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2015-02-13res_pjsip_session: Fix double re-INVITE collision crash.Richard Mudgett
A multi-asterisk box setup with direct media enabled would occasionally crash when two re-INVITE collisions on a call leg happen in a row. The re-INVITE logic only had one timer struct to defer the re-INVITE. When the second collision happens the timer struct is overwritten and put into the timer heap again. Resources for the first timer are leaked and the heap has two positions occupied by the same timer struct. Now the heap ordering is potentially corrupted, the timer will fire twice, and any resources allocated for the second timer will be released twice. * The solution is to put the collided re-INVITE into the delayed requests queue with all the other delayed requests and cherry pick the next request that can come off the queue when an event happens. * Changed to put delayed BYE requests at the head of the delayed queue. There is no sense in processing delayed UPDATEs and re-INVITEs when a BYE has been requested. * Made the start of a BYE request flush the delayed requests queue to prevent a delayed request from overlapping the BYE transaction. I saw a few cases where a delayed re-INVITE got started after the BYE transaction started. * Changed the delayed_request struct to use an enum instead of a string for the request method. Cherry picking the queue is easier with an enum than string comparisons and the compiler can warn if a switch statement does not cover all defined enum values. * Improved the debug output to give more information. It helps to know which channel is involved with an endpoint. Trunks can have many channels associated with the endpoint at the same time. ASTERISK-24727 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4414/ ........ Merged revisions 431734 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-12ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis appMatthew Jordan
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 ........ Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11res_pjsip: dtls_handler causes Asterisk to crashKevin Harwell
There have been a couple of times where a crash occurred in the dtls_handler section of the code for res_pjsip. Unfortunately, in working this issue the problem was unable to be reproduced. After looking at the backtraces and through the code the current best guess as to why this happened might be due to a reentrance problem and the strtok function. So, the current fix is to convert the strtok function into the reentrant version of the function, strtok_r. ASTERISK-24741 #close Reported by: Zane Conkle Review: https://reviewboard.asterisk.org/r/4409/ ........ Merged revisions 431698 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11ari_websockets: removed extra check on websocket session readKevin Harwell
When merging the websocket timeout issue (ASTERISK-24701) an extra, almost duplicate, check was left in the code that should not have been. This removes it. ASTERISK-24701 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4412/ ........ Merged revisions 431693 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431695 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11HTTP: Stop accepting requests on final system shutdown.Richard Mudgett
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/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11res_http_websocket: websocket write timeout fails to fully disconnectKevin Harwell
When writing to a websocket if a timeout occurred the underlying socket did not get closed/disconnected. This patch makes sure the websocket gets disconnected on a write timeout. Also a notice is logged stating that the websocket was disconnected. ASTERISK-24701 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4412/ ........ Merged revisions 431669 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431670 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-10res_pjsip_config_wizard: Add ability to auto-create hints.George Joseph
Looking at the Super Awesome Company sample reminded me that creating hints is just plain gruntwork. So you can now have the pjsip conifg wizard auto-create them for you. Specifying 'hint_exten' in the wizard will create 'exten => <hint_exten>,hint/PJSIP/<wizard_id>' in whatever is specified for 'hint_context'. Specifying 'hint_application' in the wizard will create 'exten => <hint_exten>,1,<hint_application>' in whatever is specified for 'hint_context'. The default for 'hint_context' is the endpoint's context. There's no default for 'hint_application'. If not specified, no app is added. There's no default for 'hint_exten'. If not specified, neither the hint itself nor the application will be created. Some may think this is the slippery slope to users.conf but hints are a basic necessity for phones unlike voicemail, manager, etc that users.conf creates. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4383/ ........ Merged revisions 431643 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-09res/ari/resource_channels: Add missing 'no_answer' reason to DELETE /channelsMatthew Jordan
One of the canonical reasons for hanging up a channel is because the far end failed to answer - or because someone else answered, and we want to get rid of this channel. This patch adds the missing value to the 'reason' query parameter for the DELETE /channels operation. Review: https://reviewboard.asterisk.org/r/4400 ASTERISK-24745 #close Reported by: Ben Merrills patches: add_no_answer_ari_hangup_cause.diff uploaded by Ben Merrills (License 6678) ........ Merged revisions 431622 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-09res/res_odbc: Remove unneeded queries when determining if a table existsMatthew Jordan
This patch modifies the ast_odbc_find_table function such that it only performs a lookup of the requested table if the table is not already known. Prior to this patch, a queries would be executed against the database even if the table was already known and cached. Review: https://reviewboard.asterisk.org/r/4405/ ASTERISK-24742 #close Reported by: ibercom patches: patch.diff uploaded by ibercom (License 6599) ........ Merged revisions 431617 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431618 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-08res/res_pjsip_sdp_rtp: Fix leak of local ICE candidates when applying to SDPMatthew Jordan
When an SDP is created for an outgoing request/response, the ICE candidates obtained from the RTP instance are currently leaked. This causes the ao2 container that holds the candidates to never properly be reclaimed when the RTP instance is destroyed. This patch properly decrements the ICE candidates' container if it is successfully obtained. ASTERISK-24769 #close Reported by: Matt Jordan ........ Merged revisions 431600 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-06various: cleanup issues found during leak huntScott Griepentrog
In this collection of small patches to prevent Valgrind errors are: fixes for reference leaks in config hooks, evaluating a parameter beyond bounds, and accessing a structure after a lock where it could have been already free'd. Review: https://reviewboard.asterisk.org/r/4407/ ........ Merged revisions 431583 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-04res_pjsip_keepalive: Don't crash if PJSIP module is not loaded.Joshua Colp
........ Merged revisions 431555 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431556 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-31res_format_attr_h264: Fix crash when determining joint capability.Joshua Colp
The res_format_attr_h264 module currently incorrectly attempts to copy SPS and PPS information from the wrong attribute. This change fixes that. ASTERISK-24616 #close Reported by: Yura Kocyuba Review: https://reviewboard.asterisk.org/r/4392/ ........ Merged revisions 431521 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431522 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-30res_pjsip_outbound_publish: eventually crashes when no response is ever receivedKevin Harwell
When Asterisk attempts to send SIP outbound publish information and no response is ever received (no 200 okay, 412, 423) the system eventually crashes. A response is never received because the system Asterisk is attempting to send publish information to is not available. The underlying pjsip framework attempts to send publish information. After several attempts it calls back into the Asterisk outbound publish code. At this point if the "client->queue" is empty Asterisk attempts to schedule a refresh which utilizes "rdata" and since no response was received the given "rdata" struture is NULL. Attempting to dereference a NULL object of course results in a crash. The fix here removes the dependency on rdata for schedule_publish_refresh. Instead param->expiration is now passed to it as this is set to -1 if no response is received. Also added a notification when no response is received. ASTERISK-24635 #close Reported by: Marco Paland Review: https://reviewboard.asterisk.org/r/4384/ ........ Merged revisions 431490 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-30Fix some memory leaks.Mark Michelson
These memory leaks were found and fixed by John Hardin. I'm just committing them for him. ASTERISK-24736 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/4389 ........ Merged revisions 431468 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-29stasis transfer: fix stasis bridge push race part twoScott Griepentrog
When swapping a Local channel in place of one already in a bridge (to complete a bridge attended transfer), the channel that was swapped out can actually be hung up before the stasis bridge push callback executes on the independant transfer thread. This results in the stasis app loop dropping out and removing the control that has the the app name which the local replacement channel needs so it can re-enter stasis. To avoid this race condition a new push_peek callback has been added, and called from the ast_bridge_impart thread before it launches the independant thread that will complete the transfer. Now the stasis push_peek callback can copy the stasis app name before the swap channel can hang up. ASTERISK-24649 Review: https://reviewboard.asterisk.org/r/4382/ ........ Merged revisions 431450 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-29Allow disabling of 100rel support on PJSIP endpoints.Mark Michelson
Due to an inversion error, setting 100rel=no would not actually change the current value of the setting (which defaulted to "yes"). With this fix, the inversion is corrected. ........ Merged revisions 431420 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-29Use SIPS URIs in Contact headers when appropriate.Mark Michelson
RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific scenarios when we are required to use SIPS URIs in Contact headers. Asterisk's non-compliance with this could actually cause calls to get dropped when communicating with clients that are strict about checking the Contact header. Both of the SIP stacks in Asterisk suffered from this issue. This changeset corrects the behavior in res_pjsip/chan_pjsip.c Review: https://reviewboard.asterisk.org/r/4345 ........ Merged revisions 431426 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-29res_pjsip_exten_state: Reduce log clutter... change a WARNING to a VERBOSE/2George Joseph
Reduce log clutter by changing the "Watcher for hint %s (removed|deactivated)" message from WARNING to VERBOSE/2. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4387/ ........ Merged revisions 431403 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431404 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-29res_rtp_asterisk: Fix DTLS when used with OpenSSL 1.0.1kJoshua Colp
A recent security fix for OpenSSL broke DTLS negotiation for many applications. This was caused by read ahead not being enabled when it should be. While a commit has gone into OpenSSL to force read ahead on for DTLS it may take some time for a release to be made and the change to be present in distributions (if at all). As enabling read ahead is a simple one line change this commit does that and fixes the issue. ASTERISK-24711 #close Reported by: Jared Biel ........ Merged revisions 431384 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431385 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28Fix file descriptor leak in RTP code.Mark Michelson
SIP requests that offered codecs incompatible with configured values could result in the allocation of RTP and RTCP ports that would not get reclaimed later. ASTERISK-24666 #close Reported by Y Ateya Review: https://reviewboard.asterisk.org/r/4323 AST-2015-001 ........ Merged revisions 431300 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 431303 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28media formats: update res_format_attr_opus & silkSean Bright
In r419044, we changed how formats were handled, but the return value of the format_parse_sdp_fmtp functions in res_format_attr_opus and res_format_attr_silk were not updated, causing calls to fail. Ran into this when getting codec_opus working with Asterisk 13. Once the return value was corrected, we were crashing in opus_getjoint because of NULL format attributes. I've fixed this as well in this patch. Review: https://reviewboard.asterisk.org/r/4371/ ........ Merged revisions 431267 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-28res_pjsip_outbound_registration: Fix reload race condition.Richard Mudgett
Performing a CLI "module reload" command when there are new pjsip.conf registration objects defined frequently failed to load them correctly. What happens is a race condition between res_pjsip pushing its reload into an asynchronous task processor task and the thread that does the rest of the reloads when it gets to reloading the res_pjsip_outbound_registration module. A similar race condition happens between a reload and the CLI/AMI show registrations commands. The reload updates the current_states container and the CLI/AMI commands call get_registrations() which builds a new current_states container. * Made res_pjsip.c reload_module() use ast_sip_push_task_synchronous() instead of ast_sip_push_task() to eliminate two threads processing config reloads at the same time. * Made get_registrations() not replace the global current_states container so the CLI/AMI show registrations command cannot interfere with reloading. You could never add/remove objects in the container without the possibility of the container being replaced out from under you by get_registrations(). * Added a registration loaded sorcery instance observer to purge any dead registration objects since get_registrations() cannot do this job anymore. The struct ast_sorcery_instance_observer callbacks must be used because the callback happens inline with the load process. The struct ast_sorcery_observer callbacks are pushed to a different thread. * Added some global current_states NULL pointer checks in case the container disappears because of unload_module(). * Made sorcery's struct ast_sorcery_instance_observer.object_type_loaded callbacks guaranteed to be called before any struct ast_sorcery_observer.loaded callbacks will be called. * Moved the check for non-reloadable objects to before the sorcery instance loading callbacks happen to short circuit unnecessary work. Previously with non-reloadable objects, the sorcery instance loading/loaded callbacks would always happen, the individual wizard loading/loaded would be prevented, and the non-reloadable type logging message would be logged for each associated wizard. ASTERISK-24729 #close Review: https://reviewboard.asterisk.org/r/4381/ ........ Merged revisions 431243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431251 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-27res_pjsip: make it unloadable (take 2)Kevin Harwell
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
2015-01-27bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during ↵Joshua Colp
direct media. This change fixes two issues: 1. During a swap operation bridging added the new channel before having the swap channel leave. This was not handled in bridge_native_rtp and could result in a channel not getting reinvited back to Asterisk. After this change the swap channel will leave first and the new channel will then join. 2. If a re-invite was received after a session had been established any upstream elements (such as bridge_native_rtp) were not notified that they may want to re-evaluate things. After this change an UPDATE_RTP_PEER control frame is queued when this situation occurs and upstream can react. AST-1524 #close Review: https://reviewboard.asterisk.org/r/4378/ ........ Merged revisions 431157 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-27ARI: Improve wiki documentationMatthew Jordan
This patch improves the documentation of ARI on the wiki. Specifically, it addresses the following: * Allowed values and allowed ranges weren't documented. This was particularly frustrating, as Asterisk would reject query parameters with disallowed values - but we didn't tell anyone what the allowed values were. * The /play/id operation on /channels and /bridges failed to document all of the added media resource types. * Documentation for creating a channel into a Stasis application failed to note when it occurred, and that creating a channel into Stasis conflicts with creating a channel into the dialplan. * Some other minor tweaks in the mustache templates, including italicizing the parameter type, putting the default value on its own sub-bullet, and some other nicities. Review: https://reviewboard.asterisk.org/r/4351 ........ Merged revisions 431145 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-27res_parking: Fix crash due to race condition when unloading.Joshua Colp
There is currently a race condition when unloading the res_parking module. Depending on the will of the universe the subscription invocation may occur AFTER the module is unloaded. This is because the module does NOT use stasis_unsubscribe_and_join when terminating the subscription. It merely uses stasis_unsubscribe. This change makes it use stasis_unsubscribe_and_join which is documented for usage in this exact scenario. AST-1520 #close Review: https://reviewboard.asterisk.org/r/4375/ ........ Merged revisions 431114 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-26Various fixes for OS XDavid M. Lee
This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sig*set() functions. * Fixed some format strings for portability. * Fixed compilation issues with res_timing_kqueue (although tests still fail on OS X). * Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue on OS X). ASTERISK-24539 #close Reported by: George Joseph ASTERISK-24544 #close Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/4327/ ........ Merged revisions 431092 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-23Add depend on pjproject to res_pjsip_config_wizard.cDavid M. Lee
........ Merged revisions 431030 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-23Investigate and fix memory leaks in AsteriskKevin Harwell
Fixed memory leaks that were found in Asterisk. ASTERISK-24693 #close Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/4347/ ........ Merged revisions 430999 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431010 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-23Fix typo's (retrieve, specified, address).Walter Doekes
........ Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430998 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-22res_pjsip_outbound_registration.c: Minor code cleanup.Richard Mudgett
* Add an allocation failure check and assert in sip_outbound_registration_response_cb(). * Made sip_outbound_registration_state_destroy() handle partially created state objects from sip_outbound_registration_state_alloc(). Review: https://reviewboard.asterisk.org/r/4366/ ........ Merged revisions 430957 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-22stasis transfer: fix a race condition on stasis bridge pushScott Griepentrog
After a bridge transfer completes where a local replacement channel is used, a stasis transfer message with the details of the transfer is sent. This is processed by stasis which then sets the stasis app name and replaced channel snapshot on the replacement channel. However, since a separate thread was already started to run stasis on the new replacement channel, a race was on to see if the message processing would be completed before the app name was needed, otherwise the channel would be hung up. This change moves the calls used to set the stasis app name and the replace snapshot to the bridge_stasis_push function callback from the bridge transfer logic, allowing the steps to be completed earlier and more deterministically, and the race elimianted. NOTE: the swap channel parameter to bridge_stasis_push (and thus all bridge push callbacks) must always be present when performing a swap with another channel. ASTERISK-24649 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4341/ ........ Merged revisions 430939 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-21res_pjsip_outbound_registration.c: Move unref to a better place.Richard Mudgett
Move an unconditional unref of client_state so it doesn't look like it could be used after the last ref has destroyed it. ........ Merged revisions 430902 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-20ARI: Fixed crash that occurred when updating a bridge when the optional ↵Ashley Sanders
query parameter 'name' was not supplied. Prior to this changeset, posting to the: /ari/bridges/{bridgeId} endpoint without specifying a value for the [name] query parameter, would crash Asterisk if the bridge you are attempting to create (or update) had the same ID as an existing bridge. The internal mechanism of the POST operation interpreted a null value for name, thus resulting in an error condition that crashed Asterisk. ASTERISK-24560 #close Reported By: Kinsey Moore Review: https://reviewboard.asterisk.org/r/4349/ ........ Merged revisions 430818 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-20CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching ↵Richard Mudgett
across a bridge. Calling ast_channel_bridge_peer() cannot be done while holding any channel locks. The reported issue hit the deadlock in chan_iax2, but an audit of the ast_channel_bridge_peer() calls found three more locations where the same deadlock can occur. * Made CHANNEL(peer), res_fax, and the SNMP agent not call ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I had to rework the logic to not hold the channel lock. * Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done for legacy reasons that no longer apply. * Removed the iax.conf forcejitterbuffer option. It is now always enabled when the jitterbuffer option is enabled. If you put a jitter buffer on a channel it will be on the channel. ASTERISK-24600 #close Reported by: Jeff Collell Review: https://reviewboard.asterisk.org/r/4342/ ........ Merged revisions 430817 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430819 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-19res_pjsip / res_pjsip_multihomed: Use the correct transport and addressing ↵Joshua Colp
information on UAS sessions. The first thing this patch fixes is UAS dialogs. Previously if a transport was configured on an endpoint and an inbound session was created there was no guarantee that requests sent on the dialog would use the correct transport and address information. This has now been fixed so an explicitly configured transport is taken into account. The second thing this patch fixes is res_pjsip_multihomed. The res_pjsip_multihomed module attempts to determine what transport a message should go out on and what addressing information should go into the message itself. In a scenario where multiple transports exist bound to the same IP address but a different port the code would incorrectly alter the transport and change the message to the wrong transport. This change makes the res_pjsip_multihomed module smarter so it will only change the transport and address information in the message when it is possible and makes sense. ASTERISK-24615 #close Reported by: David Justl Review: https://reviewboard.asterisk.org/r/4331/ ........ Merged revisions 430755 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430756 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-17REVERTING res_pjsip: make it unloadableKevin Harwell
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
2015-01-16Fix problem where a hung channel could occur on a failed blind transfer.Mark Michelson
Different clients react differently to being told that a blind transfer has failed. Some will simply send a BYE and be done with it. Others will attempt to reinvite themselves back onto the call. In the latter case, we were creating a new channel and then leaving it to sit forever doing nothing. With this code change, that new channel will not be created and the dialog with the transferring channel will be cleaned up properly. ASTERISK-24624 #close Reported by Zane Conkle Review: https://reviewboard.asterisk.org/r/4339 ........ Merged revisions 430714 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-16Add support for the ca_list_path option for PJSIP transports.Mark Michelson
This allows for a path to be specified that has a collection of CA certificates in it. ASTERISK-24575 #close Reported by cloos Patches: pj-ca-path-trunk.diff uploaded by cloos (License #5956) Review: https://reviewboard.asterisk.org/r/4344 ........ Merged revisions 430709 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-15res_fax.c, res_fax_spandsp.c: Remove redundant locking.Richard Mudgett
When FAX was developed, apparently the faxregistry.container used to be a linked list that was converted to an ao2 container. Some of the replacement ao2 container operations still had explicit lock/unlocks around them. Three off nominal code paths in res_fax.c and res_fax_spandsp.c unlock the channel even though the routine did not lock the channel and other code paths in the routine do not unlock the channel. Review: https://reviewboard.asterisk.org/r/4340/ ........ Merged revisions 430687 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-15res_fax.c, res_fax_spandsp.c: Fix some curlies on the end of function ↵Richard Mudgett
definitions. ........ Merged revisions 430685 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430686 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-15res_pjsip_outbound_registration: Fix race condition when reloading and ↵Joshua Colp
listing registrations. Due to the split of outbound registration state from configuration it is possible during a reload for a "pjsip show registrations" CLI command to be executed which gets an older snapshot of the configuration. This configuration may include outbound registrations which have been removed due to a reload operation occurring at the same time. The code for printing the outbound registration did not take this into account but now it does. AST-1506 #close Review: https://reviewboard.asterisk.org/r/4338/ ........ Merged revisions 430664 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-14res_pjsip: make it unloadableKevin Harwell
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
2015-01-14Prevent slow graceful shutdown when outbound publications never started.Mark Michelson
The code was missing the case for explicitly destroying an outbound publication when Asterisk had never actually published anything. The result was that Asterisk would hang for a while on a graceful shutdown. With this change, the case is taken into account, and on a graceful shutdown, these publications are destroyed without the need to actually send a PUBLISH request. ASTERISK-24655 #close Reported by Kevin Harwell Review: https://reviewboard.asterisk.org/r/4325 ........ Merged revisions 430608 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-12Revert -r430452 It needs to be redone for the next major AMI version change ↵Richard Mudgett
instead. ASTERISK-24049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09AMI: Remove no longer used parameter from astman_send_listack().Richard Mudgett
Follow-up issue to -r430435 from reviewboard review. ASTERISK-24049 Review: https://reviewboard.asterisk.org/r/4315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09AMI: Make AMI actions that generate event lists consistent.Richard Mudgett
* 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/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09res_fax: Add T.38 negotiation timeout optionKinsey Moore
This change makes the T.38 negotiation timeout configurable via 't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously hard coded to be 5000 milliseconds. This change also handles T.38 switch failures by aborting the fax since in the case where this can happen, both sides have agreed to switch to T.38 and Asterisk is unable to do so. Review: https://reviewboard.asterisk.org/r/4320/ ........ Merged revisions 430415 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430416 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-08res_pjsip_pubsub: Fix persistent subscriptions not surviving graceful shutdownGeorge Joseph
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/ ........ Merged revisions 430397 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430398 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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