asterisk

Commit Graph

Author	SHA1	Message	Date
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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 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	12 years ago

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