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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
2015-01-30HTTP: For httpd server, need option to define server name for security purposesAshley Sanders
Added a new config property [servername] to the http.conf file; updated the http server to use the new property when sending responses, for showing http status through the CLI and when reporting status through the 'httpstatus' webpage. In this version, [servername] is uncommented by default. ASTERISK-24316 #close Reported By: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4374/ ........ Merged revisions 431471 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431484 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
2015-01-29Use 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 chan_sip. ASTERISK-24646 #close Reported by Stephan Eisvogel Review: https://reviewboard.asterisk.org/r/4346 ........ Merged revisions 431423 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431424 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431425 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
2015-01-28Multiple revisions 431297-431298Mark Michelson
........ r431297 | mmichelson | 2015-01-28 11:05:26 -0600 (Wed, 28 Jan 2015) | 17 lines Mitigate possible HTTP injection attacks using CURL() function in Asterisk. CVE-2014-8150 disclosed a vulnerability in libcURL where HTTP request injection can be performed given properly-crafted URLs. Since Asterisk makes use of libcURL, and it is possible that users of Asterisk may get cURL URLs from user input or remote sources, we have made a patch to Asterisk to prevent such HTTP injection attacks from originating from Asterisk. ASTERISK-24676 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/4364 AST-2015-002 ........ r431298 | mmichelson | 2015-01-28 11:12:49 -0600 (Wed, 28 Jan 2015) | 3 lines Fix compilation error from previous patch. ........ Merged revisions 431297-431298 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431299 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 431301 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431302 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
2015-01-27tcptls: Bad file descriptor error when reloading chan_sipKevin Harwell
While running through some scenarios using chan_sip and tcp a problem would occur that resulted in a flood of bad file descriptor messages on the cli: tcptls.c:712 ast_tcptls_server_root: Accept failed: Bad file descriptor The message is received because the underlying socket has been closed, so is valid. This is probably happening because unloading of chan_sip is not atomic. That however is outside the scope of this patch. This patch simply stops the logging of multiple occurrences of that message. ASTERISK-24728 #close Reported by: Thomas Thompson Review: https://reviewboard.asterisk.org/r/4380/ ........ Merged revisions 431218 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431219 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-27Manager: Fix Manager Action ModuleLoad to give correct response when reloadingJonathan Rose
Prior to this patch, ModuleLoad would respond with an error indicating that the requested module wasn't found in spite of finding and reloading the module. Review: https://reviewboard.asterisk.org/r/4373/ ASTERISK-24721 #close ........ Merged revisions 431153 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-27chan_sip: stale nonce causes failureKevin Harwell
When refreshing (with a small expiration) a registration that was sent to chan_sip the nonce would be considered stale and reject the registration. What was happening was that the initial registration's "dialog" still existed in the dialogs container and upon refresh the dialog match algorithm would choose that as the "dialog" instead of the newly created one. This occurred because the algorithm did not check to see if the from tag matched if authentication info was available after the 401. So, it ended up assuming the original "dialog" was a match and stopped the search. The old "dialog" of course had an old nonce, thus the stale nonce message. This fix attempts to leave the original functionality alone except in the case of a REGISTER. If a REGISTER is received if searches for an existing "dialog" matching only on the callid. If the expires value is low enough it will reuse dialog that is there, otherwise it will create a new one. ASTERISK-24715 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4367/ ........ Merged revisions 431187 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431194 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431197 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
2015-01-27app_confbridge: Repeatedly starting and stopping recording ref leaks the ↵Richard Mudgett
recording channel. Starting and stopping conference recording more than once causes the recording channels to be leaked. For v13 the channels also show up in the CLI "core show channels" output. * Reworked and simplified the recording channel code to use ast_bridge_impart() instead of managing the recording thread in the ConfBridge code. The recording channel's ref handling easily falls into place and other off nominal code paths get handled better as a result. ASTERISK-24719 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4368/ Review: https://reviewboard.asterisk.org/r/4369/ ........ Merged revisions 431135 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431160 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431161 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
2015-01-27app_confbridge: Restore user's menu name to CLI output of 'confbridge list'Matthew Jordan
When issuing a 'confbridge list XXXX' CLI command, the resulting output no longer displays the menu associated with a ConfBridge participant. The issue was caused by ASTERISK-22760. When that patch was done, it removed the copying of the menu name associated with the user from the actual user profile. This patch fixes the issue by copying the menu name over to the user profile when the menu hooks are applied to the user. Since that function now does a little bit more than just apply the hooks, the name of the function has been changed to cover the copying of the menu name over as well. In addition, there is a disparity between the menu name length as it is stored on the conf_menu structure and the confbridge_user structure; this patch makes the lengths match so that a strcpy can be used. Review: https://reviewboard.asterisk.org/r/4372/ ASTERISK-24723 #close Reported by: Steve Pitts ........ Merged revisions 431134 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431136 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
2015-01-25dynamic realtime: Updates fail to work due to update fields being passed overMatthew Jordan
When a crash was fixed due to usage of the REALTIME function in r423003, a regression was introduced into ast_update2_realtime where the update fields passed to the function would be skipped and the lookup field processed twice. The use of this function is a bit interesting: A variable argument list is used with two sentinel values - the first marks the end of the lookup fields/values; the second marks the end of the update fields/values. Unfortunately, ast_update2_realtime parses over the lookup fields twice, as opposed to parsing over the update fields. This causes the lookups to succeed, but the updates itself to have no effect. Note that the most common instance of this problem occurred in app_voicemail during the updating of a mailbox password. Thanks to the issue reporter, Paddy Grice, for pointing out the problem. Review: https://reviewboard.asterisk.org/r/4356/ ASTERISK-24231 ASTERISK-24626 #close Reported by: Paddy Grice ........ Merged revisions 431072 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431073 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-23app_confbridge: Shorten CBRec channel names to CBRec/<conf_name>-<seq-num>Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-23app_confbridge: Make CBRec channel names more unique.Richard Mudgett
Channel names should be different from other channels in the system while the channel exists. * Use a sequence number for CBRec channels instead of a random number because the same random number could be picked again for the next CBRec channel. ........ Merged revisions 431052 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-23app_confbridge: WhitespaceRichard Mudgett
Because there is sometimes no sence to any whitespace. ........ Merged revisions 431049 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431050 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431051 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
2015-01-23chan_sip: Case insensitive comparison of "defaultuser" parameter.Walter Doekes
All the other configuration options are case insensitive, so this one should be too. ASTERISK-24355 #close Reported by: HZMI8gkCvPpom0tM patches: ast.patch uploaded by HZMI8gkCvPpom0tM (License 6658) ........ Merged revisions 430993 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430994 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-22Bridge core: Pass a ref with the swap channel when joining a bridge.Richard Mudgett
When code imparts a channel into a bridge to swap with another channel, a ref needs to be held on the swap channel to ensure that it cannot dissapear before finding it in the bridge. * The ast_bridge_join() swap channel parameter now always steals a ref for the swap channel. This is the only change to the bridge framework's public API semantics. * bridge_channel_internal_join() now requires the bridge_channel->swap channel to pass in a ref. ASTERISK-24649 Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4354/ ........ Merged revisions 430975 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430976 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
2015-01-22apps/app_voicemail: Trigger MWI notification with MixMonitor m() optionMatthew Jordan
The MixMonitor m() option allows a recording to be pushed to a specific voicemail mailbox. If the message is delivered to the mailbox's INBOX, however, no MWI notification is currently raised. This patch corrects the issue by properly calling notify_new_state from the msg_create_from_file function. This will cause MWI to be triggered if the message was placed in the mailbox's INBOX. ASTERISK-24709 #close Reported by: Gareth Palmer patches: app_voicemail-430919.patch uploaded by Gareth Palmer (License 5169) ........ Merged revisions 430920 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430921 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430922 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
2015-01-21channels/chan_sip: Fix registration leak during reloadMatthew Jordan
When the SIP registrations were migrated to using ao2 in what was then trunk, the explicit destruction of the registrations on module reload was removed and not replaced with an ao2 equivalent. Debugging done by Stefan Engström, the issue reporter, on ASTERISK-24673 confirmed that the reference in the registry_list container was being leaked. Since the purpose of cleanup_all_regs is to prep a registration for destruction, this function now calls an ao2_callback function callback with the OBJ_MULTIPLE | OBJ_NODATA | OBJ_UNLINK flags used to remove the registrations. This cleans up each registration, and also removes it from the registration container registry_list. Review: https://reviewboard.asterisk.org/r/4355/ ASTERISK-24640 #close Reported by: Max Man ASTERISK-24673 #close Reported by: Stefan Engström Tested by: Stefan Engström ........ Merged revisions 430864 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-21AMI: Add documentation for the missing Cdr/CEL events.Matthew Jordan
This patch adds AMI event documentation for the Cdr and CEL AMI events. Note that while these events do share fields with each other and with other channel related events, they do not contain all of the fields in a standard channel snapshot, nor is the description of the fields identical. As such, the patch opts for documentation for each field, for each event. Review: https://reviewboard.asterisk.org/r/4350/ ASTERISK-24671 #close Reported by: Dan Jenkins ........ Merged revisions 430862 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-21apps/app_dial: Don't publish DialEnd twice on unexpected GoSub/Macro valuesMatthew Jordan
The Dial application has some interesting options with the mid-call Macro (M) and GoSub (U) options. If the MACRO_RESULT/GOSUB_RESULT returns specific values, the Dial application will take some action upon the channels involved in the dial operation (such as hanging up a particular party, etc.) The Dial application ensures that a Stasis message is published in the event that MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial operation, so that there is a corresponding DialEnd event published in AMI/ARI for the DialBegin event that preceeded it. A bug exists where that same DialEnd event will be published on Stasis even if the value returned in MACRO_RESULT/GOSUB_RESULT is not one that the Dial application cares about. This causes two DialEnd events to be published - one with the MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is all sorts of wrong. This patch fixes the bug by ensuring that we only publish a DialEnd message to Stasis if the Dial application's mid-call Macro/GoSub returns something that Dial cares about. Review: https://reviewboard.asterisk.org/r/4336 ASTERISK-24682 #close Reported by: Matt Jordan ........ Merged revisions 430842 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430844 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-21main/rtp_engine: Format NTP timestamps as unsigned longsMatthew Jordan
When the RTCP reports are created, the NTP timestamps are stored as strings, as JSON does not have an integer type long enough to store the value. However, on 32-bit systems, a signed long may overflow for some portion of the timestamp. This patch corrects the overflow by formatting the timestamps as unsigned longs. ........ Merged revisions 430840 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430841 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
2015-01-20contrib/scripts/install_prereq: Don't install 32-bit packages on 64-bit hostsMatthew Jordan
On Debian based systems, the install_prereq tool uses a search command on Debian that results in selecting both 64-bit and 32-bit packages. Besides the waste of disk space, this can actually cause aptitude use 100% of memory on a VM with 1GB of RAM as it tried to work out all of the 32-bit package dependencies. This patch filters out the 32-bit packages on a 64-bit machine, and leaves 32-bit machines alone. ASTERISK-24048 #close Reported by: Ben Klang Tested by: Ben Klang, Matt Jordan patches: install_prereq_64-bit_compat.patch uploaded by Ben Klang (License 5876) ........ Merged revisions 430798 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430799 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-20app_voicemail: Temp message left after review/hangup with ODBC/IMAP backendMatthew Jordan
When using ODBC or IMAP storage, temporary files created on the file system must be disposed of using the DISPOSE macro. The DELETE macro will map to a deletion function for the backend storage, but does not clean up any local files created as a result of the operation. When using voicemail with the operator and review options enabled, pressing 0 to enter the menu, followed by 1 to save the message, followed by any other DTMF press to delete the message, will result in the temporary file lingering on the file system. This patch properly calls DISPOSE after the DELETE. This causes the local file to be disposed of. ASTERISK-24288 #close Reported by: LEI FU patches: voicemail_odbc_review_fix.diff uploaded by LEI FU (License 6640) ........ Merged revisions 430795 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430796 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-19Call extension state callbacks at hint creation.Mark Michelson
When a hint gets created, any subsequent device or presence state changes result in extension status events getting sent out to interested parties. However, at the time of hint creation, no such event gets sent out, so watchers of extension state are potentially left in the dark until the first state change after hint creation. Patch contributed by John Hardin (License #6512) ........ Merged revisions 430776 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430777 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
2015-01-16Change PJProject version requirement for ca_list_path transport option in ↵Mark Michelson
CHANGES file. ........ Merged revisions 430716 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430717 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
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