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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
2014-02-14format.c: correct possible null pointer dereferenceScott Griepentrog
In ast_format_sdp_parse and ast_format_sdp_generate the check checks for a valid interface and function were potentially confusing, and hid an error in the test of the presence of the function that is called later. This patch clears up and corrects the test. Review: https://reviewboard.asterisk.org/r/3208/ (closes issue ASTERISK-23098) Reported by: marcelloceschia Patches: main_format.patch uploaded by marcelloceschia (license 6036) ASTERISK-23098.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 408137 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408138 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408139 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-14buildsystem: Don't force main to depend on everything else.Walter Doekes
Directory 'main' only needs to depend on embedded modules. If no module embedding is selected, the dependency is dropped. Review: https://reviewboard.asterisk.org/r/3212/ ........ Merged revisions 408083 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408084 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408085 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-14chnan_sip: Set SIP_DEFER_BYE_ON_TRANSFER prior to calling bridge blind transferMatthew Jordan
This patch moves setting SIP_DEFER_BY_ON_TRANSFER prior to calling ast_bridge_transfer_blind. This prevents a BYE from being sent prior to the NOTIFY request that informs the transferor if the transfer succeeded or failed. This patch also clears said flag from the off nominal NOTIFY paths in the local_attended_transfer code, as once we've sent the NOTIFY request it is safe to send by the BYE request. This was caught by the blind-transfer-accountcode test in the Asterisk Test Suite. (closes issue ASTERISK-23290) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3214/ ........ Merged revisions 408069 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-14install_subst: helper script for installing with path substitutionTzafrir Cohen
A helper script to copy a source file substituting any __ASTERISK_<foo>_DIR__ with the content of $AST<foo>DIR. Review: https://reviewboard.asterisk.org/r/3202/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408059 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-13Remove all PJSIP MWI-specific use from our MWI code.Mark Michelson
PJSIP has built-in MWI code that could be useful to some degree, but our utilization of the API actually made our code a bit more cluttered since we had to have special cases peppered throughout. With this change, we move to using the pjsip_evsub API instead, which streamlines the code by removing special cases. Review: https://reviewboard.asterisk.org/r/3205 ........ Merged revisions 408005 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-13Fix crash in AMI PJSIPShowEndpoint action.Mark Michelson
If an AOR has no permanent contacts, then the permanent_contacts container is never allocated. This makes the code safe in the face of NULLs. I also changed the variable that counts contacts from "num" to "total_contacts" since there are now two variables that are indicate numbers of things. ........ Merged revisions 407988 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-13Logger: Add dynamic logger channelsKinsey Moore
This adds the ability to dynamically add and remove logger channels from Asterisk via the CLI. (closes issue AST-1150) Review: https://reviewboard.asterisk.org/r/3185/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-12realtime: Fix ast_update2_realtime() on raspberry pi.Walter Doekes
The old code depended on undefined va_arg behaviour: calling a function twice with the same va_list parameter and expecting it to continue where it left off. The changed code behaves like the manpage says it should. Also added a bunch of early returns to trap errors (e.g. OOM) instead of crashing. The problem was found by Julian Lyndon-Smith. The deviant behaviour on the raspberry PI also uncovered another bug (fixed in r407875) in the res_config_pgsql.so driver. Reported by: jmls Tested by: jmls Review: https://reviewboard.asterisk.org/r/3201/ ........ Merged revisions 407968 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407970 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-11scheduler: Remove hashtab usage.Joshua Colp
This is a first stab at tweaking the performance profile of the scheduler. Removing the hashtab usage removes an extra memory allocation when scheduling something and makes it so rescheduling does not incur any memory allocation at all. Review: https://reviewboard.asterisk.org/r/3199/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-11ari/resource_channels: Add channel variables earlier in the creation processMatthew Jordan
This patch tweaks the behaviour of POST /channels with channel variables such that the variables are passed into the pbx.c routines that perform the origination. This allows the variables to be assigned to the newly created channels immediately upon their construction, as opposed to be assigned after the originate has completed. The upshot of this is that the variables are available on the channels if they execute in the dialplan, as opposed to only being available once the channels are answered. Review: https://reviewboard.asterisk.org/r/3183/ ........ Merged revisions 407937 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-10chan_sip: Isolate code that manages struct sip_route.Corey Farrell
* Move route code to sip/route.c + sip/include/route.h * Rename functions to sip_route_* * Replace ad-hoc list code with macro's from linkedlists.h * Create sip_route_process_header() to processes Path and Record-Route headers (previously done with different code in build_route and build_path) * Add use of const where possible * Move struct uriparams, struct contact and contactliststruct from sip.h to reqresp_parser.h. sip/route.c uses reqresp_parser.h but not sip.h, this was a problem. These moved declares are not used outside of reqresp_parser. * While modifying reqprep() the lack of {} caused me trouble. I added them. * Code outside route.c treats sip_route as an opaque structure, using macro's or procedures for all access. (closes issue ASTERISK-22582) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3173/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-10res_config_pgsql: Fix ast_update2_realtime calls.Walter Doekes
Fix so multiple updates from a single call works (add missing ','). Remove bogus ast_free's that weren't supposed to be there. Moved a few spaces for readability. Review: https://reviewboard.asterisk.org/r/3194/ ........ Merged revisions 407873 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407874 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407875 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407876 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-10ConfBridge: Correct prompt playback targetKinsey Moore
Currently, when the first marked user enters the conference that contains waitmarked users, a prompt is played indicating that the user is being placed into the conference. Unfortunately, this prompt is played to the marked user and not the waitmarked users which is not very helpful. This patch changes that behavior to play a prompt stating "The conference will now begin" to the entire conference after adding and unmuting the waitmarked users since the design of confbridge is not conducive to playing a prompt to a subset of users in a conference in an asynchronous manner. (closes issue PQ-1396) Review: https://reviewboard.asterisk.org/r/3155/ Reported by: Steve Pitts ........ Merged revisions 407857 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407858 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-07chan_iax2: Add some more iaxs[] NULL checks to a routine already full of them.Richard Mudgett
........ Merged revisions 407764 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407765 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407766 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-07security_events: Fix assertion failure in dev-mode on optional IE parsingMatthew Jordan
When formatting an optional IE, the value is, of course, optional. As such, it is entirely appropriate for ast_json_object_get to return NULL. If that occurs, we now simply skip the IE that was requested, as it was not provided by the entity that raised the event. Thanks to George Joseph (gtjoseph) for catching this and reporting it in #asterisk-dev ........ Merged revisions 407750 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-07timing: Improve performance for most timing implementations.Joshua Colp
This change allows timing implementation data to be stored directly on the timer itself thus removing the requirement for many implementations to do a container lookup for the same information. This means that API calls into timing implementations can directly access the information they need instead of having to find it. Review: https://reviewboard.asterisk.org/r/3175/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-07funcs/func_cdr: Handle empty time values when extracting parsed valuesMatthew Jordan
When extracting timestamps that are parsed, time stamp values that are not set (time values of 0.000000) should not actually result in a parsed string. The value should be skipped, and the result of the CDR function should be an empty string. Prior to this patch, the result was fed to the time formatting, which would result in an output of a date/time in 1969. ........ Merged revisions 407747 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-07chan_iax2: Block unnecessary control frames to/from the wire.Richard Mudgett
Establishing an IAX2 call between Asterisk v1.4 and v1.8 (or later) results in an unexpected call disconnect. The problem happens because newer values in the enum ast_control_frame_type are not consistent between the branch versions of Asterisk. For example: 1) v1.4 calls v1.8 (or later) using IAX2 2) v1.8 answers and sends a connected line update control frame. (on v1.8 AST_CONTROL_CONNECTED_LINE = 22) 3) v1.4 receives the control frame as an end-of-q (on v1.4 AST_CONTROL_END_OF_Q = 22) 4) v1.4 disconnects the call once the receive queue becomes empty. Several things are done by this patch to fix the problem and attempt to prevent it from happening again in the future: * Added a warning at the definition of enum ast_control_frame_type about how to add new control frame values. * Made block sending and receiving control frames that have no reason to go over the wire. * Extended the connectedline iax.conf parameter to also include the redirecting information updates. * Updated the connectedline iax.conf parameter documentation to include a notice that the parameter must be "no" when the peer is an Asterisk v1.4 instance. (closes issue AST-1302) Review: https://reviewboard.asterisk.org/r/3174/ ........ Merged revisions 407678 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407727 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407729 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-07security_events: Fix error caused by DTD validation errorMatthew Jordan
The appdocsxml.dtd specifies that a "required" attribute in a parameter may have a value of yes, no, true, or false. On some systems, specifying "False" instead of "false" would cause a validation error. This patch fixes the casing to explicitly match the DTD. ........ Merged revisions 407676 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-07indications.conf: add stutter tone; end properlyTzafrir Cohen
* If the "stutter" (voicemail indication) tone is indeed a stutter tone, and it ends with a constant tone, make sure that it is the dial tone. This was done for India (in), Mexico (mx) and the Philippines (ph). * If no "stutter" tone exists for a country, provide one. This was done for Spain (es), Malaysia (my) and Venezuela (ve). Review: https://reviewboard.asterisk.org/r/3158/ ........ Merged revisions 407622 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407623 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407624 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-06security_events: Add AMI documentation; output optional fieldsMatthew Jordan
This patch adds documentation for the Security Events that are emited over AMI. It also notes these events in the UPGRADE/CHANGES file. ........ Merged revisions 407589 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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