asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	64c172deba	res_pjsip: Move URI validation to use time. In a realtime based system with a limited number of threadpool threads it is possible for a deadlock to occur. This happens when permanent endpoint state is updated, which will cause database queries to be done. These queries may result in URI validation being done which is done synchronously using a PJSIP thread. If all PJSIP threads are in use processing traffic they themselves may be blocked waiting to get the permanent endpoint container lock when identifying an endpoint. This change moves URI validation to occur at use time instead of configuration time. While this comes at a cost of not seeing a problem until you use it it does solve the underlying deadlock problem. ASTERISK-25486 #close Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a	10 years ago
Richard Mudgett	b8f07527b2	res_pjsip/location.c: Use the builtin ao2_callback() match function instead. Change-Id: I364906d6d2bad3472929986704a0286b9a2cbe3f	10 years ago
Joshua Colp	647cdcd6a8	Merge "res_pjsip: Use hash for contact object identity instead of Contact URI."	10 years ago
Joshua Colp	3628e380b8	res_pjsip: Use hash for contact object identity instead of Contact URI. In the wild it is possible for Contact URIs to be quite long as parameters can exist on them. This can present a problem when storing them in the AstDB as the URI is used as part of the object name and there is a fixed length limit for the AstDB. This will cause the contact to not get stored. This change uses the MD5 hash of the Contact URI as part of the object name instead. This has a fixed length which is guaranteed to not exceed the AstDB length limit. ASTERISK-25295 #close Change-Id: Ie8252a75331ca00b41b9f308f42cc1fbdf701a02	10 years ago
Matt Jordan	ef3358d0c0	res/res_pjsip: Purge contacts when an AoR is deleted When an AoR is deleted by an external mechanism, such as through ARI, we currently do not remove dynamic contacts that were created for that AoR as a result of a received REGISTER request. As a result, re-creating the AoR will cause the dynamic contact to be interpreted as a persistent contact, leading to some rather strange state being created for the contacts/endpoints. This patch adds a sorcery observer for the 'aor' object. When a delete is issued on the underlying sorcery object, the observer is called, and all contacts created and persisted in sorcery for that AoR are also removed. Note that we don't want to perform this action when an AO2 object that is an AoR is destroyed, as the AoR can still exist in the backing storage (and we would thus be removing valid contacts from an AoR that still "exists".) ASTERISK-25381 #close Change-Id: I6697e51ef6b2858b5d63401f35dc378bb0f90328	10 years ago
Joshua Colp	7846f73432	res_pjsip_mwi: Set up unsolicited MWI upon registration. The res_pjsip_mwi previously required a reload to set up the proper subscriptions to allow unsolicited MWI to work. This change makes it so the act of registering will also cause this to occur. This is particularly useful if realtime is involved as no reload needs to occur within Asterisk to cause the MWI information to get sent. ASTERISK-25180 #close Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252	10 years ago
George Joseph	d355ee7ff3	res_pjsip/location: Fix ref leak in contact_apply_handler contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status to force the creation of a contact_status object whenever a new contact is added but it didn't unref the returned object. Added an ao2_cleanup(status) to plug the leak. ASTERISK-25141 Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40 Reported-by: Corey Farrell	10 years ago
George Joseph	bef000dd7c	res_pjsip/location: Fix memory leak in permanent_uri_handler When permanent_uri_handler was creating the contact status object for each contact, it wasn't unreffing it at the end of the loop. ASTERISK-25141 #close Reported-by: Corey Farrell Change-Id: I7bb127994677bb3d459f87952f8425c9b9967b12	10 years ago
George Joseph	356568dc7f	res_pjsip: Fix SEGV on pending-qualify contacts Permanent contacts that hadn't been qualified yet were missing their contact_status entries causing SEGVs when running CLI commands. This patch makes sure that contact_statuses are created for both dynamic and permanent contacts when they are created. It also adds checks in the CLI code to make sure there's a contact_status, just in case. ASTERISK-25018 #close Reported-by: Ivan Poddubny Tested-by: Ivan Poddubny Tested-by: George Joseph Change-Id: I3cc13e5cedcafb24c400368b515b02d7fb81e029	10 years ago
George Joseph	75666ad7c6	res_pjsip: Validate that contact uris start with sip: or sips: Currently we use pjsip_parse_hdr to validate contact uris but it appears that it allows uris without a scheme if there's a port supplied. I.E myexample.com will fail but myexample.com:5060 will pass even though it has no scheme. This causes SEGVs later on whenever the uri is used. To prevent this, permanent_contact_validate has been updated to check that the scheme is either 'sip' or 'sips'. 2 uses of possibly-null endpoint have also been fixed in create_out_of_dialog_request. ASTERISK-24999 Change-Id: Ifc17d16a4923e1045d37fe51e43bbe29fa556ca2 Reported-by: Brad Latus	10 years ago
George Joseph	298faf7c50	pjsip_options: Fix non-qualified contacts showing as unavailable The "Add qualify_timeout processing and eventing" patch introduced an issue where contacts that had qualify_frequency set to 0 were showing Unavailable instead Unknown. This patch checks for qualify_frequency=0 and create an "Unknown" contact_status with an RTT = 0. Previously, the lack of contact_status implied Unknown but since we're now changing endpoint state based on contact_status, I've had to add new UNKNOWN status so that changes could trigger the appropriate contact_status observers. ASTERISK-24977: #close Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7	10 years ago
George Joseph	51886c68dc	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
Kevin Harwell	e62bd46511	res_pjsip: make it unloadable (take 2) Due to the original patch causing memory corruptions it was removed until the problem could be resolved. This patch is the original patch plus some added locking around stasis router subcription that was needed to avoid the memory corruption. Description of the original problem and patch (still applicable): The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue (listed below) by Corey Farrell with a few modifications. Namely, removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. This patch is the first step (should hopefully be followed by another/others at some point) in allowing res_pjsip and the modules that depend on it to be unloadable. At this time, res_pjsip and some of the modules that depend on res_pjsip cannot be unloaded without causing problems of some sort. The goal of this patch is to get res_pjsip and only res_pjsip to be able to unload successfully and/or shutdown without incident (crashes, leaks, etc...). Other dependent modules may still cause problems on unload. Basically made sure, with the patch applied, that res_pjsip (with no other dependent modules loaded) could be succesfully unloaded and Asterisk could shutdown without any leaks or crashes that pertained directly to res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4363/ patches: pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) ........ Merged revisions 431179 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	07e2a48ab1	REVERTING res_pjsip: make it unloadable Due to the original patch causing memory corruptions the patch is being removed until the problem can be resolved. ........ Merged revisions 430734 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	49542a794b	res_pjsip: make it unloadable The res_pjsip module was previously unloadable. With this patch it can now be unloaded. This patch is based off the original patch on the issue (listed below) by Corey Farrell with a few modifications. Namely, removed a few changes not required to make the module unloadable and also fixed a bug that would cause asterisk to crash on unloading. This patch is the first step (should hopefully be followed by another/others at some point) in allowing res_pjsip and the modules that depend on it to be unloadable. At this time, res_pjsip and some of the modules that depend on res_pjsip cannot be unloaded without causing problems of some sort. The goal of this patch is to get res_pjsip and only res_pjsip to be able to unload successfully and/or shutdown without incident (crashes, leaks, etc...). Other dependent modules may still cause problems on unload. Basically made sure, with the patch applied, that res_pjsip (with no other dependent modules loaded) could be succesfully unloaded and Asterisk could shutdown without any leaks or crashes that pertained directly to res_pjsip. ASTERISK-24485 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4311/ patches: pjsip_unload-broken-r1.patch submitted by Corey Farrell (license 5909) ........ Merged revisions 430628 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
George Joseph	7dc0c88fc6	pjsip cli: Fix sorting of contacts for 'pjsip list contacts' For some reason I was using a hash container instead of a list to gather the contacts for 'pjsip list/show contacts' so even though I had a sort function, the output wasn't sorted. This patch just changes the hash container to a list container and the contacts now appear sorted in the CLI. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4305/ ........ Merged revisions 430221 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Joshua Colp	f7cf988a82	pjsip: Add 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions. The PJSIP_AOR dialplan function allows inspection of configured AORs including what contacts are currently bound to them. The PJSIP_CONTACT dialplan function allows inspection of contacts in existence. These can include both externally added (by way of registration) or permanent ones. ASTERISK-24341 Reported by: xrobau Review: https://reviewboard.asterisk.org/r/4308/ ........ Merged revisions 430179 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kinsey Moore	4d2c7c23f8	PJSIP: Handle defaults properly This updates the code behind PJSIP configuration options with custom handlers to deal with the assigned default values properly where it makes sense and adjusting the default value where it doesn't. Before applying this patch, there were several cases where the default value for an option would prevent that config section from loading properly. Reported by: Thomas Thompson Review: https://reviewboard.asterisk.org/r/4019/ ........ Merged revisions 424263 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 424266 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424267 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	126334a7aa	res_pjsip: ami: Fix error in AMI output when an endpoint has no transport When no transport is associated to an endpoint, the AMI output for PJSIPShowEndpoint indicates an error instead of silently ignoring the missing transport. This patch causes the error to appear only if a transport was specified on the endpoint and the transport doesn't exist. It also fixes an issue with counting the objects that were actually found. ASTERISK-24161 #close ASTERISK-24331 #close Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3998/ ........ Merged revisions 423282 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 423284 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423285 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	abd3e4040b	Allow Asterisk to compile under GCC 4.10 This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	f5b8ab445f	Allow for multiple contacts to be configured in a single contact= line. This is useful for configuring multiple permanent contacts for an AOR when using realtime AORs. Review: https://reviewboard.asterisk.org/r/3462 ........ Merged revisions 412582 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	1a9ff2fffb	res_pjsip: Handle reloading when permanent contacts exist and qualify is configured. This change fixes a problem where permanent contacts being qualified were not being updated. This was caused by the permanent contacts getting a uuid and not a known identifier, causing an inability to look them up when updating in the qualify code. A bug also existed where the new configuration may not be available immediately when updating qualifies. (closes issue ASTERISK-23514) Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/3448/ ........ Merged revisions 412551 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	45ade68cb4	Remove unused RAII_VAR() declarations. * Remove unused RAII_VAR() declarations. The compiler cannot catch these because the cleanup function "references" the unused variable. Some actually allocated and released resources that were never used. * Fixed some whitespace issues in stasis_bridges.c. ........ Merged revisions 412399 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412400 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	3ff60b75b1	pjsip_cli: Create pjsip show channel and contact, and general cli code cleanup. Created the 'pjsip show channel' and 'pjsip show contact' commands. Refactored out the hated ast_hashtab. Replaced with ao2_container. Cleaned up function naming. Internal only, no public name changes. Cleaned up whitespace and brace formatting in cli code. Changed some NULL checking from "if"s to ast_asserts. Fixed some register/unregister ordering to reduce deadlock potential. Fixed ast_sip_location_add_contact where the 'name' buffer was too short. Fixed some self-assignment issues in res_pjsip_outbound_registration. (closes issue ASTERISK-23276) Review: http://reviewboard.asterisk.org/r/3283/ ........ Merged revisions 410287 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410288 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	a4906e9f86	sorcery: Create AST_SORCERY dialplan function. This patch creates the AST_SORCERY dialplan function which allows someone to retrieve any value from a sorcery-based config file. It's similar to AST_CONFIG. The creation of the function itself was fairly straightforward but it required changes to the underlying sorcery infrastructure that rippled into individual sorcery objects. The changes stemmed from inconsistencies in how sorcery created ast_variable objectsets from sorcery objects and the inconsistency in how individual objects used that feature especially when it came to parameters that can be specified multiple times like contact in aor and match in identify. You can read more here... http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html So, what this patch does, besides actually creating the AST_SORCERY function, is the following... * Creates ast_variable_list_append which is a helper to append one ast_variable list to another. * Modifies the ast_sorcery_object_field_register functions to accept the already-defined sorcery_fields_handler callback. * Modifies ast_sorcery_objectset_create to accept a parameter indicating return type preference...a single ast_variable with all values concatenated or an ast_variable list with multiple entries. Also fixed a few bugs. * Modifies individual sorcery object implementations to use the new function definition of the ast_sorcery_object_field_register functions. * Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement sorcery_fields_handler handlers so they return multiple occurrences as an ast_variable_list. * Added a whole bunch of tests to test_sorcery. (closes issue ASTERISK-22537) Review: http://reviewboard.asterisk.org/r/3254/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	39a450d924	pjsip_cli: Add pjsip commands 'show registrations' and 'show contacts'. Added 'show registrations' and 'show contacts' to pjsip cli to make things a little more consistent. The output is exactly the same as the list command. Just needed to add entries to their respective ast_cli_entry structures. (closes issue ASTERISK-23275) Review: http://reviewboard.asterisk.org/r/3210/ ........ Merged revisions 408522 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	ed66eefdf0	Store SIP User-Agent information in contacts. When an endpoint sends a REGISTER request to Asterisk, we now will associate the User-Agent header with all contacts that were bound in that REGISTER request. ........ Merged revisions 408270 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	db0d0363af	Fix crash in AMI PJSIPShowEndpoint action. If an AOR has no permanent contacts, then the permanent_contacts container is never allocated. This makes the code safe in the face of NULLs. I also changed the variable that counts contacts from "num" to "total_contacts" since there are now two variables that are indicate numbers of things. ........ Merged revisions 407988 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b5ca213e34	res_pjsip: Updates and adds more PJSIP CLI commands. * Adds identify, transport, and registration support to the PJSIP CLI. * Creates three additional callbacks, one for an iterator, one for a comparator, and one for a container. This eliminates the link dependency from higher level modules to lower level ones. * Eliminates duplicate sorting in PJSIP CLI commands. * Cleans up PJSIP CLI output formatting. * Pushes CLI command registration down to the implementing source file. * Adds several ast_sip_destroy_sorcery functions to complement existing ast_sip_sorcery_initialize functions. The destroy functions unregister PJSIP CLI commands and PJSIP CLI formatters. Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/3104/ ........ Merged revisions 407568 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	7cbb6eab15	PJSIP: Add Path header support This adds Path support to chan_pjsip in res_pjsip_path.c with minimal additions in res_pjsip_registrar.c to store the path and additions in res_pjsip_outbound_registration.c to enable advertisement of path support to registrars and intervening proxies. Path information is stored on contacts and is enabled via Address of Record (AoRs) and Registration configuration sections. While adding path support, it became necessary to be able to add SIP supplements that handled messages outside of sessions, so a framework for handling these types of hooks was added in parallel to the already-existing session supplements and several senders of out-of-dialog requests were refactored as a result. (closes issue ASTERISK-21084) Review: https://reviewboard.asterisk.org/r/3050/ ........ Merged revisions 405565 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	0ae89e7b7e	res_pjsip: Ensure more URI validation happens in pj threads. ........ Merged revisions 404737 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404738 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b172d369c4	res_pjsip: Add PJSIP CLI commands Implements the following cli commands: pjsip list aors pjsip list auths pjsip list channels pjsip list contacts pjsip list endpoints pjsip show aor(s) pjsip show auth(s) pjsip show channels pjsip show endpoint(s) Also... Minor modifications made to the AMI command implementations to facilitate reuse. New function ast_variable_list_sort added to config.c and config.h to implement variable list sorting. (issue ASTERISK-22610) patches: pjsip_cli_v2.patch uploaded by george.joseph (License 6322) ........ Merged revisions 404480 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	72c282cc66	ao2_iterator: Mini-audit of the ao2_iterator loops in the new code files. * Fixed several places where ao2_iterator_destroy() was not called. * Fixed several iterator loop object variable reference problems. * Fixed res_parking AMI actions returning non-zero. Only the AMI logoff action can return non-zero. Review: https://reviewboard.asterisk.org/r/3087/ ........ Merged revisions 404434 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d7b84b0368	res_pjsip: Apply outbound proxy to all SIP requests. Objects which are involved in SIP request creation and sending now allow an outbound proxy to be specified. For cases where an endpoint is used the outbound proxy specified there will be applied. (closes issue ASTERISK-22673) Reported by: Antti Yrjola Review: https://reviewboard.asterisk.org/r/3022/ ........ Merged revisions 403811 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403812 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	0620cc0c00	res_pjsip_transport_websocket: Fix security events and simplify implementation. Transport type determination for security events has been simplified to use the type present on the message itself instead of searching through configured transports to find the transport used. The actual WebSocket transport has also been simplified. It now leverages the existing PJSIP transport manager for finding the active WebSocket transport for outgoing messages. This removes the need for res_pjsip_transport_websocket to store a mapping itself. (closes issue ASTERISK-22897) Reported by: Max E. Reyes Vera J. Review: https://reviewboard.asterisk.org/r/3036/ ........ Merged revisions 403256 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	05cbf8df9b	res_pjsip: AMI commands and events. Created the following AMI commands and corresponding events for res_pjsip: PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few select attributes on each. Events: EndpointList - for each endpoint a few attributes. EndpointlistComplete - after all endpoints have been listed. PJSIPShowEndpoint - Provides a detail list of attributes for a specified endpoint. Events: EndpointDetail - attributes on an endpoint. AorDetail - raised for each AOR on an endpoint. AuthDetail - raised for each associated inbound and outbound auth TransportDetail - transport attributes. IdentifyDetail - attributes for the identify object associated with the endpoint. EndpointDetailComplete - last event raised after all detail events. PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound registrations. Events: InboundRegistrationDetail - inbound registration attributes for each registration. InboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowRegistrationsOutbound - Provides a detail listing of all outbound registrations. Events: OutboundRegistrationDetail - outbound registration attributes for each registration. OutboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. (issue ASTERISK-22609) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2959/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	113c281782	Perform validation of permanent contacts on AORs in res_pjsip. ........ Merged revisions 400833 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400834 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	735b30ad71	The large GULP->PJSIP renaming effort. The general gist is to have a clear boundary between old SIP stuff and new SIP stuff by having the word "SIP" for old stuff and "PJSIP" for new stuff. Here's a brief rundown of the changes: * The word "Gulp" in dialstrings, functions, and CLI commands is now "PJSIP" * chan_gulp.c is now chan_pjsip.c * Function names in chan_gulp.c that were "gulp_" are now "chan_pjsip_" * All files that were "res_sip" are now "res_pjsip" * The "res_sip" directory is now "res_pjsip" * Files in the "res_pjsip" directory that began with "sip_" are now "pjsip_" * The configuration file is now "pjsip.conf" instead of "res_sip.conf" * The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP" * CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as the starting word instead of "sip" git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395764 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

38 Commits (64c172debadfb6c5d6d4e424fdefd8ad86268b3b)