multiple HOLD control frames) and can be triggered from ARI by starting MoH on a channel multiple times. This is fairly jarring/annoying to users. This change prevents MoH from being restarted if the requested music class is the same as the one currently playing. This includes an extra check to prevent the errors previously experienced in the testsuite and has 100+ test runs behind it. Review: https://reviewboard.asterisk.org/r/3615/ ........ Merged revisions 416439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416440 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416441 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	bd0aa4fb04	res_http_websocket: read/write string fixup There was a problem when reading a string from the websocket. It assumed the received data had a null terminator and tried to write the data to an ast_str. This of course could/would read past the end of the given buffer while writing the data to the internal buffer of ast_str. Modified the the code to correctly place a null terminator on the result string. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416394 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	a1e0a5e4b0	We have faced situation when using CDR and CEL by sqlite3 modules. With system having high load (~100 concurrent calls created by sipp) we found many cdr and cel records missed. There is special finction in sqlite3, that make able to fix this situation - sqlite3_wait_timeout, that also can replace awful code cdr_sqlite3 ad cel_sqlite3 modules. Also this function can be used for aastdb and res_config_sqlite3 to avoid missed writes to sqlite db.	12 years ago
Matthew Jordan	b52c6d0903	MoH: Undo commit r416150 (1.8) This patch reverts r416150. When the comparison between mohclass->name and state->class->name is made, you are not guaranteed that (a) state->class is non-NULL or that state or state->class are in a safe state. Crashes caught by the bridges/transfer_capabilities test. ........ Merged revisions 416251 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416252 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416255 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	eade1d490c	res_manager_devicestate and res_manager_presencestate missing support level Add MODULEINFO comment block to define support level core for these new modules. Review: https://reviewboard.asterisk.org/r/3620/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416237 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9cc1a8e893	stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	0acc626530	MoH: Don't restart stream on repeated start calls Currently, music on hold will stop and then start again from the beginning if ast_moh_start() is called multiple times. This can happen if a call is put on hold repeatedly (the channel receives multiple HOLD control frames) and can be triggered from ARI by starting MoH on a channel multiple times. This is fairly jarring/annoying to users. This change prevents MoH from being restarted if the requested music class is the same as the one currently playing. Review: https://reviewboard.asterisk.org/r/3615/ ........ Merged revisions 416150 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416151 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416152 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	27430374db	Fix build in devmode for GCC 4.10 ........ Merged revisions 415980 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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