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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
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
2015-01-15configure: If cross-compiling, assume we have working semaphoresMatthew Jordan
The Asterisk 13 configure.ac checks for HAS_WORKING_SEMAPHORE but does not have an option for cross-compiling so it fails with an exit. Since we're cross- compiling, we can't exactly go looking for the header. The semaphore.h header is relatively common: * It's part of the POSIX standard * It's part of GNU C Library As such, we assume that it will be present when cross-compiling. As such, this patch defaults "HAS_WORKING_SEMAPHORE" to "1" if cross-compiling is detected. If you're cross-compiling to a platform that doesn't support this, then make sure you re-define this to 0. ASTERISK-24663 #close Reported by: abelbeck patches: asterisk-13-anonymous-semaphores.patch uploaded by abelbeck (License 5903) ........ Merged revisions 430646 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430647 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
2015-01-14build_tools/mkpkgconfig: Fix Cflags concatenation error in asterisk.pcMatthew Jordan
The mkpkgconfig script incorrectly concatenates Cflags options together. As an example, the following: Cflags: -I/usr/include/libxml2 -g3 Is instead generated as: Cflags: -I/usr/include/libxml2-g3 This patch corrects the generation of Cflags in mkpkgconfig such that the Cflags options are output correctly. Review: https://reviewboard.asterisk.org/r/3707/ ASTERISK-23991 #close Reported by: Diederik de Groot patches: fix_mkpkgconfig.diff uploaded by Diederik de Groot (License 6600) ........ Merged revisions 430589 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430590 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-13app_macro: Don't restore the calling location on a channel redirect.Richard Mudgett
v11: If a channel redirect to a macro exten of a macro that is active happens, the redirect location doesn't get executed. Instead the original macro location is restored and gets reexecuted. v13: An additional effect happens if a parked call times out to an extension in the macro that parked the call then the macro is reexecuted instead of the expected park return location. * Made not restore the macro calling location on an AST_SOFTHANGUP_ASYNCGOTO. * Increased the locked channel range when setting up the macro execution environment to cover things that should be done while the channel is locked. * Removed unnecessary NULL tests before calling ast_free() in _macro_exec(). ASTERISK-23850 #close Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4292/ ........ Merged revisions 430564 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430565 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430567 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-13chan_pjsip: Add configure check for 'pjsip_get_dest_info' function.Joshua Colp
The 'pjsip_get_dest_info' function is used to determine if the signaling transport of the dialog is secure or not. This function was added in PJSIP 2.3 and does not exist in earlier versions. This configure check allows Asterisk to build and run with older versions at the loss of the 'secure' argument for the PJSIP CHANNEL dialplan function. Usage of this argument will require upgrading to PJSIP 2.3. ASTERISK-24665 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4329/ ........ Merged revisions 430546 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-12AMI: Revert non-backwards compatible changes from earlier commit.Richard Mudgett
* Reverted the change to astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. Unfortunately changing the case of a returned value is not a backward compatible change so for now FAXSessions is going to have to remain inconsistent with all of the other AMI list actions. * Reverted the minor protocol error fix in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". Caught by the testsuite. ASTERISK-24049 ........ Merged revisions 430528 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-12configs/samples/features.conf.sample: Document attended transfer DTMF optionsMatthew Jordan
The sample config was missing the configuration options for DTMF attended transfer completion scenarios. The configuration options 'atxferabort', 'atxfercomplete', 'atxferthreeway', and 'atxferswap' are now documented in the appropriate configuration file. ASTERISK-24678 #close Reported by: Niklas Larsson patches: features.conf.sample.diff uploaded by Niklas Larsson (License 5068) ........ Merged revisions 430526 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430527 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
2015-01-12main/syslog: Allow dynamic logs, such as security events, to log to the syslogMatthew Jordan
The security event log uses a dynamic log level (SECURITY) that is registered with the Asterisk logging core. Unfortunately, the syslog would ignore log statements that had a dynamic log level associated with them. Because the syslog cannot handle ad hoc dynamic log levels, this patch treats any dynamic log entries sent to the syslog as logs with a level of NOTICE. ASTERISK-20744 #close Reported by: Michael Keuter Tested by: Michael L. Young, Jacek Konieczny patches: asterisk-20744-syslog-dynamic-logging_trunk.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 430506 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430507 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-12funcs/func_curl: Fix memory leak when CURLOPT channel datastore is destroyedMatthew Jordan
When the channel datastore associated with the usage of CURLOPT on a specific channel is freed, the underlying structure holding the list of options is not disposed of. This patch properly frees the structure in the datastore .destroy callback. ASTERISK-24672 #close Reported by: Kristian Hogh patches: func_curl-memory-leak.diff uploaded by Kristian Hogh (License 6639) ........ Merged revisions 430487 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430488 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09sip_to_pjsip: improve ability to parse input filesScott Griepentrog
General improvements to SIP to PJSIP conversion utility: 1) track default section of input file to allow parsing an include file that doesn't specify a [section] 2) informatively handle case of assignment without [section] 3) correctly handle getting sections from included files - [section]'s are inherited by included file 4) provide null string as default transport bind ip 5) gracefully handle missing portions of registration string 6) denote steps of operation during conversion and confirm top level files as a convenience ASTERISK-24474 #close Review: https://reviewboard.asterisk.org/r/4280/ Reported by: John Kiniston ........ Merged revisions 430469 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09app_bridge: return to the next dialplan priorityScott Griepentrog
When app_bridge grabs a channel and puts it into a bridge, the channel should then continue where it left off in the dialplan after the bridge has ended. Although it stores the current dialplan location as an after bridge goto on the channel, it was executing the same priority again instead of going to the next priority. By swapping the "specific" version of bridge_set_after_goto with bridge_set_after_go_on, the next priority in the dialplan is executed instead. ASTERISK-24637 #close Review: https://reviewboard.asterisk.org/r/4322/ Reported by: John Bigelow ........ Merged revisions 430467 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430468 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
2015-01-08res_pjsip_outbound_registration: Fix reference leak.George Joseph
Every time a registration started, sip_outbound_registration_response_cb bumps the ref count on client_state then pushes a handle_registration_response task. handle_registration_response never unreffed it though. So every time a registration goes out, the ref count goes up by one. This patch adds the unreffs to handle_registration_response. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4303/ ........ Merged revisions 430395 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-08res_pjsip_outbound_registration: Fix several reload issuesGeorge Joseph
There are 2 issues with reloading registrations... 1. The 'can_reuse_registration' test wasn't considering the intervals or expiration in its determination of whether a registration changed or not so if you changed any of the intervals or the expiration and reloaded, the object would get reloaded but the actual timers wouldn't change. can_reuse_registration now does a sorcery diff on the old and new objects instead of discretely testing certain fields. Now if you change expiration for instance, and reload, the timer is updated and re-registration will occur on the new value. 2. If you mung up your password on an outbound registration you get a permanent failure. If you fix the password (on the outbound_auth object) and reload, nothing tells outbound_registration to try again because the registration itself didn't change. This patch adds an observer on the "auth" object type and if any auth changes, existing registration states are searched and those in a REJECTED_PERMANENT state are retried. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4304/ ........ Merged revisions 430373 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07ARI: Allow usage of ASYNCGOTO with Stasis()Kinsey Moore
When the AMI Redirect action is used with a channel bridged inside Stasis() and not running a pbx, the channel is hung up instead of proceeding to the desired location in dialplan. This change allows such channels to be Redirected properly by detecting the operation used by Redirect (ASYNCGOTO) and using the code already established for functionality of the ARI channel continue operation. ASTERISK-24591 #close Review: https://reviewboard.asterisk.org/r/4271/ ........ Merged revisions 430355 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430356 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07Add the ability to continue and originate using priority labels.Mark Michelson
With this patch, the following two ARI commands POST /channels POST /channels/{id}/continue Accept a new parameter, label, that can be used to continue to or originate to a priority label in the dialplan. Because this is adding a new parameter to ARI commands, the API version of ARI has been bumped from 1.6.0 to 1.7.0. This patch comes courtesy of Nir Simionovich from Greenfield Tech. Thanks! ASTERISK-24412 #close Reported by Nir Simionovich Review: https://reviewboard.asterisk.org/r/4285 ........ Merged revisions 430337 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07res_pjsip_exten_state: Change 'does not exist' warning to noticeGeorge Joseph
The 'new_subscribe: Extension <> does not exist or has no associated hint' is a config issue and doesn't need to clutter up logs with warnings. Changed to notice. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4307/ ........ Merged revisions 430319 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07res_pjsip_mwi: Change "MWI Subscription failed" message from warning to noticeGeorge Joseph
The "MWI Subscription failed" message means the client is trying to subscribe to a mailbox that doesn't exist. There's no need to clutter up logs with warnings for a client misconfiguration so I changed it to a notice. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4306/ ........ Merged revisions 430317 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07func_config: Add ability to retrieve specific occurrence of a variableGeorge Joseph
I guess nobody uses templates with AST_CONFIG because today if you have a context that inherits from a template and you call AST_CONFIG on the context, you'll get the value from the template even if you've overridden it in the context. This is because AST_CONFIG only gets the first occurrence which is always from the template. This patch adds an optional 'index' parameter to AST_CONFIG which lets you specify the exact occurrence to retrieve, or '-1' to retrieve the last. The default behavior is the current behavior. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4313/ ........ Merged revisions 430315 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07Fix ability to perform a remote attended transfer with PJSIP.Mark Michelson
This fix has two parts: * Corrected an error message to properly state that external_replaces is an extension. The error message also prints what dialplan context the external_replaces extension was being looked for in. * Corrected the printing of the Replaces: header in an INVITE request. We were duplicating "Replaces: " in the header. ASTERISK-24376 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/4296 ........ Merged revisions 430313 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07config: Add option to NOT preserve effective context when changing a templateGeorge Joseph
Let's say you have a template T with variable VAR1 = ON and you have a context C(T) that doesn't specify VAR1. If you read C, the effective value of VAR1 is ON. Now you change T VAR1 to OFF and call ast_config_text_file_save. The current behavior is that the file gets re-written with T/VAR1=OFF but C/VAR1=ON is added. Personally, I think this is a bug. It's preserving the effective state of C even though I didn't specify C/VAR1 in th first place. I believe the behavior should be that if I didn't specify C/VAR1 originally, then the effective value of C/VAR1 should continue to follow the inherited state. Now, if I DID explicitly specify C/VAR1, the it should be preserved even if the template changes. Even though I think the existing behavior is a bug, it's been that way forever so I'm not changing it. Instead, I've created ast_config_text_file_save2() that takes a bitmask of flags, one of which is to preserve the effective context (the current behavior). The original ast_config_text_file_save calls *2 with the preserve flag. If you want the new behavior, call *2 directly without a flag. I've also updated Manager UpdateConfig with a new parameter 'PreserveEffectiveContext' whose default is 'yes'. If you want the new behavior with UpdateConfig, set 'PreserveEffectiveContext: no'. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4297/ ........ Merged revisions 430295 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07Fix dev-mode build on recent gccKinsey Moore
........ Merged revisions 430274 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-06Blocked revisions 430252Matthew Jordan
........ contrib/ast-db-manage: Correct down_revision path for user_eq_phone When the user_eq_phone patch was backported to 13, it referenced the downward revision that the PJSIP optimistic encryption option also references. This creates a multi-path upgrade Exception when generating the SQL files. This patch corrects this in the 13 branch. Note that trunk, which already contained both of these features, is unaffected by this problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-06res_pjsip_mwi: Change warning to noticeGeorge Joseph
When res_pjsip loads and an endpoint auto-subscribes a mailbox for mwi, if a contact hasn't registered yet, res_pjsip_mwi spits out a warning. This is a perfectly normal situation though and doesn't require something as serious as a warning. It's also self correcting. The device will start getting mwi as soon as it registers. This patch changes the warning to a notice. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4314/ ........ Merged revisions 430227 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-06bridge_native_rtp: Change local/remote message from debug/2 to verb/4George Joseph
Change the "Locally bridged"/"Remotely bridged" messages from dbg/2 to verb/4. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4300/ ........ Merged revisions 430225 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430226 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-06outbound_registration: Add 'pjsip send register' and update 'send unregister'George Joseph
The current behavior of 'pjsip send unregister' is to send the unregister (REGISTER with 0 exp) but let the next scheduled register proceed normally. I don't think that's a good idea. If you unregister, it should stay unregistered until you decide to start registrations again. So this patch just adds a cancel_registration call to the current unregister_task to cancel the timer. Of course, now you need a way to start registration again so I've added a 'pjsip send register' command that unregisters and cancels any existing registration (the same as send unregister), then sends an immediate registration and starts the timer back up again. Both changes also ripple to AMI. There's a new PJSIPRegister command. There's no harm in calling either command repeatedly. They don't care about the actual state. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4301/ ........ Merged revisions 430223 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-06pjsip cli: Fix sorting of contacts for 'pjsip list contacts'George Joseph
For some reason I was using a hash container instead of a list to gather the contacts for 'pjsip list/show contacts' so even though I had a sort function, the output wasn't sorted. This patch just changes the hash container to a list container and the contacts now appear sorted in the CLI. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4305/ ........ Merged revisions 430221 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-05bridge: avoid leaking channel during blond transfer pt2Scott Griepentrog
A blond transfer to a failed destination, when followed by a recall attempt, lead to a leak of the reference to the destination channel. In addition to correcting the regression on the previous attempt (r429826) this fixes the leak and two additional reference leaks on failures of bridge_import. ASTERISK-24513 #close Review: https://reviewboard.asterisk.org/r/4302/ ........ Merged revisions 430199 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 430200 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-05pjsip: Document addition of 'PJSIP_AOR' and 'PJSIP_CONTACT' in CHANGES file.Joshua Colp
........ Merged revisions 430181 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-05pjsip: Add 'PJSIP_AOR' and 'PJSIP_CONTACT' dialplan functions.Joshua Colp
The PJSIP_AOR dialplan function allows inspection of configured AORs including what contacts are currently bound to them. The PJSIP_CONTACT dialplan function allows inspection of contacts in existence. These can include both externally added (by way of registration) or permanent ones. ASTERISK-24341 Reported by: xrobau Review: https://reviewboard.asterisk.org/r/4308/ ........ Merged revisions 430179 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430180 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-31rtp_engine: keep payload types in correct rangeScott Griepentrog
In r428708 additional codecs were added including a payload type of 128 which is outside of nominal range of 0-127. This change moves changes 128 to 96 to avoid causing a pjsip assertion when making a call to an endpoint configured with allow=all. ASTERISK-24367 #close Review: https://reviewboard.asterisk.org/r/4286/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-29PJSIP: Update transport method documentationKinsey Moore
This updates the documentation for the 'method' configuration option to be more verbose about the behaviors of values 'unspecified' and 'default'. They do exactly the same thing which is to select the default as defined by PJSIP which is currently TLSv1. Review: https://reviewboard.asterisk.org/r/4264/ ........ Merged revisions 430145 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-24app_queue: Update sample conf documenationKevin Harwell
Updated the queues.conf.sample file to explicitly state which channel queue variables are propagated to. ASTERISK-24267 Reported by: Mitch Claborn ........ Merged revisions 430126 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430127 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-24main/pbx.c: Fix double lock of contexts lock introduced by r429967Matthew Jordan
We only need to hold the context_merge_lock once. Locking it twice will make many other parts of Asterisk very sad. ASTERISK-24641 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-23pjsip_options: Fix continued qualifies after endpoint/aor deletionGeorge Joseph
If you remove an endpoint/aor from pjsip.conf then do a core reload, qualifies will continue even though the object are gone. This happens because nothing clears out the qualify tasks. This patch unschedules all existing qualify tasks before scheduling new ones on reload. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4290/ ........ Merged revisions 430064 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430067 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-23test_astobj2: Fix warning for missing trailing slash in categoryGeorge Joseph
This patch adds a trailing slash to the category for this test. No more warning. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4295/ ........ Merged revisions 430059 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-22DTMF atxfer: Setup recall channels as if the transferee initiated the call.Richard Mudgett
After the initial DTMF atxfer call attempt to the transfer target fails to answer during a blonde transfer, the recall callback channels do not get setup with information from the initial transferrer channel. As a result, the recall callback to the transferrer does not have callid, channel variables, datastores, accountcode, peeraccount, COLP, and CLID setup. A similar situation happens with the recall callback to the transfer target but it is less visible. The recall callback to the transfer target does not have callid, channel variables, datastores, accountcode, peeraccount, and COLP setup. * Added missing information to the recall callback channels before initiating the call. callid, channel variables, datastores, accountcode, peeraccount, COLP, and CLID * Set callid of the transferrer channel on the DTMF atxfer controller thread attended_transfer_monitor_thread(). * Added missing channel unlocks and props unref to off nominal paths in attended_transfer_properties_alloc(). ASTERISK-23841 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4259/ ........ Merged revisions 430034 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-22Fix compilation since the patch for ASTERISK-24363 went in.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-12-22queue_log: Post QUEUESTART entry when Asterisk fully boots.Richard Mudgett
The QUEUESTART log entry has historically acted like a fully booted event for the queue_log file. When the QUEUESTART entry was posted to the log was broken by the change made by ASTERISK-15863. * Made post the QUEUESTART queue_log entry when Asterisk fully boots. This restores the intent of that log entry and happens after realtime has had a chance to load. AST-1444 #close Reported by: Denis Martinez Review: https://reviewboard.asterisk.org/r/4282/ ........ Merged revisions 430009 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430010 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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