967 Commits (c68c66c61fc1757aeb7e77832249bc855cec9691)

Author	SHA1	Message	Date
George Joseph	137fe5ae01	res_pjproject: Add module providing pjproject logging and utils ... res_pjsip_log_forwarder has been renamed to res_pjproject and enhanced as follows: As a follow-on to the recent 'Add CLI "pjsip show buildopts"' patch, a new ast_pjproject_get_buildopt function has been added. It allows the caller to get the value of one of the buildopts. The initial use case is retrieving the runtime value of PJ_MAX_HOSTNAME to insure we don't send a hostname greater than pjproject can handle. Since it can differ between the version of pjproject that Asterisk was compiled against and the version of pjproject that Asterisk is running against, we can't use the PJ_MAX_HOSTNAME macro directly in Asterisk source code. Change-Id: Iab6e82fec3d7cf00c1cf6185c42be3e7569dee1e	9 years ago
Joshua Colp	236896f391	Merge "pjsip: Add option global/regcontext" into 13	10 years ago
George Joseph	219c204a41	pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address ... On a system with multiple ip addresses in the same subnet, if a transport is bound to a specific ip address and endpoint/media_address is set, the SIP/SDP will have the correct address in all fields but the rtp stream MAY still originate from one of the other ip addresses, most probably the "primary" ip address. This happens because res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with the "all" ip address (0.0.0.0 or ::). The new option causes res_pjsip_sdp_rtp/create_rtp to call ast_rtp_instance_new with the endpoint's media_address (if specified) instead of the "all" address. This causes the packets to originate from the specified address. ASTERISK-25632 ASTERISK-25637 Reported-by: Olivier Krief Reported-by: Dan Journo Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88	10 years ago
Daniel Journo	22801a06ee	pjsip: Add option global/regcontext ... Added new global option (regcontext) to pjsip. When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given endpoint who registers or unregisters with us. ASTERISK-25670 #close Reported-by: Daniel Journo Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62	10 years ago
Rodrigo Ramírez Norambuena	e13719bff1	app_queue: Added reason pause of member ... In app_queue added value Paused Reason on QueueMemberStatus when a member on queue is paused and the reason was set. ASTERISK-25480 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e	10 years ago
George Joseph	4ec85a9f07	voicemail: Move app_voicemail / res_mwi_external conflict to runtime ... The menuselect conflict between app_voicemail and res_mwi_external makes it hard to package 1 version of Asterisk. There no actual build dependencies between the 2 so moving this check to runtime seems like a better solution. The ast_vm_register and ast_vm_greeter_register functions in app.c were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there is already a voicemail module registered. The modules' load_module functions were then modified to return DECLINE instead of -1 to the loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE, the modules were incorrectly causing Asterisk to stop so this needed to be cleaned up anyway. Now you can build both and use modules.conf to decide which voicemail implementation to load. The default menuselect options still build app_voicemail and not res_mwi_external but if both ARE built, res_mwi_external will load first and become the voicemail provider unless modules.conf rules prevent it. This is noted in CHANGES. Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247	10 years ago
Matt Jordan	dde7f3c1c4	res_pjsip_history: Add a module that provides PJSIP history for debugging ... This patch adds a new module, res_pjsip_history, that provides a slightly better way of debugging SIP message traffic on a busy Asterisk system. The existing mechanisms all rely on passively dumping a SIP message to the CLI. While this is perfectly fine for logging purposes and well controlled environments, on many installations, the amount of SIP messages Asterisk receives will quickly swamp the CLI. This makes it difficult to view/capture those messages that you want to diagnose in real time. This patch provides another way of handling this. When enabled, the module will store SIP message traffic in memory. This traffic can then be queried at leisure. In order to make the querying useful, a CLI command has been implemented, 'pjsip show history', that supports a basic expression syntax similar to SQL or other query languages. A small number of useful fields have been added in this initial patch; additional fields can easily be added in later improvements. Those fields are: - number: The entry index in the history - timestamp: The time the message was recieved - addr: The source/destination address of the message - sip.msg.request.method: The request method - sip.msg.call-id: The Call-ID header Note - this is a resurrection of the module initially proposed on Review Board here: https://reviewboard.asterisk.org/r/4053/ Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36	10 years ago
Joshua Colp	59d5bb0613	res_sorcery_memory_cache: Add support for a full backend cache. ... This change introduces the configuration option 'full_backend_cache' which changes the cache to be a full mirror of the backend instead of a per-object cache. This allows all sorcery retrieval operations to be carried out against it and is useful for object types which are used in a "retrieve all" or "retrieve some" pattern. ASTERISK-25625 #close Change-Id: Ie2993487e9c19de563413ad5561c7403b48caab5	10 years ago
Matt Jordan	529535f0c2	Revert "bridges/bridge_t38: Add a bridging module for managing T.38 state" ... This reverts commit 6614babea2. Unfortunately, using a bridge to manage T.38 state will cause severe deadlocks in core_unreal/chan_local. Local channels attempt to reach across both their peer and the peer's bridge to inspect T.38 state. Given the propensity of Local channel chains, managing the locking situation in such a scenario is practically infeasible. Change-Id: Ic687397ffea08dfb899345a443bd990ec3d0416a	10 years ago
Alexander Traud	5a18193dc0	res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8. ... ASTERISK-25584 #close Change-Id: Iae00071b4ff1ae76f24995aeac4d00284fd14f91	10 years ago
Matt Jordan	6614babea2	bridges/bridge_t38: Add a bridging module for managing T.38 state ... When 4875e5ac32 was merged, it fixed several issues with a direct media bridge transitioning to handling a T.38 fax. However, it uncovered a race condition caused by the bridging core. When a channel involved in a T.38 fax leaves a bridge, the frame queued by the channel driver that should inform the far side that it is no longer in a T.38 fax may not make it across the bridge. The bridging framework is *extremely* aggressive in tearing down the bridge, and control frames that are currently in flight *may* get dropped. This patch adds a new module to the bridging framework, bridge_t38. This module maintains some notion of the T.38 state for the two channels in a bridge. When the bridge detects that it is being torn down or when one of the two channels leaves, it informs the respective channel(s) that they should stop faxing. This ensures that channels switch back to audio if they survive and are ejected out of a bridge while faxing. ASTERISK-25582 Change-Id: If5b0bb478eb01c4607c9f4a7fc17c7957d260ea0	10 years ago
Niklas Larsson	3fcf160fae	CHANGES: Fix a typo ... Change-Id: Iceb3d9bb78140c376174a7bee197dfcf8ef9cda7	10 years ago
David M. Lee	59881fbb99	Fixed some typos ... Fixes some minor typos in the CHANGES file, plus an embarrasing typo in the StatsD API. Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7	10 years ago
Matt Jordan	d27aac0a9d	res/res_endpoint_stats: Add module to emit endpoint StatsD statistics ... This patch adds a module that emits StatsD statistics about Asterisk endpoints. This includes: * A GUAGE statistic for endpoint states, tracking how many endpoints are in a particular state. * A GUAGE statistic for each endpoint, counting the number of channels currently associated with an endpoint. ASTERISK-25572 Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305	10 years ago
Matt Jordan	90d9a70789	res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts ... This patch adds the ability to send StatsD statistics related to the state of PJSIP contacts. This includes: * A GUAGE statistic measuring the count of contacts in a particular state. This measures how many contacts are reachable, unreachable, etc. * The RTT time for each contact, if those contacts are qualified. This provides StatsD engines useful time-based data about each contact. ASTERISK-25571 Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c	10 years ago
Matt Jordan	75097a0955	res/res_pjsip_outbound_registration: Add registration statistics for StatsD ... This patch adds outbound registration statistics for StatsD. This includes the following: * A GUAGE metric for the overall count of outbound registrations. * A GUAGE metric for each state an outbound registration can be in. As the outbound registrations change state, the overall count of how many outbound registrations are in the particular state is changed. These statistics are particularly useful for systems with a large number of SIP trunks, and where measuring the change in state of the trunks is useful for monitoring. ASTERISK-25571 Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37	10 years ago
Mark Michelson	fdd2afcd16	Confbridge: Add a user timeout option ... This option adds the ability to specify a timeout, in seconds, for a participant in a ConfBridge. When the user's timeout has been reached, the user is ejected from the conference with the CONFBRIDGE_RESULT channel variable set to "TIMEOUT". The rationale for this change is that there have been times where we have seen channels get "stuck" in ConfBridge because a network issue results in a SIP BYE not being received by Asterisk. While these channels can be hung up manually via CLI/AMI/ARI, adding some sort of automatic cleanup of the channels is a nice feature to have. ASTERISK-25549 #close Reported by Mark Michelson Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98	10 years ago
Walter Doekes	6d1bdb9d3b	func_callerid: Document that CALLERID(pres) is available. ... CALLERPRES() says that it's deprecated in favor of CALLERID(num-pres) and CALLERID(name-pres). But for channel driver that don't make a distinction between the two (e.g. SIP), it makes more sense to get/set both at once. This change reveals the availability of CALLERID(pres), CONNECTEDLINE(pres), REDIRECTING(orig-pres), REDIRECTING(to-pres) and REDIRECTING(from-pres). ASTERISK-25373 #close Change-Id: I5614ae4ab7d3bbe9c791c1adf147e10de8698d7a	10 years ago
Corey Farrell	0393bd6bed	chan_sip: Allow websockets to be disabled. ... This patch adds a new setting "websockets_enabled" to sip.conf. Setting this to false allows chan_sip to be used without causing conflicts with res_pjsip_transport_websocket. ASTERISK-24106 #close Reported by: Andrew Nagy Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7	10 years ago
George Joseph	162acd45f7	res_pjsip: Add "like" processing to pjsip list and show commands ... Add the ability to filter output from pjsip list and show commands using the "like" predicate like chan_sip. For endpoints, aors, auths, registrations, identifyies and transports, the modification was a simple change of an ast_sorcery_retrieve_by_fields call to ast_sorcery_retrieve_by_regex. For channels and contacts a little more work had to be done because neither of those objects are true sorcery objects. That was just removing the non-matching object from the final container. Of course, a little extra plumbing in the common pjsip_cli code was needed to parse the "like" and pass the regex to the get_container callbacks. Some of the get_container code in res_pjsip_endpoint_identifier was also refactored for simplicity. ASTERISK-25477 #close Reported by: Bryant Zimmerman Tested by: George Joseph Change-Id: I646d9326b778aac26bb3e2bcd7fa1346d24434f1	10 years ago
Kevin Harwell	c58091737d	res_pjsip_outbound_registration: registration stops due to fatal 4xx response ... During outbound registration it is possible to receive a fatal (any permanent/ non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due to a problem with the registrar itself. Upon receiving the failure response Asterisk terminates outbound registration for the given endpoint. This patch adds an option, 'fatal_retry_interval', that when set continues outbound registration at the given interval up to 'max_retries' upon receiving a fatal response. ASTERISK-25485 #close Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2	10 years ago
Matt Jordan	72cbb6df55	funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function ... When ab803ec342 was committed, it accidentally forgot to actually *add* the HOLD_INTERCEPT function. This highlights two interesting points: * Gerrit forces you to put the patch as it is going to into the repo up for review, which Review Board did not. Yay Gerrit. * No one apparently bothered to use this feature, or else they don't know about it. I'm going to go with the latter explanation. ASTERISK-24922 Change-Id: Ida38278f259dd07c334a36f9b7d5475b5db72396	10 years ago
Kevin Harwell	d30939b6e8	ARI: Changed version from 1.8.0 to 1.9.0 ... Change-Id: I510991c60d28d171f47c4b58bba4947f7fc71b13	10 years ago
Matt Jordan	b50e372394	ARI: Add events for Contact and Peer Status changes ... This patch adds support for receiving events regarding Peer status changes and Contact status changes. This is particularly useful in scenarios where we are subscribed to all endpoints and channels, where we often want to know more about the state of channel technology specific items than a single endpoint's state. ASTERISK-24870 Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9	10 years ago
Matt Jordan	4c9f613309	ARI: Add the ability to subscribe to all events ... This patch adds the ability to subscribe to all events. There are two possible ways to accomplish this: (1) On initial WebSocket connection. This patch adds a new query parameter, 'subscribeAll'. If present and True, Asterisk will subscribe the applications to all ARI events. (2) Via the applications resource. When subscribing in this manner, an ARI client should merely specify a blank resource name, i.e., 'channels:' instead of 'channels:12354'. This will subscribe the application to all resources of the 'channels' type. ASTERISK-24870 #close Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6	10 years ago
Matt Jordan	78d0b9d97e	channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id ... This patch adds a new option to the CHANNEL function that allows for the extraction of the SIP call-id. It is used in conjunction with the 'pjsip' option, and will return the Call-ID of the INVITE request that established the PJSIP channel. ASTERISK-25352 Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a	10 years ago
Benjamin Ford	1ae762634c	ARI: Rotate log channels. ... An http request can be sent to rotate a specified log channel. If the channel does not exist, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/logging/logChannelName/rotate'" can be run in the terminal to access this new functionality. * Added the ability to rotate log files through ARI ASTERISK-25252 Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01	10 years ago
Joshua Colp	2749721791	pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. ... This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9	10 years ago
Mark Michelson	d9094ddd73	res_pjsip: Add rtp_keepalive endpoint option. ... This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I06660ba672c0a343814af4cec838e6025cafd54b	10 years ago
Matt Jordan	8bcf6d2801	ARI: Add support for push configuration of dynamic object ... This patch adds support for push configuration of dynamic, i.e., sorcery, objects in Asterisk. It adds three new REST API calls to the 'asterisk' resource: * GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current object given its ID. This returns back a list of ConfigTuples, which define the fields and their present values that make up the object. * PUT /asterisk/{configClass}/{objectType}/{id}: create or update an object. A body may be passed with the request that contains fields to populate in the object. The same format as what is retrieved using the GET operation is used for the body, save that we specify that the list of fields to update are contained in the "fields" attribute. * DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic object from its backing storage. Note that the success/failure of these operations is somewhat configuration dependent, i.e., you must be using a sorcery wizard that supports the operation in question. If a sorcery wizard does not support the create or delete mechanisms, then the REST API call will fail with a 403 forbidden. ASTERISK-25238 #close Change-Id: I28cd5c7bf6f67f8e9e437ff097f8fd171d30ff5c	10 years ago
Benjamin Ford	1aafadf814	ARI: Added new functionality to reload a single module. ... An http request can be sent to reload an Asterisk module. If the module can not be reloaded or is not already loaded, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/modules/{moduleName}'" (or something similar, based on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Asterisk modules can be reloaded through http requests ASTERISK-25173 Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1	10 years ago
Benjamin Ford	9dcae23cfc	ARI: Added new functionality to unload a single module. ... An http request can be sent to unload an Asterisk module. If the module can not be unloaded or is already unloaded, an error response will be returned. The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 /ari/asterisk/modules/{moduleName}'" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Asterisk modules can be unloaded through http requests ASTERISK-25173 Change-Id: I535a95f5676deb02651522761ecbdc0b00b5ac57	10 years ago
Benjamin Ford	c219a98d2b	ARI: Added new functionality to load a single module. ... An http request can be sent to load an Asterisk module. If the module can not be loaded or is loaded already, an error response will be returned. The command curl -v -u user:pass -X POST 'http://localhost:8088/ari /asterisk/modules/{moduleName}'" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Asterisk modules can be loaded through http requests ASTERISK-25173 Change-Id: I9e05d5b8c5c666ecfef341504f9edc1aa84fda33	10 years ago
Benjamin Ford	73e35d20de	ARI: Added new functionality to get information on a single module. ... An http request can be sent to retrieve information on a single module, including the resource name, description, use count, status, and support level. The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari /asterisk/modules/{moduleName}'" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Information on a single module can now be retrieved ASTERISK-25173 Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463	10 years ago
Benjamin Ford	47ea312b24	ARI: Added new functionality to get all module information. ... An http request can be sent to retrieve a list of all existing modules, including the resource name, description, use count, status, and support level. The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/ asterisk/modules" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Information on modules can now be retrieved Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0	10 years ago
Richard Mudgett	890c923786	AMI: Add Linkedid to the standard channel snapshot AMI event headers. ... * The AMI version is bumped to 2.8.0. ASTERISK-25189 #close Reported by: John Hardin Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac	10 years ago
Kevin Harwell	31c77b157b	res_pjsip: Add option to force G.726 to be treated as AAL2 packed. ... Some phones send g.726 audio packed for AAL2, which differs from what is recommended by RFC 3351. If Asterisk receives audio formatted as such when negotiating g.726 then it sounds a bit distorted. Added an option to res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 AAL2 packed. ASTERISK-25158 #close Reported by: Steve Pitts Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615	10 years ago
George Joseph	262d590819	res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes ... Add a new ContactStatus AMI event. Publish the following status/state changes: Created Removed Reachable Unreachable Unknown Contact URI, new status/state, aor and endpoint names, and the last qualify rtt result are included in the event. ASTERISK-25114 #close Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Joshua Colp	89f6719f7a	res_pjsip_outbound_registration: Add virtual line support. ... Virtual line support establishes a relationship between messages related to an outbound registration and a local endpoint. This is accomplished by attaching a parameter to the Contact of the outbound registration and looking for it on any received requests. If the parameter exists and can be matched to an outbound registration the configured endpoint is associated with the request. ASTERISK-24949 #close Reported by: Joshua Colp Change-Id: I7df909d2625479110a83fdd354c21ac539e8615d	10 years ago
George Joseph	674b18bdf0	pjsip_options: Add qualify_timeout processing and eventing ... This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html The basic issues are that changes in contact status don't cause events to be emitted for the associated endpoint. Only dynamic contact add/delete actions update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds which is a long time. This patch makes use of the new transaction timeout feature in r4585 and provides the following capabilities... 1. A new aor/contact variable 'qualify_timeout' has been added that allows the user to specify the maximum time in milliseconds to wait for a response to an OPTIONS message. The default is 3000ms. When the timer expires, the contact is marked unavailable. 2. Contact status changes are now propagated up to the endpoint as follows... When any contact is 'Available', the endpoint is marked as 'Reachable'. When all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The existing endpoint events are generated appropriately. ASTERISK-24863 #close Change-Id: Id0ce0528e58014da1324856ea537e7765466044a Tested-by: Dmitriy Serov Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
George Joseph	1b6f6ff841	res_pjsip: Add global option to limit the maximum time for initial qualifies ... Currently when Asterisk starts initial qualifies of contacts are spread out randomly between 0 and qualify_timeout to prevent network and system overload. If a contact's qualify_frequency is 5 minutes however, that contact may be unavailable to accept calls for the entire 5 minutes after startup. So while staggering the initial qualifies is a good idea, basing the time on qualify_timeout could leave contacts unavailable for too long. This patch adds a new global parameter "max_initial_qualify_time" that sets the maximum time for the initial qualifies. This way you could make sure that all your contacts are initialy, randomly qualified within say 30 seconds but still have the contact's ongoing qualifies at a 5 minute interval. If max_initial_qualify_time is > 0, the formula is initial_interval = min(max_initial_interval, qualify_timeout * random(). If not set, qualify_timeout is used. The default is "0" (disabled). ASTERISK-24863 #close Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4 Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Kevin Harwell	af458e2e60	chan_sip: make progressinband default to no ... After the "progressinband" value setting of "never" was updated to never send a 183 this separated its use from the "no" value. Since "never" was the default, but most users probably expect "no" this patch updates the default for the "progressinband" setting to "no." ASTERISK-24835 #close Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4606/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434654 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	5737650a67	res_pjsip: add CLI command to show global and system configuration ... Added a new CLI command for res_pjsip that shows both global and system configuration settings: pjsip show settings ASTERISK-24918 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/4597/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	ab803ec342	ARI: Add the ability to intercept hold and raise an event ... For some applications - such as SLA - a phone pressing hold should not behave in the fashion that the Asterisk core would like it to. Instead, the hold action has some application specific behaviour associated with it - such as disconnecting the channel that initiated the hold; only playing MoH to channels in the bridge if the channels are of a particular type, etc. One way of accomplishing this is to use a framehook to intercept the hold/unhold frames, raise an event, and eat the frame. Tasty. This patch accomplishes that using a new dialplan function, HOLD_INTERCEPT. In addition, some general cleanup of raising hold/unhold Stasis messages was done, including removing some RAII_VAR usage. Review: https://reviewboard.asterisk.org/r/4549/ ASTERISK-24922 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	b1e9552b08	chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages. ... Incoming PJSIP call legs that have not been answered yet send unnecessary "180 Ringing" or "183 Progress" messages every time a connected line update happens. If the outgoing channel is also PJSIP then the incoming channel will always send a "180 Ringing" or "183 Progress" message when the outgoing channel sends the INVITE. Consequences of these unnecessary messages: * The caller can start hearing ringback before the far end even gets the call. * Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of the first Asterisk box. When connected line first went into Asterisk in v1.8, chan_sip received an undocumented option "rpid_immediate" that defaults to disabled. When enabled, the option immediately passes connected line update information to the caller in "180 Ringing" or "183 Progress" messages as described above. * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or "183 Progress" messages. The default is "no" to disable sending the unnecessary messages. ASTERISK-24781 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4473/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	1f428f25f0	res_pjsip: Allow configuration of endpoint identifier query order ... This patch fixes previously reverted code that caused binary incompatibility problems with some modules. And like the original patch it makes sure that no matter what order the endpoint identifier modules were loaded, priority is given based on the ones specified in the new global 'endpoint_identifier_order' option. ASTERISK-24840 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4489/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	0497b7b155	Revert - res_pjsip: Allow configuration of endpoint identifier query order ... Due to a break in binary compatibility with some other modules these changes are being reverted until the issue can be resolved. ASTERISK-24840 Reported by: Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432868 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	110b99646c	res_pjsip: Allow configuration of endpoint identifier query order ... It's possible to have a scenario that will create a conflict between endpoint identifiers. For instance an incoming call could be identified by two different endpoint identifiers and the one chosen depended upon which identifier module loaded first. This of course causes problems when, for example, the incoming call is expected to be identified by username, but instead is identified by ip. This patch adds a new 'global' option to res_pjsip called 'endpoint_identifier_order'. It is a comma separated list of endpoint identifier names that specifies the order by which identifiers are processed and checked. ASTERISK-24840 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4455/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432638 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	1995baad71	ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app ... This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Mark Michelson	e257244bbb	Change PJProject version requirement for ca_list_path transport option in CHANGES file. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago