98 Commits (828a0611bceb1c17556e290aabc4d6c7abdbae3b)

Author	SHA1	Message	Date
Mark Michelson	3327560cb2	res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs. ... When SUBSCRIBE dialogs were established, we never associated the endpoint that created the subscription with the dialog we end up creating. In most cases, this ended up not causing any problems. The actual bug that was observed was that when a device that was behind NAT established a subscription with Asterisk, Asterisk would end up sending in-dialog NOTIFY requests to the device's private IP addres instead of the public address of the NAT router. When Asterisk receives the initial SUBSCRIBE from the device, res_pjsip_nat rewrites the contact to the public address on which the SUBSCRIBE was received. This allows for the dialog to have its target address set to the proper public address. Asterisk then would send a 200 OK response to the SUBSCRIBE, then a NOTIFY with the initial subscription state. The device would then send a 200 OK response to Asterisk's NOTIFY. Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat did not rewrite the address in the Contact header. Then, when the PJSIP dialog layer processed the 200 OK, PJSIP would perform a comparison between the IP address in the Contact header and its saved target address for the dialog. Since they differed, PJSIP would update the target dialog address to be the address in the Contact header. From this point, if Asterisk needed to send a NOTIFY to the device, the result was that the NOTIFY would be sent to the private address that the device placed in the Contact header. The reason why res_pjsip_nat did not rewrite the address when it received the 200 OK response was that it could not associate the incoming response with a configured endpoint. This is because on a response, the only way to associate the response to an endpoint is by finding the dialog that the response is associated with and then finding the endpoint that is associated with that dialog. We do not perform endpoint lookups on responses. res_pjsip_pubsub skipped the step of associating the endpoint with the dialog we created, so res_pjsip_nat could not find the associated endpoint and therefore couldn't rewrite the contact. This commit message is like 50x longer than the actual fix. ASTERISK 24981 #close Reported by Mark Michelson Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd	11 years ago
Scott Griepentrog	8d4ce7cc2b	res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced ... This change makes the send_notify of the sub_tree not happen when the sub_tree has been deleted due to the notify call failing, which avoids a crash. ASTERISK-24970 #close Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf	11 years ago
Corey Farrell	d0df545a44	res_pjsip: Enable unload of all modules at shutdown. ... * Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes caused by running PJSIP functions from non-PJSIP threads. * Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing crashes in some cases. In theory pj_shutdown() should take care of this. * Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at shutdown. * Resolve leaked config global in res_pjsip_notify. * Unregister pubsub pjsip service module. * Implement cleanup for res_pjsip_session. ASTERISK-24731 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4498/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	feddab7944	HTTP: Stop accepting requests on final system shutdown. ... There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	630eea087d	Investigate and fix memory leaks in Asterisk ... Fixed memory leaks that were found in Asterisk. ASTERISK-24693 #close Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/4347/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430999 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4b363688d4	AMI: Make AMI actions that generate event lists consistent. ... * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	b937438c17	res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdown ... If you do a 'core (shutdown|restart) graceful' persistent subscriptions won't survive. If you do a 'core (shutdown|restart) now' or asterisk terminates for some reason, they do. Here's why... When asterisk shuts down gracefully, it sends a 'NOTIFY/terminated' to subscribers for each subscription. This not only tells the subscribers that the dialog/state machine is done, it also frees the last reference to the subscription tree which causes the persistent subscription to get deleted from astdb. When asterisk restarts, nothing's left. Just preventing the delete from astdb doesn't work because we already told the subscriber to terminate the dialog so we can't restart it even if it was still in astdb. Everything works OK if asterisk terminates unexpectedly because we never send the 'terminated' message so on restart, the subscription is still in astdb and the subscriber is none the wiser. This patch suppresses the sending of 'NOTIFY/terminated' on shutdown for persistent connections. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4318/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	b85f79c0c1	Activate persistent subscriptions when they are recreated. ... Prior to this change, recreating persistent subscriptions would create the subscription but would not activate it. This led to subscriptions being listed in the "NULL" state by diagnostics and not sending NOTIFYs when expected. Review: https://reviewboard.asterisk.org/r/4261 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d79c68d3fb	main/stasis: Allow subscriptions to use a threadpool for message delivery ... Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	77c57f2b1a	Fix race condition where duplicated requests may be handled by multiple threads. ... This is the Asterisk 13 version of the patch. The main difference is in the pubsub code since it was completely refactored between Asterisk 12 and 13. Review: https://reviewboard.asterisk.org/r/4175 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	289830cdc6	PJSIP: Enforce module load dependencies ... This enforces that res_pjsip, res_pjsip_session, and res_pjsip_pubsub have loaded properly before attempting to load any modules that depend on them since the module loader system is not currently capable of resolving module dependencies on its own. ASTERISK-24312 #close Reported by: Dafi Ni Review: https://reviewboard.asterisk.org/r/4062/ ........ Merged revisions 425690 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425691 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	9f5c73586c	res/res_pjsip_pubsub: Fix typo in WARNING message ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424713 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f36b64f58e	res/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptions ... A subscription that has been persisted can - for various reasons - fail to be re-created on startup. This patch resolves a number of crashes that occurred when a subscription cannot be re-created on several off-nominal paths. #SIPit31 ASTERISK-24368 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424601 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2a7c5208ee	Simplify UUID generation in several places. ... Replace code using ast_uuid_generate() with simpler and faster code using ast_uuid_generate_str(). The new code avoids a malloc(), free(), and copy. ........ Merged revisions 424103 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	23f58d6f80	Add subscription state test events. ... These are needed for a set of batched notification RLS tests that are about to be committed to the testsuite. Review: https://reviewboard.asterisk.org/r/3967 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423462 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	7a35800551	res_pjsip_pubsub: Add some type safety when generating NOTIFY bodies. ... res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE arrives. * It checks that there is a subscription handler for the Event * It checks that there are body generators for the types in the Accept header The problem is, there's nothing that ensures that these two things will actually mesh with each other. For instance, Asterisk will accept a subscription to MWI that accepts pidf+xml bodies. That doesn't make sense. With this commit, we add some type information to the mix. Subscription handlers state they generate data of type X, and body generators state that they consume data of type X. This way, Asterisk doesn't end up in some hilariously mismatched situation like the one in the previous paragraph. ASTERISK-24136 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3877 Review: https://reviewboard.asterisk.org/r/3878 ........ Merged revisions 423344 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	9a5ee513d5	Pre-allocate transmission data buffer for RLS NOTIFY requests. ... PJSIP, unless a constant is modified at compilation time, limits SIP requests to 4000 bytes. Full-state RLS notifications can easily exceed this limit with moderately small lists. This changeset allows for Asterisk to work around this size limit by performing its own allocation of the transmission data buffer. This way, Asterisk can allocate a buffer that exceeds the built-in maximum. We still impose our own limit of 64000 bytes, mainly because making allocations larger than that is a bit absurd. ASTERISK-24181 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3977 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	346877f6c9	res_pjsip_pubsub: Check supported headers for eventlist when subscribing to ... resource list https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan According to the off-nominal plan, if evenlist support is not specified in a SUBSCRIBE's supported header(s), that subscription should be rejected with an error. ASTERISK-23871 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3960/diff/#index_header git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	7918e67d35	Fix bug that did not allow for multiple batched RLS notifications to be sent. ... A misunderstanding of how the scheduler worked caused further batched notifications beyond the first not to get scheduled. Now we reset our scheduler ID to -1 after the batched notification is sent. This way, further notifications can be scheduled when they arise. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	12891b608b	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	ef70c08dc7	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	1b500d2fa1	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420779 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	c43c22fe89	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	965ba7c36f	Fix build in dev mode ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420389 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	99d0bccd35	Add support for RFC 4662 resource list subscriptions. ... This commit adds the ability for a user to configure a resource list in pjsip.conf. Subscribing to this list simultaneously subscribes the subscriber to all resources listed. This has the potential to reduce the amount of SIP traffic when loads of subscribers on a system attempt to subscribe to each others' states. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	a8829490b6	res_pjsip_publish_asterisk: Add support for exchanging device and mailbox state using SIP. ... This module uses the inbound and outbound PUBLISH support to exchange device and mailbox state between Asterisk instances. Each instance is configured to publish to the other and requires no intermediary server. The functionality provided is similar to the XMPP and Corosync support. Review: https://reviewboard.asterisk.org/r/3780/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "module show" . ... ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	0a99e4099b	astobj2: assert on invalid ref and backtrace cleanup ... If a reference count goes negative, instead of just logging that fact, be more helpful with a backtrace and an assert that will DO_CRASH. This patch also removes the duplicate ao2_bt() function and cleans up extraneous usage of the ast_log_backtrace() call. Review: https://reviewboard.asterisk.org/r/3765/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	688bb204dc	Reverse logic during subscription persistence recreation. ... In the abstraction effort, this bit of logic got messed up. We want to recreate the persistence if things go well, not if things fail. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	15dcaeef82	res_pjsip: Add ActionID to events created as a result of PJSIP AMI actions ... A number of various PJSIP AMI actions were failing to parse out and place the ActionID into their responses. This patch updates the various PJSIP actions such that the passed in ActionID is emitted on any event list complete events, as well as any intermediate events created as a result of the action. #ASTERISK-23947 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3675/ ........ Merged revisions 417460 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417461 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	bc8c08c609	Abstract PJSIP-specific elements from the pubsub API. ... This helps to pave the way for RLS work that is to come. Since this is a self-contained change and subscription tests still pass, this work is being committed directly to trunk instead of a working branch. ASTERISK-23865 #close Review: https://reviewboard.asterisk.org/r/3628 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417233 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	27430374db	Fix build in devmode for GCC 4.10 ... ........ Merged revisions 415980 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	870394c051	res_pjsip_pubsub: unauthenticated remote crash in PJSIP pub/sub framework ... A remotely exploitable crash vulnerability exists in the PJSIP channel driver's pub/sub framework. If an attempt is made to unsubscribe when not currently subscribed and the endpoint's "sub_min_expiry" is set to zero, Asterisk tries to create an expiration timer with zero seconds, which is not allowed, so an assertion raised. The fix was to reject a subscription that is attempting to unsubscribe when not being already subscribed. Asterisk now checks for this situation appropriately and responds with a 400 instead of crashing. AST-2014-005 ASTERISK-23489 #close ........ Merged revisions 415812 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	58f4c18ab6	res_pjsip_pubsub: Persist subscriptions in sorcery so they are recreated on startup. ... This change makes res_pjsip_pubsub persist inbound subscriptions in sorcery. By default this uses the local astdb but it can also be configured to store within an outside database. When Asterisk is started these subscriptions are recreated if they have not expired. Notifications are sent to the devices which have subscribed and they are none the wiser that the system has restarted. Review: https://reviewboard.asterisk.org/r/3598/ ........ Merged revisions 415766 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 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	12 years ago
Kevin Harwell	6905ac0f5e	res_pjsip_pubsub: Set the body generation result to 0 for a valid path ... The result of the "ast_sip_pubsub_generate_body_content" was not set/initialized. Consequently, the nominal path potentially returned an invalid value, thus not sending mwi notifications. ........ Merged revisions 412074 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412075 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	fcb04d889a	PJSIP: Ensure test event has new state ... The change that fixed the pubsub test event's use of a dangling pointer also changed when it was processed relative to the pjsip subscription state change processing. This change corrects the order of events while holding a reference to the pointer that was previously dangling. ........ Merged revisions 411883 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	5d9a1281ee	PJSIP: Fix crash introduced in r411671 ... The test event introduced in revision 411671 uses a dangling pointer to access information about pubsub state changes. This moves the event to within the lifetime of the pointer. ........ Merged revisions 411790 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	045285f8e3	res_pjsip_pubsub: Add test event for state change ... This adds a test event when subscription state changes so that integration tests may trigger new actions at the appropriate times. Review: https://reviewboard.asterisk.org/r/3383/ ........ Merged revisions 411670 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	584f9bafa0	Remove all PJSIP MWI-specific use from our MWI code. ... PJSIP has built-in MWI code that could be useful to some degree, but our utilization of the API actually made our code a bit more cluttered since we had to have special cases peppered throughout. With this change, we move to using the pjsip_evsub API instead, which streamlines the code by removing special cases. Review: https://reviewboard.asterisk.org/r/3205 ........ Merged revisions 408005 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408006 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	f55abe9cf1	Decouple subscription handling from NOTIFY/PUBLISH body generation. ... When the PJSIP pubsub framework was created, subscription handlers were required to state what event they handled along with what body types they knew how to generate. While this serves well when implementing a base RFC, it has problems when trying to extend the body to support non-standard or proprietary body elements. The code also was NOTIFY-specific, meaning that when the time comes that we start writing code to send out PUBLISH requests with MWI or presence bodies, we would likely find ourselves duplicating code that had previously been written. This changeset introduces the concept of body generators and body supplements. A body generator is responsible for allocating a native structure for a given body type, providing the primary body content, converting the native structure to a string, and deallocating resources. A body supplement takes the primary body content (the native structure, not a string) generated by the body generator and adds nonstandard elements to the body. With these elements living in their own module, it becomes easy to extend our support for body types and to re-use resources when sending a PUBLISH request. Body generators and body supplements register themselves with the pubsub core, similar to how subscription and publish handlers had done. Now, subscription handlers do not need to know what type of body content they generate, but they still need to inform the pubsub core about what the default body type for a given event package is. The pubsub core keeps track of what body generators and body supplements have been registered. When a SUBSCRIBE arrives, the pubsub core will check that there is a subscription handler for the event in the SUBSCRIBE, then it will check that there is a body generator that can provide the content specified in the Accept header(s). Because of the nature of body generators and supplements, it means res_pjsip_exten_state and res_pjsip_mwi have been completely gutted. They no longer worry about body types, instead calling ast_sip_pubsub_generate_body_content() when they need to generate a NOTIFY body. Review: https://reviewboard.asterisk.org/r/3150 ........ Merged revisions 407016 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	565198b44b	res_pjsip_pubsub: potential crash on timeout ... What seems to be happening is if a subscription has been terminated and the subscription timeout/expires is less than the time it takes for all pending transactions (currently on the subscription) to end then the subscription timer will not have been canceled yet and sub will be null. Since the subscription has already been canceled nothing needs to be done so a null check in the asterisk code is sufficient in working around this problem. (closes issue ASTERISK-23129) Reported by: Dan Jenkins ........ Merged revisions 406847 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	c3d5c41dae	res_pjsip_pubsub: Ensure dialog manipulation happens on proper thread. ... When destroying a subscription we remove the serializer from its dialog and decrease its reference count. Depending on which thread dropped the subscription reference count to 0 it was possible for this to occur in a thread where it is not possible. (closes issue ASTERISK-22952) Reported by: Matt Jordan ........ Merged revisions 404553 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404554 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
Mark Michelson	9deb416397	Create more accurate Contact headers for dialogs when we are the UAS. ... (closes issue AST-1207) reported by John Bigelow Review: https://reviewboard.asterisk.org/r/2842 ........ Merged revisions 399083 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8990924953	Fix when the subscription_terminated callback is called for subscription handlers. ... The previous placement would result in the resubscribe() callback called instead of the subscription_terminated() callback being called when a subscription was ended via a SUBSCRIBE request. This would result in confusing PJSIP and having it throw an assertion. ........ Merged revisions 397955 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8049bf94f7	Handle default body types for SIP event packages in res_pjsip_pubsub ... Prior to this change, we would reject SUBSCRIBE requests that had no Accept headers. Now event package handlers that handle the default type for the event package indicate that they do so. Therefore, if we have a handler that can handle the default type, we can allow SUBSCRIBEs for the handler's event package that have no Accept headers. (closes issue ASTERISK-22067) reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/2774 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397441 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