Release Summary asterisk-12.7.0-rc1 Date: 2014-11-03 ---------------------------------------------------------------------- Table of Contents 1. Summary 2. Contributors 3. Closed Issues 4. Other Changes 5. Diffstat ---------------------------------------------------------------------- Summary [Back to Top] This release includes only bug fixes. The changes included were made only to address problems that have been identified in this release series. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous release series are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series. The data in this summary reflects changes that have been made since the previous release, asterisk-12.6.0. ---------------------------------------------------------------------- Contributors [Back to Top] This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were closed by commits that went into this release. Coders Testers Reporters 25 mjordan 1 abelbeck 11 coreyfarrell 14 coreyfarrell 1 Dmitry Melekhov 7 mjordan 14 rmudgett 1 Etienne Lessard 3 rmudgett 10 file 1 gtjoseph 3 tzafrir 10 gtjoseph 1 Nick Adams 2 dafi 8 wdoekes 1 opsmonitor 2 jcolp 6 kmoore 1 Paolo Compagnini 2 kharwell 3 kharwell 1 Yuriy Gorlichenko 2 marquis 3 mdavenport 2 sharky 3 Torrey Searle 2 tsearle 2 igorg 1 abelbeck 2 jcolp 1 boroda 2 jrose 1 damianivereigh 2 Nitesh Bansal 1 dhanapathy 1 abelbeck 1 hexanol 1 Damian Ivereigh 1 ibercom 1 ibercom 1 jbigelow 1 jbigelow 1 jrose 1 Jeremy Laine 1 jvanvleet 1 Jeremy LainA(c) 1 laimbock 1 may 1 looserouting 1 Michael Myles 1 mclaborn 1 oej 1 mmichelson 1 Peter Katzmann 1 mores 1 sgriepentrog 1 Narkov 1 tzafrir 1 nbansal 1 oej 1 pk16208 1 rogger.padilla 1 rustamxp 1 sgriepentrog 1 shaneblaser 1 slesru 1 snuffy 1 spitts 1 wdoekes 1 xdrive ---------------------------------------------------------------------- Closed Issues [Back to Top] This is a list of all issues from the issue tracker that were closed by changes that went into this release. Category: Addons/chan_ooh323 ASTERISK-24393: rtptimeout=0 doesn't disable rtptimeout Revision: 425589 Reporter: slesru Testers: Dmitry Melekhov Coders: may Category: Applications/app_mixmonitor ASTERISK-24195: bridge_native_rtp: Removing mixmonitor from a native RTP capable smart bridge doesn't cause the bridge to resume being a native rtp bridge Revision: 424506 Reporter: jrose Coders: rmudgett Category: Applications/app_queue ASTERISK-24454: app_queue: ao2_iterator not destroyed, causing leak Revision: 426260 Reporter: coreyfarrell Coders: coreyfarrell ASTERISK-24466: app_queue: fix a couple leaks to struct call_queue Revision: 426806 Reporter: coreyfarrell Coders: coreyfarrell Category: Applications/app_voicemail ASTERISK-24190: IMAP voicemail causes segfault Revision: 426696 Reporter: Narkov Testers: Nick Adams Coders: wdoekes ASTERISK-24476: main/app.c / app_voicemail: ast_writestream leaks Revision: 427025 Reporter: coreyfarrell Coders: coreyfarrell Category: Applications/app_voicemail/IMAP ASTERISK-24190: IMAP voicemail causes segfault Revision: 426696 Reporter: Narkov Testers: Nick Adams Coders: wdoekes Category: Bridges/bridge_native_rtp ASTERISK-24195: bridge_native_rtp: Removing mixmonitor from a native RTP capable smart bridge doesn't cause the bridge to resume being a native rtp bridge Revision: 424506 Reporter: jrose Coders: rmudgett ASTERISK-24327: bridge_native_rtp: Smart bridge operation to softmix sometimes fails to properly re-INVITE remotely bridged participants Revision: 425760 Reporter: mjordan Coders: mjordan Category: CDR/General ASTERISK-24394: CDR: FRACK with PJSIP directed pickup. Revision: 425430 Reporter: rmudgett Coders: rmudgett ASTERISK-24426: CDR Batch mode: size used as time value after first expire Revision: 425735 Reporter: shaneblaser Coders: mjordan Category: Channels/General ASTERISK-24415: Missing AMI VarSet events when channels inherit variables. Revision: 425782 Reporter: rmudgett Coders: rmudgett Category: Channels/chan_local ASTERISK-24415: Missing AMI VarSet events when channels inherit variables. Revision: 425782 Reporter: rmudgett Coders: rmudgett Category: Channels/chan_motif ASTERISK-24384: chan_motif: format capabilities leak on module load error Revision: 424551 Reporter: coreyfarrell Coders: coreyfarrell Category: Channels/chan_pjsip ASTERISK-24356: PJSIP: Directed pickup causes deadlock Revision: 424471 Reporter: rmudgett Coders: rmudgett ASTERISK-24382: chan_pjsip: Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel results in an invalid reference of a channel pvt and a FRACK Revision: 424621 Reporter: mjordan Coders: mjordan Category: Channels/chan_sip/General ASTERISK-15879: [patch] Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak Revision: 425070 Reporter: tsearle Coders: Torrey Searle, Nitesh Bansal ASTERISK-20784: Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak Revision: 425070 Reporter: nbansal Coders: Torrey Searle, Nitesh Bansal ASTERISK-22791: asterisk sends Re-INVITE after receiving a BYE Revision: 425298 Reporter: looserouting Testers: Paolo Compagnini Coders: wdoekes ASTERISK-22945: [patch] Memory leaks in chan_sip.c with realtime peers Revision: 424178 Reporter: ibercom Testers: Yuriy Gorlichenko Coders: ibercom ASTERISK-24063: [patch]Asterisk does not respect outbound proxy when sending qualify requests Revision: 425820 Reporter: damianivereigh Coders: Damian Ivereigh ASTERISK-24321: SIP deadlock when running automated queues tests Revision: 425503 Reporter: spitts Coders: jrose ASTERISK-24335: [PATCH] Asterisk incorrectly responds 503 to INVITE retransmissions of rejected calls Revision: 423722 Reporter: tsearle Coders: Torrey Searle ASTERISK-24385: chan_sip: process_sdp leaks on an error path Revision: 424575 Reporter: coreyfarrell Coders: coreyfarrell Category: Channels/chan_sip/Interoperability ASTERISK-21721: SIP Failed to parse multiple Supported: headers Revision: 426596 Reporter: oej Coders: oej Category: Channels/chan_sip/T.38 ASTERISK-22791: asterisk sends Re-INVITE after receiving a BYE Revision: 425298 Reporter: looserouting Testers: Paolo Compagnini Coders: wdoekes Category: Channels/chan_unistim ASTERISK-23846: Unistim multilines. Loss of voice after second call drops (on a second line). Revision: 425668 Reporter: rustamxp Coders: igorg ASTERISK-24304: asterisk crashing randomly because of unistim channel Revision: 426667 Reporter: dhanapathy Coders: igorg Category: Contrib/General ASTERISK-23781: outgoing missing as enum from contrib/ast-db-manage/config Revision: 424372 Reporter: mores Coders: jrose ASTERISK-24011: [patch]safe_asterisk tries to set ulimit -n too high on linux systems with lots of RAM Revision: 424879 Reporter: xdrive Coders: Michael Myles ASTERISK-24432: Install refcounter.py when REF_DEBUG is enabled Revision: 426832 Reporter: coreyfarrell Coders: coreyfarrell Category: Core/Bridging ASTERISK-24437: Review implementation of ast_bridge_impart for leaks and document proper usage Revision: 426431 Reporter: sgriepentrog Coders: mjordan Category: Core/BuildSystem ASTERISK-13797: [patch] relax badshell tilde test Revision: 425293 Reporter: tzafrir Coders: wdoekes Category: Core/CallerID ASTERISK-24406: Some caller ID strings are parsed differently since 11.13.0 Revision: 425154 Reporter: hexanol Testers: Etienne Lessard Coders: kmoore Category: Core/General ASTERISK-24476: main/app.c / app_voicemail: ast_writestream leaks Revision: 427025 Reporter: coreyfarrell Coders: coreyfarrell Category: Core/ManagerInterface ASTERISK-24262: AMI CoreShowChannel missing several output fields and event documentation Revision: 424423 Reporter: mclaborn Coders: kmoore ASTERISK-24354: AMI sendMessage closes AMI connection on error Revision: 424691 Reporter: pk16208 Coders: Peter Katzmann ASTERISK-24378: Release AMI connections on shutdown Revision: 424579 Reporter: coreyfarrell Coders: coreyfarrell ASTERISK-24430: missing letter "p" in word response in OriginateResponse event documentation Revision: 426367 Reporter: dafi Coders: mjordan ASTERISK-24453: manager: acl_change_sub leaks Revision: 426524 Reporter: coreyfarrell Coders: coreyfarrell Category: Core/Sorcery ASTERISK-24312: SIGABRT when improperly configured realtime pjsip Revision: 425690 Reporter: dafi Coders: kmoore Category: Documentation ASTERISK-23768: [patch] Asterisk man page contains a (new) unquoted minus sign Revision: 423917 Reporter: sharky Coders: Jeremy LainA(c) ASTERISK-24122: Documentaton for res_pjsip option use_avpf needs to be fixed Revision: 425868 Reporter: jvanvleet Coders: mjordan ASTERISK-24262: AMI CoreShowChannel missing several output fields and event documentation Revision: 424423 Reporter: mclaborn Coders: kmoore ASTERISK-24300: API docs don't conform to stated Swagger version Revision: 423620 Reporter: marquis Coders: mjordan ASTERISK-24430: missing letter "p" in word response in OriginateResponse event documentation Revision: 426367 Reporter: dafi Coders: mjordan Category: General ASTERISK-20567: bashism in autosupport Revision: 424125 Reporter: tzafrir Coders: wdoekes ASTERISK-24321: SIP deadlock when running automated queues tests Revision: 425503 Reporter: spitts Coders: jrose Category: Resources/res_ari ASTERISK-24339: Swagger API Docs have incorrect basePath Revision: 423617 Reporter: marquis Coders: mjordan Category: Resources/res_calendar_ews ASTERISK-24325: res_calendar_ews: cannot be used with neon 0.30 Revision: 425288 Reporter: tzafrir Coders: wdoekes Category: Resources/res_fax ASTERISK-22791: asterisk sends Re-INVITE after receiving a BYE Revision: 425298 Reporter: looserouting Testers: Paolo Compagnini Coders: wdoekes ASTERISK-24357: [fax] Out of bounds error in update_modem_bits Revision: 423987 Reporter: sharky Coders: Jeremy Laine ASTERISK-24392: res_fax: fax gateway sessions leak Revision: 425458 Reporter: coreyfarrell Coders: coreyfarrell ASTERISK-24457: res_fax: fax gateway frames leak Revision: 426528 Reporter: coreyfarrell Coders: coreyfarrell Category: Resources/res_fax_spandsp ASTERISK-18923: res_fax_spandsp usage counter is wrong Revision: 425411 Reporter: boroda Coders: coreyfarrell Category: Resources/res_hep ASTERISK-24362: res_hep leaks reference to configuration Revision: 424312 Reporter: coreyfarrell Coders: coreyfarrell Category: Resources/res_hep_pjsip ASTERISK-24369: res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations Revision: 424618 Reporter: mjordan Coders: mjordan Category: Resources/res_jabber ASTERISK-24425: [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) Revision: 425987 Reporter: abelbeck Testers: abelbeck, opsmonitor, gtjoseph Coders: abelbeck, mjordan Category: Resources/res_pjsip ASTERISK-24122: Documentaton for res_pjsip option use_avpf needs to be fixed Revision: 425868 Reporter: jvanvleet Coders: mjordan ASTERISK-24199: 'ALL' is specified in pjsip.conf.sample for TLS cipher but it is not valid Revision: 424393 Reporter: jcolp Coders: rmudgett ASTERISK-24295: crash: creating out of dialog OPTIONS request crashes Revision: 423866 Reporter: rogger.padilla Coders: rmudgett ASTERISK-24312: SIGABRT when improperly configured realtime pjsip Revision: 425690 Reporter: dafi Coders: kmoore ASTERISK-24350: PJSIP shows commands prints unneeded headers Revision: 424128 Reporter: snuffy Coders: gtjoseph ASTERISK-24369: res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations Revision: 424618 Reporter: mjordan Coders: mjordan ASTERISK-24370: res_pjsip/pjsip_options: OPTIONS request sent to Asterisk with no user in request is always 404'd Revision: 424624 Reporter: mjordan Coders: mjordan ASTERISK-24387: res_pjsip: rport sent from UAS MUST include the port that the UAC sent the request on Revision: 425131 Reporter: mjordan Coders: jcolp ASTERISK-24462: res_pjsip: Stale qualify statistics after disablementation Revision: 426755 Reporter: kharwell Coders: kharwell Category: Resources/res_pjsip_logger ASTERISK-24369: res_pjsip: Large message on reliable transport can cause empty messages to be passed from the PJSIP stack up, causing crashes in multiple locations Revision: 424618 Reporter: mjordan Coders: mjordan Category: Resources/res_pjsip_outbound_registration ASTERISK-24398: Initialize auth_rejection_permanent on client state to the configuration parameter value Revision: 424730 Reporter: mjordan Coders: mjordan ASTERISK-24411: [patch] Status of outbound registration is not changed upon unregistering. Revision: 426923 Reporter: jbigelow Coders: jbigelow Category: Resources/res_pjsip_sdp_rtp ASTERISK-24381: res_pjsip_sdp_rtp: Declined media streams are interpreted, leading to erroneous 488 rejections Revision: 425868 Reporter: mjordan Coders: mjordan Category: Resources/res_rtp_asterisk ASTERISK-24326: res_rtp_asterisk: ICE-TCP candidates are incorrectly attempted Revision: 424853 Reporter: jcolp Coders: jcolp ASTERISK-24383: res_rtp_asterisk: Crash if no candidates received for component Revision: 425030 Reporter: kharwell Coders: kharwell Category: Resources/res_srtp ASTERISK-24436: Missing header in res/res_srtp.c when compiling against libsrtp-1.5.0 Revision: 426142 Reporter: laimbock Coders: mjordan Category: Resources/res_xmpp ASTERISK-24425: [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) Revision: 425987 Reporter: abelbeck Testers: abelbeck, opsmonitor, gtjoseph Coders: abelbeck, mjordan ---------------------------------------------------------------------- Commits Not Associated with an Issue [Back to Top] This is a list of all changes that went into this release that did not directly close an issue from the issue tracker. The commits may have been marked as being related to an issue. If that is the case, the issue numbers are listed here, as well. +------------------------------------------------------------------------+ |Revision|Author |Summary |Issues | | | | |Referenced| |--------+------------+---------------------------------------+----------| |423802 |wdoekes |chan_sip: Unref outbound proxy | | | | |structure on dialog/pvt destruction. | | |--------+------------+---------------------------------------+----------| | | |res_pjsip.c: Add missing off nominal | | |423894 |rmudgett |cleanup in | | | | |ast_sip_push_task_synchronous(). | | |--------+------------+---------------------------------------+----------| |424056 |file |res_pjsip_session: Add additional | | | | |checks for delaying session refreshes. | | |--------+------------+---------------------------------------+----------| |424096 |rmudgett |threadpool.c: Minor cleanup fixes. | | |--------+------------+---------------------------------------+----------| |424103 |rmudgett |Simplify UUID generation in several | | | | |places. | | |--------+------------+---------------------------------------+----------| | | |res_rtp_asterisk: Ensure that the base | | |424152 |file |and mapped address for candidates is | | | | |present in SDP. | | |--------+------------+---------------------------------------+----------| | | |res_pjsip_sdp_rtp: Don't place an extra| | |424155 |file |whitespace before 'rport' and don't put| | | | |IPv6 addresses in brackets. | | |--------+------------+---------------------------------------+----------| |424183 |wdoekes |chan_sip: Simplify some unref code by | | | | |removing unlink_peer_from_tables. | | |--------+------------+---------------------------------------+----------| |424244 |kmoore |PJSIP: Force transport on contact | | | | |rewrite | | |--------+------------+---------------------------------------+----------| |424263 |kmoore |PJSIP: Handle defaults properly | | |--------+------------+---------------------------------------+----------| | | |res_pjsip_sdp_rtp: Accept DTLS | | |424287 |file |attributes in top level, not just media| | | | |session. | | |--------+------------+---------------------------------------+----------| | | |res_pjsip: Add 'dtls_fingerprint' | | |424290 |file |option to configure DTLS fingerprint | | | | |hash. | | |--------+------------+---------------------------------------+----------| |424337 |sgriepentrog|res_pjsip: document use of | | | | |rewrite_contact in sample conf | | |--------+------------+---------------------------------------+----------| |424414 |file |res_pjsip_session: Reduce SDP size by | | | | |removing duplicate connection lines. | | |--------+------------+---------------------------------------+----------| |424426 |kmoore |PJSIP: Restore functional default for | | | | |callerid_privacy | | |--------+------------+---------------------------------------+----------| | | |sorcery: Prevent SEGV in | | |424447 |gtjoseph |sorcery_wizard_create when there's no | | | | |create function | | |--------+------------+---------------------------------------+----------| |424528 |rmudgett |res_pjsip: Fix XML typo and update | | | | |UPGRADE.txt. | | |--------+------------+---------------------------------------+----------| |424646 |mjordan |sdp_srtp: Add new lines to some WARNING| | | | |messages | | |--------+------------+---------------------------------------+----------| |424941 |rmudgett |cdr.c: Make turning on CDR debug a one | | | | |step process instead of two. | | |--------+------------+---------------------------------------+----------| |424963 |gtjoseph |res_phoneprov: Refactor phoneprov to | | | | |allow pluggable config providers | | |--------+------------+---------------------------------------+----------| |424985 |mjordan |res/res_phoneprov: Don't cancel | | | | |Asterisk load on module load failure | | |--------+------------+---------------------------------------+----------| |425007 |gtjoseph |res_pjsip_phoneprov_provider: Provides | | | | |pjsip integration with res_phoneprov | | |--------+------------+---------------------------------------+----------| |425216 |file |res_pjsip_phoneprov_provider: Add | | | | |missing dependency on pjproject. | | |--------+------------+---------------------------------------+----------| |425220 |mjordan |res/res_phoneprov: Bail on registration| | | | |if res_phoneprov didn't load | | |--------+------------+---------------------------------------+----------| | | |bridge: During a smart bridge operation| | |425242 |file |provide a more complete bridge to the | | | | |old technology. | | |--------+------------+---------------------------------------+----------| |425264 |gtjoseph |res_phoneprov: Cleanup module load | | | | |error handling | | |--------+------------+---------------------------------------+----------| | | |res_rtp_asterisk: Make the ICE | | |425361 |file |transport check case insensitive as | | | | |some implementations use 'udp'. | | |--------+------------+---------------------------------------+----------| |425383 |gtjoseph |manager/config: Support templates and | | | | |non-unique category names via AMI | | |--------+------------+---------------------------------------+----------| |425480 |gtjoseph |res_phoneprov: Create accessor for | | | | |ast_phoneprov_std_variable_lookup | | |--------+------------+---------------------------------------+----------| |425525 |gtjoseph |config: Fix SEGV in unit test with | | | | |MALLOC_DEBUG | | |--------+------------+---------------------------------------+----------| | | |res_rtp_asterisk: Fix a bug where ICE | | |425645 |file |state would get reset when it | | | | |shouldn't. | | |--------+------------+---------------------------------------+----------| | | |config: Fix inf loop using | | |425713 |gtjoseph |ast_category_browse and | | | | |ast_variable_retrieve | | |--------+------------+---------------------------------------+----------| |425757 |mjordan |test_cel: Update pickup test to expect | | | | |CANCEL instead of ANSWSER | | |--------+------------+---------------------------------------+----------| |425921 |mjordan |res/res_pjsip_sdp_rtp: Check joint caps| | | | |when looking to decline outgoing media | | |--------+------------+---------------------------------------+----------| |425943 |mjordan |res/res_pjsip_sdp_rtp: Undo 425921 | | |--------+------------+---------------------------------------+----------| |425964 |gtjoseph |build: Force -fsigned-char on platforms| | | | |where the default for char is unsigned | | |--------+------------+---------------------------------------+----------| |426174 |mjordan |res/res_phoneprov: Fix crash on | | | | |shutdown caused by container cleanup | | |--------+------------+---------------------------------------+----------| |426210 |mjordan |res/res_http_websocket: Fix minor nits | | | | |found by wdoekes on r409681 | | |--------+------------+---------------------------------------+----------| | | |ASTERISK-24419, fix incorrect syntax | | |426293 |mdavenport |for setting language in | | | | |extensions.conf.sample | | |--------+------------+---------------------------------------+----------| | | |ASTERISK-24323, fix bug in | | |426361 |mdavenport |documentation of AGI STREAM FILE | | | | |CONTROL | | |--------+------------+---------------------------------------+----------| |426458 |mdavenport |ASTERISK-23512, correct inaccurate | | | | |comment in manager.conf.sample | | |--------+------------+---------------------------------------+----------| | | |bridge_builtin_features: Add missing | | |426531 |rmudgett |channel locks around | | | | |ast_get_chan_features_general_config().| | |--------+------------+---------------------------------------+----------| |426601 |mjordan |channels/chan_sip: Add improved support| | | | |for 4xx error codes | | |--------+------------+---------------------------------------+----------| | | |res_pjsip_exten_state: | | |426779 |kharwell |PJSIPShowSubscriptionsInbound causes | | | | |crash | | |--------+------------+---------------------------------------+----------| |426863 |mjordan |channels/sip/reqresp_parser: Fix unit | | | | |tests for r426594 | | |--------+------------+---------------------------------------+----------| |426933 |tzafrir |install init.d files on GNU/kFreeBSD | | |--------+------------+---------------------------------------+----------| |426995 |mjordan |res/res_stasis: Fix crash on module | | | | |unload while performing operation | | |--------+------------+---------------------------------------+----------| |427020 |coreyfarrell|func_jitterbuffer: fix frame leaks. | | |--------+------------+---------------------------------------+----------| |427088 |coreyfarrell|Fix compile error caused by review 4138| | |--------+------------+---------------------------------------+----------| |427129 |rmudgett |res_pjsip: Add disable_tcp_switch | | | | |option. | | +------------------------------------------------------------------------+ ---------------------------------------------------------------------- Diffstat Results [Back to Top] This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility. Makefile | 8 UPGRADE.txt | 22 addons/chan_ooh323.c | 3 apps/app_chanspy.c | 13 apps/app_directory.c | 2 apps/app_mixmonitor.c | 18 apps/app_queue.c | 9 apps/app_voicemail.c | 44 bridges/bridge_builtin_features.c | 4 bridges/bridge_native_rtp.c | 143 channels/chan_iax2.c | 3 channels/chan_misdn.c | 1 channels/chan_motif.c | 7 channels/chan_pjsip.c | 60 channels/chan_sip.c | 118 channels/chan_skinny.c | 1 channels/chan_unistim.c | 25 channels/pjsip/dialplan_functions.c | 12 channels/sip/include/reqresp_parser.h | 5 channels/sip/reqresp_parser.c | 6 configs/extensions.conf.sample | 2 configs/manager.conf.sample | 2 configs/phoneprov.conf.sample | 10 configs/pjsip.conf.sample | 140 configure.ac | 19 contrib/Makefile | 29 contrib/ast-db-manage/config/versions/10aedae86a32_add_outgoing_enum_va.py | 83 contrib/scripts/autosupport | 12 contrib/scripts/safe_asterisk | 11 doc/asterisk.8 | 2 funcs/func_frame_trace.c | 4 include/asterisk/chanvars.h | 18 include/asterisk/config.h | 184 + include/asterisk/frame.h | 1 include/asterisk/phoneprov.h | 124 include/asterisk/res_pjsip.h | 9 include/asterisk/res_pjsip_pubsub.h | 9 include/asterisk/res_pjsip_session.h | 76 include/asterisk/taskprocessor.h | 10 main/abstract_jb.c | 15 main/app.c | 18 main/audiohook.c | 20 main/bridge.c | 58 main/bridge_channel.c | 4 main/callerid.c | 16 main/cdr.c | 9 main/channel.c | 62 main/chanvars.c | 63 main/config.c | 338 +- main/core_unreal.c | 9 main/editline/readline.c | 1 main/features.c | 22 main/framehook.c | 14 main/manager.c | 438 ++ main/message.c | 6 main/sched.c | 1 main/sdp_srtp.c | 4 main/sorcery.c | 6 main/stasis_channels.c | 400 +- main/taskprocessor.c | 18 main/tcptls.c | 22 main/threadpool.c | 19 makeopts.in | 1 pbx/pbx_realtime.c | 2 res/res_agi.c | 4 res/res_calendar_ews.c | 13 res/res_fax.c | 31 res/res_hep.c | 1 res/res_hep_pjsip.c | 10 res/res_http_websocket.c | 16 res/res_jabber.c | 5 res/res_phoneprov.c | 1473 ++++++---- res/res_phoneprov.exports.in | 6 res/res_pjsip.c | 236 + res/res_pjsip/config_system.c | 8 res/res_pjsip/config_transport.c | 137 res/res_pjsip/location.c | 7 res/res_pjsip/pjsip_cli.c | 6 res/res_pjsip/pjsip_configuration.c | 49 res/res_pjsip/pjsip_distributor.c | 4 res/res_pjsip/pjsip_options.c | 130 res/res_pjsip_acl.c | 2 res/res_pjsip_authenticator_digest.c | 11 res/res_pjsip_caller_id.c | 2 res/res_pjsip_dialog_info_body_generator.c | 2 res/res_pjsip_diversion.c | 2 res/res_pjsip_dtmf_info.c | 2 res/res_pjsip_endpoint_identifier_anonymous.c | 2 res/res_pjsip_endpoint_identifier_ip.c | 6 res/res_pjsip_endpoint_identifier_user.c | 2 res/res_pjsip_exten_state.c | 6 res/res_pjsip_header_funcs.c | 2 res/res_pjsip_logger.c | 6 res/res_pjsip_messaging.c | 2 res/res_pjsip_multihomed.c | 2 res/res_pjsip_mwi.c | 2 res/res_pjsip_mwi_body_generator.c | 2 res/res_pjsip_nat.c | 9 res/res_pjsip_notify.c | 2 res/res_pjsip_one_touch_record_info.c | 2 res/res_pjsip_outbound_authenticator_digest.c | 2 res/res_pjsip_outbound_registration.c | 31 res/res_pjsip_path.c | 2 res/res_pjsip_phoneprov_provider.c | 424 ++ res/res_pjsip_pidf_body_generator.c | 2 res/res_pjsip_pidf_digium_body_supplement.c | 2 res/res_pjsip_pidf_eyebeam_body_supplement.c | 2 res/res_pjsip_pubsub.c | 15 res/res_pjsip_refer.c | 3 res/res_pjsip_registrar.c | 2 res/res_pjsip_registrar_expire.c | 2 res/res_pjsip_rfc3326.c | 2 res/res_pjsip_sdp_rtp.c | 137 res/res_pjsip_send_to_voicemail.c | 2 res/res_pjsip_session.c | 361 +- res/res_pjsip_session.exports.in | 2 res/res_pjsip_t38.c | 2 res/res_pjsip_transport_websocket.c | 2 res/res_pjsip_xpidf_body_generator.c | 2 res/res_rtp_asterisk.c | 37 res/res_sorcery_config.c | 8 res/res_sorcery_realtime.c | 7 res/res_srtp.c | 1 res/res_stasis.c | 28 res/res_xmpp.c | 6 rest-api/api-docs/applications.json | 2 rest-api/api-docs/asterisk.json | 2 rest-api/api-docs/bridges.json | 2 rest-api/api-docs/channels.json | 2 rest-api/api-docs/deviceStates.json | 2 rest-api/api-docs/endpoints.json | 2 rest-api/api-docs/events.json | 5 rest-api/api-docs/mailboxes.json | 2 rest-api/api-docs/playbacks.json | 2 rest-api/api-docs/recordings.json | 2 rest-api/api-docs/sounds.json | 2 tests/test_callerid.c | 4 tests/test_cel.c | 2 tests/test_config.c | 575 +++ tests/test_sorcery.c | 4 tests/test_sorcery_realtime.c | 15 141 files changed, 5064 insertions(+), 1662 deletions(-) ----------------------------------------------------------------------