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2014-03-04Minor whitespace change to 'sip show peers' output.Sean Bright
(closes issue ASTERISK-23406) Reported by: ibercom Tested by: ibercom Patches: asterisk-11.patch uploaded by ibercom ........ Merged revisions 409472 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409473 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409474 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-03res_stasis_recording: Fix memory leak of the absolute name.Joshua Colp
........ Merged revisions 409422 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-03doxygen: Tweak the link back to ye olde Digium websiteMatthew Jordan
........ Merged revisions 409361 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409362 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409363 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409364 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-02Makefile: replace -O6 with -O3Tzafrir Cohen
-O6 is not a legal option of gcc. Unofficially gcc considers it to be equivalent of -O3. clang chalks on it, though. This commit sets the default optimization flag to be -O3, like gcc actually considered it. Review: https://reviewboard.asterisk.org/r/3280/ ........ Merged revisions 409308 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409344 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409346 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-01res_pjsip_session: Set options (100rel, timers) on incoming sessions.Joshua Colp
This change passes options to the UAS creation function. This in turn sets up 100rel and session timer properties on the incoming session. Reported by Julian Russell on asterisk-users mailing list. ........ Merged revisions 409287 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409288 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-01devicestate.c: Simplified some logic in _ast_device_state().Richard Mudgett
........ Merged revisions 409274 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-01stasis_cache.c: Remove some unnecessary RAII_VAR() usage.Richard Mudgett
........ Merged revisions 409272 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-28stasis.c: Misc code cleanups.Richard Mudgett
* Remove some unnecessary RAII_VAR() usage. * Made the struct stasis_subscription ao2 object use the ao2 lock instead of a redundant join_lock in the struct for ast_cond_wait(). * Removed locks on some ao2 objects that don't need the lock. * Made the topic pool entries container use the ao2 template functions. * Add some missing allocation failure checks. * Add missing cleanup in off nominal path of dispatch_message(). ........ Merged revisions 409270 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409271 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-28chan_sip: Add precautionary p->owner checks.Richard Mudgett
* Add precautionary p->owner checks in sip_hangup(), get_refer_info(), get_also_info(), and interpret_t38_parameters(). * Simplify some tangled logic in get_refer_info(), get_also_info(), and add_rpid(). * Removed some dead code in handle_request_invite(). (closes issue ASTERISK-23323) Reported by: Walter Doekes Patches: issueA23323-more_p_owner_checks-1.8.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-11.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-12.x.patch (license #5674) uploaded by wdoekes (modified) issueA23323-more_p_owner_checks-trunk.patch (license #5674) uploaded by wdoekes (modified) ........ Merged revisions 409207 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409255 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409256 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-28app_queue: Fix documented AMI event nameKinsey Moore
During the rewrite of AMI events to use the Stasis bus, the name of the QueueMemberPaused event was changed to QueueMemberPause. This corrects documentation to reflect that. ........ Merged revisions 409234 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-28chan_sip: Fix crash in ast_channel_hangupcause_set().Richard Mudgett
* Fix crash in ast_channel_hangupcause_set() because p->owner not checked before calling. Regression introduced by the fix for ASTERISK-22621. (closes issue ASTERISK-23135) Reported by: OK (issue ASTERISK-23323) Reported by: Walter Doekes ........ Merged revisions 409156 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409157 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409158 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-27Multiple revisions 409129-409130Jonathan Rose
........ r409129 | jrose | 2014-02-27 13:19:02 -0600 (Thu, 27 Feb 2014) | 15 lines res_rtp_asterisk: Fix checklist creating problems in ICE sessions Prior to this patch, local candidate lists including SRFLX would fail to start properly when building ICE candidate check lists. This patch fixes that problem by making sure that each SRFLX candidate is associated with the proper base address so that the check list can create matches properly. This patch was written by jcolp. The issue will be left open to await testing by the issue participants. (issue ASTERISK-23213) Reported by: Andrea Suisani Review: https://reviewboard.asterisk.org/r/3256/ ........ r409130 | jrose | 2014-02-27 13:38:10 -0600 (Thu, 27 Feb 2014) | 8 lines res_rtp_asterisk: correct build error from r409129 Accidentally placed a declaration below functional code (issue ASTERISK-23213) Reported by: Andrea Suisani Review: https://reviewboard.asterisk.org/r/3256/ ........ Merged revisions 409129-409130 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-27Fix memory stomping bug in astman.David M. Lee
This memset complained in dev mod on my Ubuntu box. The memset is both unnecessary and dangerous. At this point, m hasn't been initialized yet, so the memset will write off to whatever address happens to be on the stack at the time. ........ Merged revisions 409077 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409083 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409087 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409091 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-27res_fax: Comment out default settings from res_fax.conf.Corey Farrell
Comment out many settings in res_fax.conf.sample. The defaults are set in res_fax.c, so setting the same value in sample config does nothing but make the sample config more fragile. (closes issue ASTERISK-23231) Reported by: David Brillert Review: https://reviewboard.asterisk.org/r/3261/ ........ Merged revisions 409052 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409053 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409054 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-27res_pjsip_sdp_rtp: Apply packetization rules on inbound SDP handlingMatthew Jordan
The setting 'use_ptime' is supposed to tell Asterisk to honour the ptime attribute in an offer, preferring it to whatever packetization preferences have been set internally. Currently, however, something rather quirky will happen: (1) The SDP answer will be constructed in create_outgoing_sdp_stream. This will use the preferences from the endpoint, such that the 200 OK response will add the packetization preferences from the endpoint, and not what was offered. (2) When the 200 response is issued, apply_negotiated_sdp_stream is called. This will call apply_packetization, which will use the ptime attribute from the offer internally. We end up telling the offerer to use the internal ptime attribute, but we end up using the offered ptime attribute. Hilarity ensues. This patch modifies the behaviour by calling apply_packetization from negotiate_incoming_sdp_stream, which is called prior to create_outgoing_sdp_stream. This causes the format preferences on the session's media object to be set to the inbound ptime value (if 'use_ptime' is enabled), such that the construction of the answer gets the right value immediately. Review: https://reviewboard.asterisk.org/r/3244/ ........ Merged revisions 408999 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-26test_stasis.c: Misc cleanups.Richard Mudgett
* Make the consumer ao2 object use the ao2 lock instead of a redundant lock in the struct for ast_cond_wait(). * Fixed some curly brace placements. * Fixed use of malloc(0). malloc(0) has variant behavior. It is up to the implementation to determine if it returns NULL or a valid pointer that can be later passed to free(). ........ Merged revisions 408983 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408984 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-26pjsip: avoid edge case potential crash in answer()Scott Griepentrog
When accidentally compiling against a wrong version of pjsip headers with a different pjsip_inv_session size, the invite_tsx structure could be null in the answer() function. This led to a crash because it attempted to send the session response with an uninitialized packet pointer. This patch presets packet to null and adds a diagnostic log message to explain why the call fails. Review: https://reviewboard.asterisk.org/r/3267/ ........ Merged revisions 408970 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408971 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-26res_ari: Make some additional error responses consistent with the rest of ↵Joshua Colp
the system. This change makes some error cases use ast_ari_response_error to construct their error responses instead of manually doing it. This ensures they are consistent with the other error responses. Based on the original patch as done by Paul Belanger on the associated review. Review: https://reviewboard.asterisk.org/r/2904/ ........ Merged revisions 408957 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-26PJSIP: Fix some bad spacingKinsey Moore
........ Merged revisions 408943 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408944 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-26PJSIP: Prevent crash if channel has gone awayKinsey Moore
It is currently possible for an ast_sip_session to exist without an associated channel as is the case when a new invite is coming in or just after a hangup is issued on a chan_pjsip channel. Part of the attended transfer code assumed the channel would be non-NULL and used it as such causing a crash. This bug was exposed thanks to the attended transfer ARI test in the test suite. (closes issue ASTERISK-23287) Reported by: Matt Jordan ........ Merged revisions 408941 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-26Implement functions handling keypress, display icons and text for i2004 KEM ↵Igor Goncharovskiy
support. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-25res_pjsip_exten_state: Presence for digium phonesKevin Harwell
Added presence support for digium phones. Review: https://reviewboard.asterisk.org/r/3239/ ........ Merged revisions 408882 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-25res_pjsip_send_to_voicemail: transferring to voicemail for digium phonesKevin Harwell
Added the ability for transferring directly to voicemail on digium phones. Added a new module that checks for the presence of a custom header and/or diversion header within a sip REFER. If either is found and they specify a sending to voicemail action then variables are added to the channel allowing the user access to them in the dialplan. Dialplan can then be written that branches based upon these values allowing, for instace, for a single number to be used for dialing and/or accessing voicemail directly. Also fixed a problem where the PJSIP_HEADER function was allowing non pjsip channels through (checked to make sure it has the correct channel type before proceeding). Review: https://reviewboard.asterisk.org/r/3245/ ........ Merged revisions 408880 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408881 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-25configs/voicemail.conf.sample - Make mailcmd sample text more explicitRusty Newton
Made the wording a bit more explicit. Didn't really change the meaning. ........ Merged revisions 408876 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408877 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408878 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-22main: Initialize dialplan providing core components prior to module pre-loadMatthew Jordan
It is possible to pre-load pbx_config. As a result, pbx_config - which will load and parse the dialplan - will attempt to use various dialplan components, such as device state providers and presence state providers, prior to them being initialized by the core. This would lead to a crash, as the components had not created their Stasis cache entries. This patch moves a number of core component initializations before the module pre-load. This guarantees that if someone does pre-load pbx_config - or other pbx modules - that the Stasis caches for the various core components are created. (closes issue ASTERISK-23320) Reported by: xrobau (closes issue ASTERISK-23265) Reported by: Andrew Nagy Tested by: Andrew Nagy, Rusty Newton ........ Merged revisions 408855 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-22ignore AST_CONTROL_PVT_CAUSE_CODE without any messagesAlexandr Anikin
(closes issue ASTERISK-23336) Reported by: Alexander Semych ........ Merged revisions 408838 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408839 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408840 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-22Remove extra defines of AST_PBX_MAX_STACK.Corey Farrell
* Ensure AST_PBX_MAX_STACK is only defined in extconf.h and pbx.h. * Fix incorrect function parameters in utils/extconf.c. (closes issue ASTERISK-23141) Reported by: Maxim Review: https://reviewboard.asterisk.org/r/3241/ ........ Merged revisions 408785 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408786 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408787 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408788 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21rtp_engine: Dynamic payload change in rtp mapping not supportedKevin Harwell
Asterisk didn't support the dynamic payload change in rtp mapping in the 200 OK response. Scenario: Asterisk sends the INVITE proposing alaw and telephone-event, it proposes rtpmap:101 for telephone-event. Peer responds with 2xx, it answers with alaw and telephone-event also, but it proposes a different rtpmap number (rtpmap:103) for telephone-event. Expected Behaviour: Asterisk should honour the rtpmapping in the response and send DTMF packets using 103 as payload type for DTMF. Actual Behaviour: Asterisk sends DTMF packets using payload type 101. With this patch asterisk now supports changes that can occur in the rtp mapping in the response. (closes issue ASTERISK-23279) Reported by: NITESH BANSAL Review: https://reviewboard.asterisk.org/r/3225/ Patches: dynamic_payload_change.patch uploaded by nbansal (license 6418) ........ Merged revisions 408729 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408730 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21manager: Fix AMI Status action of a single channel.Richard Mudgett
Fixed use of uninitialized ao2 container iterator in an off-nominal condition. Either a memory allocation error or the requested channel is an internal channel not exposed to the outside. ........ Merged revisions 408715 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408723 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21json: Fix off-nominal json ref counting issues.Richard Mudgett
* Fixed off-nominal json ref counting issue with using the following API calls: ast_json_object_set() and ast_json_array_append(). * Fixed off-nominal error reporting in ast_ari_endpoints_list(). * Fixed some miscellaneous off-nominal json ref counting issues in report_receive_fax_status() and dial_to_json(). ........ Merged revisions 408713 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21json: Fix json API wrapper code for json library versions earlier than 2.3.0.Richard Mudgett
* Fixed json ref counting issue with json API wrapper code for ast_json_object_update_existing() and ast_json_object_update_missing() when the json library is earlier than version 2.3.0. ........ Merged revisions 408711 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408712 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21chan_sip: prevent add_route from adding empty header.Corey Farrell
Fix regression caused by ASTERISK-22582. Empty Route headers were added when the route had a single strict hop. (closes issue ASTERISK-23306) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3236/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21rtp_engine: Output mixup in ${CHANNEL(rtpqos,audio,all)}Kevin Harwell
Fixed the output of CHANNEL(rtpqos,audio,all) to use txjitter instead of rxjitter. (closes issue ASTERISK-23261) Reported by: rsw686 Patches: rtpqos.patch uploaded by rsw686 (license 5887) ........ Merged revisions 408646 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408647 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408649 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21channel.c: MOH is not working for transferee after attended transferKevin Harwell
Updated the code to check to see if MOH is playing on the transferor and if so then start it on the channel that replaces it during a masquerade. Example scenario of the problem: Alice calls Bob and then Bob begins the attended transfer process into a queue. Upon going on hold Alice hears music and so does Bob once he is in the queue. Bob then transfers Alice into the queue and then music for Alice stops even though she should be hearing it since has now replaced Bob in the queue. The problem that was occurring is that once the channel was masqueraded the app (queues, confbridge, etc...) had no way of knowing that the channel had just been swapped out thus it did not start music for the present channel. Credit to Olle Johansson for pointing me in the right direction on this issue. (closes issue ASTERISK-19499) Reported by: Timo Teräs Review: https://reviewboard.asterisk.org/r/3226/ ........ Merged revisions 408642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408643 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408644 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408645 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21Fix type of roundTripDelay variablesAlexandr Anikin
........ Merged revisions 408589 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408590 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408591 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408592 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21app_chanspy: Documentation Update To Clarify "x" OptionMichael L. Young
When using the "x" option (specify a DTMF digit to exit the application), it is not obvious in the documentation that this only works when spying on a channel. If a channel being used to spy on other channels is waiting to connect to a channel or is no longer attached to a channel, the DTMF is ignored. As noted on the issue tracker, since there are workarounds available and this is a rarely used option we are opting for a documentation change here. (closes issue ASTERISK-22661) Reported by: Chris Hillman Patches: asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2990/ ........ Merged revisions 408536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408537 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408538 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-20pjsip_cli: Add pjsip commands 'show registrations' and 'show contacts'.George Joseph
Added 'show registrations' and 'show contacts' to pjsip cli to make things a little more consistent. The output is exactly the same as the list command. Just needed to add entries to their respective ast_cli_entry structures. (closes issue ASTERISK-23275) Review: http://reviewboard.asterisk.org/r/3210/ ........ Merged revisions 408522 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-20pjsip_cli: Fix memory leak in ast_sip_cli_print_sorcery_objectset.George Joseph
Fixed memory leaks in ast_sip_cli_print_sorcery_objectset and ast_variable_list_sort. (closes issue ASTERISK-23266) Review: http://reviewboard.asterisk.org/r/3200/ ........ Merged revisions 408520 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408521 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-20sorcery: Create sorcery instance registry.George Joseph
In order to retrieve an arbitrary sorcery instance from a dialplan function (or any place else) there needs to be a registry of sorcery instances. ast_sorcery_init now creates a hashtab as a registry. ast_sorcery_open now checks the hashtab for an existing sorcery instance matching the caller's module name. If it finds one, it bumps the refcount and returns it. If not, it creates a new sorcery instance, adds it to the hashtab, then returns it. ast_sorcery_retrieve_by_module_name is a new function that does a hashtab lookup by module name. It can be called by the future dialplan function. res_pjsip/config_system needed a small change to share the main res_pjsip sorcery instance. tests/test_sorcery was updated to include a test for the registry. (closes issue ASTERISK-22537) Review: http://reviewboard.asterisk.org/r/3184/ ........ Merged revisions 408518 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-20res_pjsip: Update documentation for 'use_avpf' optionMatthew Jordan
When 'use_avpf' is set to True, inbound offers must use the AVPF/SAVPF RTP profile. However, when 'use_avpf' is set to False, Asterisk will accept both AVP/SAVP or AVPF/SAVPF RTP profiles in inbound offers. The documentation previously implied that Asterisk would reject AVPF/SAVPF if 'use_avpf' was set to False and a UA offered said profile in an INVITE request. ........ Merged revisions 408502 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-20apps/app_queue - Fix incorrect Macro parameter documentationRusty Newton
Macro is executed on the called channel, not the calling channel. (closes issue ASTERISK-23069) Reported By: Bryan Anderson ........ Merged revisions 408447 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408448 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408449 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-19config: Add file size and nanosecond resolution fields to the cached ↵Richard Mudgett
modified config file information. Repeatedly modifying config files and reloading too fast sometimes fails to reload the configuration because the cached modification timestamp has one second resolution. * Added file size and nanosecond resolution fields to the cached config file modification timestamp information. Now if the file size changes or the file system supports nanosecond resolution the modified file has a better chance of being detected for reload. * Added a missing unlock in an off-nominal code path. (closes issue AST-1303) Review: https://reviewboard.asterisk.org/r/3235/ ........ Merged revisions 408387 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408388 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408389 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408390 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-19res_sorcery_astdb.c: Fix regex handling and keep simple prefix matching ↵Richard Mudgett
performance. The sorcery astDB wizzard does not handle regex correctly if the pattern begins with an anchor character. This patch attempts to convert the anchored regex pattern to a prefix pattern supported by astDB for performance reasons. If it is not able to convert the pattern it falls back to getting all astDB members of the family and doing a normal regex pattern matching on the retrieved records. Review: https://reviewboard.asterisk.org/r/3161/ ........ Merged revisions 408385 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-19process receiveAndTransmit user input remote caps instead of receive onlyAlexandr Anikin
send receiveAndTransmit user input our caps instead of receive only ........ Merged revisions 408328 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408330 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408331 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-19Allow different socket and signalling ip on h.323 connection if gk mode is ↵Alexandr Anikin
active Reported by: Gabriele Odone Patches: ASTERISK-22738-1.patch Tested by: Gabriele Odone (closes issue ASTERISK-22738) ........ Merged revisions 408312 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408314 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408315 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-18alembic: Add svn:ignore *.pyc to directories and svn:executable to *.py files.Richard Mudgett
........ Merged revisions 408297 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-17Store SIP User-Agent information in contacts.Mark Michelson
When an endpoint sends a REGISTER request to Asterisk, we now will associate the User-Agent header with all contacts that were bound in that REGISTER request. ........ Merged revisions 408270 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-16pbx: Handle a completely empty dialplan during a context mergeMatthew Jordan
It is highly unlikely, but - at least in Asterisk 12 - theoretically possible to load Asterisk with no dialplan whatsoever. If that occurs, and some other module (that is not a pbx module) attempts to merge its contexts into the dialplan, the existing merge routine will crash. This is because it is not insane, and rightly believes that you provided some sort of dialplan, somewhere. This patch will gracefully merge the contexts in such a case. Note that this is highly unlikely to occur in 1.8/11, as features will most likely provide some dialplan via parking. However, in Asterisk 12, parking is now provided by res_parking, and hence may create its dialplan later. (closes issue ASTERISK-23297) Reported by: CJ Oster Review: https://reviewboard.asterisk.org/r/3222 ........ Merged revisions 408200 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408201 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408220 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-16buildsystem: Unbreak the build (infloop) on Asterisk 11+Matthew Jordan
Apparently r408084 ( https://reviewboard.asterisk.org/r/3212/ ) broke the build. This patch fixes it by ignoring the .lastclean dependencies if the MENUSELECT_EMBED variable is not defined. patches: tmp.diff uploaded by wdoekes (License 5674) Review: https://reviewboard.asterisk.org/r/3228/ ........ Merged revisions 408193 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408194 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408199 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-14ARI: correct upper/lower case URI discrepanciesScott Griepentrog
URI's are supposed to be case sensitive and all lower case. In practice some portions of URI's in ARI are case insensitive and others are not, such as TECH, which in one instance would match a lower case name and in another would not. In this patch, the ast_endpoint_lastest_snapshot() function is modified to change the TECH portion to full upper case before lookup. This resolves the discrepancy noted by the reporter. However I chose to avoid forcing the /ari prefix of the URI's to be lower case for now. Except for the two cases here, all URI's should be lower case, unless they are part of a resource name or id. Review: https://reviewboard.asterisk.org/r/3211/ Reported by: Zane Conkle (closes issue ASTERISK-23125) ........ Merged revisions 408140 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408141 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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