commit 27a51ad64a9a1f61531c082ec5ecbe2c7dc41233cd6b1ab4878ed1c138eefba3 Author: Jan Engelhardt Date: Wed Oct 23 20:43:06 2024 +0200 Initial import: asterisk 18.12.1 diff --git a/asterisk-18.12.1.tar.gz b/asterisk-18.12.1.tar.gz new file mode 100644 index 0000000..50ccc92 Binary files /dev/null and b/asterisk-18.12.1.tar.gz differ diff --git a/asterisk-18.12.1.tar.gz.asc b/asterisk-18.12.1.tar.gz.asc new file mode 100644 index 0000000..a74a968 --- /dev/null +++ b/asterisk-18.12.1.tar.gz.asc @@ -0,0 +1,17 @@ +-----BEGIN PGP SIGNATURE----- +Version: GnuPG v2.0.22 (GNU/Linux) + +iQIcBAABAgAGBQJihmfMAAoJEF2YS+M3GRznVJwP/1k4f0YhR5UjbhjA4O9jiGT1 +KuCWF5c5aPL5Z1ne6XkQ6NdJbokv+MkY44q0pkDuypPNpCPc0nxFCkpcAestncL+ +6PZVHx3VJFnhrTu+Lm9FNY5WYh7wV+eEedNW3dkw7lx95grw3BaG/w1RReY964kh +6YPA9QhuSWXl1Yr5J6kgsLAygNcNjI4PzyzmdUpij5VV+vwpiL3xebgROne7hMO4 +qPWbN82RZwq1PPte6mfL4lhM1dO5FyeFVdzYZwIEsuT9InmCftQRh6sST9PS39vP +uY38HAhzCRUr8nZxZv3BWvT4eiWf5Lm2IhZSvSqG14o6AG8Sy+/ttwlTaWjzO9A4 +KLhFRrgduMI0+enZ/rrD3C0mG71ecfvMeiz6vXfD4YPqLjdYwDVw3pfhJ4/zV6kN +Mk685PtLP2vnYN8Av7y6Ub3dWVTN92qnbXXulsmLKEhTvQ4wwg/Gk6eF75lNJ7Gj +scxF2oayya9tOOgbcwSEtbJ8AbV4bJr27twDmYcgGxVV9GmcSaDVNBFRkg8GplLl +Jl6OJh9thIt1FtRTlVrBpnlBoEqzlBTEeWWihB/lvP6wYTK/OwRX2XBP0PiZ93K8 +FqmRTiHpMO1fnrBMXfDvFez8o/5bSw53qKTK7LxK9Bxa1MZs0J9LCPXSPTTGDJiM +LbfE3dEJkCgmi2ezAqMn +=mfGE +-----END PGP SIGNATURE----- diff --git a/asterisk-cflags.diff b/asterisk-cflags.diff new file mode 100644 index 0000000..12f3c3e --- /dev/null +++ b/asterisk-cflags.diff @@ -0,0 +1,30 @@ +From: Jan Engelhardt +Date: 2016-10-01 17:23:37.085881788 +0200 +References: https://bugzilla.opensuse.org/show_bug.cgi?id=1002419 + +build: restore dropping of -march=native + +--- + configure.ac | 9 +++++++-- + 1 file changed, 7 insertions(+), 2 deletions(-) + +Index: asterisk-14.0.0/configure.ac +=================================================================== +--- asterisk-14.0.0.orig/configure.ac ++++ asterisk-14.0.0/configure.ac +@@ -1220,8 +1220,13 @@ AC_SUBST(AST_SHADOW_WARNINGS) + + AC_MSG_CHECKING(for -march=native support) + if $(${CC} -march=native -S -o /dev/null -xc /dev/null > /dev/null 2>&1); then +- AC_MSG_RESULT(yes) +- AST_NATIVE_ARCH=1 ++ if test "${CONFIG_CFLAGS}" = ""; then ++ AC_MSG_RESULT(yes) ++ AST_NATIVE_ARCH=1 ++ else ++ AC_MSG_RESULT(user CFLAGS present) ++ AST_NATIVE_ARCH= ++ fi + else + AC_MSG_RESULT(no) + AST_NATIVE_ARCH=0 diff --git a/asterisk-configure-paths.diff b/asterisk-configure-paths.diff new file mode 100644 index 0000000..fefb1fb --- /dev/null +++ b/asterisk-configure-paths.diff @@ -0,0 +1,25 @@ +From: Jan Engelhardt +Date: 2011-10-09 13:06:02.765141186 +0200 +Upstream: tbd + +build: put data files into the right FHS spot + +Data should go into, well, ${datadir}! + +--- + configure.ac | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +Index: asterisk-11.0.0-rc2/configure.ac +=================================================================== +--- asterisk-11.0.0-rc2.orig/configure.ac ++++ asterisk-11.0.0-rc2/configure.ac +@@ -51,7 +51,7 @@ AC_SUBST([astlibdir], ['${libdir + AC_SUBST([astmoddir], ['${libdir}/asterisk/modules'])dnl + AC_SUBST([astmandir], ['${mandir}'])dnl + AC_SUBST([astvarlibdir], ['${localstatedir}/lib/asterisk'])dnl +-AC_SUBST([astdatadir], ['${astvarlibdir}'])dnl ++AC_SUBST([astdatadir], ['${datadir}/asterisk'])dnl + AC_SUBST([astdbdir], ['${astvarlibdir}'])dnl + AC_SUBST([astkeydir], ['${astvarlibdir}'])dnl + AC_SUBST([astspooldir], ['${localstatedir}/spool/asterisk'])dnl diff --git a/asterisk-init.diff b/asterisk-init.diff new file mode 100644 index 0000000..912cfa6 --- /dev/null +++ b/asterisk-init.diff @@ -0,0 +1,27 @@ +From: Jan Engelhardt +Date: 2011-10-10 01:33:55.485966709 +0200 +Upstream: tbd + +config: fix wrong runlevel assignments in suse init script + +Runlevel 4 is not used in SUSE, never was. + +--- + contrib/init.d/rc.suse.asterisk | 4 ++-- + 1 file changed, 2 insertions(+), 2 deletions(-) + +Index: asterisk-10.0.0-beta2/contrib/init.d/rc.suse.asterisk +=================================================================== +--- asterisk-10.0.0-beta2.orig/contrib/init.d/rc.suse.asterisk ++++ asterisk-10.0.0-beta2/contrib/init.d/rc.suse.asterisk +@@ -26,8 +26,8 @@ + # Required-Stop: $network $syslog $named $local_fs $remote_fs + # Should-Start: dahdi misdn lcr wanrouter mysql postgresql + # Should-Stop: dahdi misdn lcr wanrouter mysql postgresql +-# Default-Start: 2 3 4 5 +-# Default-Stop: 0 1 6 ++# Default-Start: 3 5 ++# Default-Stop: 0 1 2 4 6 + # Short-Description: Asterisk PBX + # Description: the Asterisk Open Source PBX + ### END INIT INFO diff --git a/asterisk-rundir.diff b/asterisk-rundir.diff new file mode 100644 index 0000000..7595c6e --- /dev/null +++ b/asterisk-rundir.diff @@ -0,0 +1,26 @@ +From: Jan Engelhardt +Date: 2011-10-24 03:40:25.085674838 +0200 +Upstream: unlikely + +config: use /run + +(Only apply this for systems that have /run in their filesystem +tree.) + +--- + configure.ac | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +Index: asterisk-10.0.0-beta2/configure.ac +=================================================================== +--- asterisk-10.0.0-beta2.orig/configure.ac ++++ asterisk-10.0.0-beta2/configure.ac +@@ -55,7 +55,7 @@ AC_SUBST([astdbdir], ['${astvar + AC_SUBST([astkeydir], ['${astvarlibdir}'])dnl + AC_SUBST([astspooldir], ['${localstatedir}/spool/asterisk'])dnl + AC_SUBST([astlogdir], ['${localstatedir}/log/asterisk'])dnl +-AC_SUBST([astvarrundir], ['${localstatedir}/run/asterisk'])dnl ++AC_SUBST([astvarrundir], ['/run/asterisk'])dnl + + case "${host_os}" in + *bsd*) diff --git a/asterisk.changes b/asterisk.changes new file mode 100644 index 0000000..57d8bee --- /dev/null +++ b/asterisk.changes @@ -0,0 +1,1610 @@ +------------------------------------------------------------------- +Mon Apr 17 10:51:03 UTC 2023 - Jan Engelhardt + +- Enable chan_mobile + +------------------------------------------------------------------- +Thu May 19 17:16:31 UTC 2022 - Michael Ströder + +- Updated pjproject to 2.12 +- Update to release 18.12.1 + * Release 18.12.1: + - [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing requests + * Release 18.12.0: + + Security fixes + - [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities + - [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \ + - [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files + + New Features + - [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter but everyone else can + - [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at prefix + - [ASTERISK-29486] - Hint-like extension value lookup function without device state + - [ASTERISK-29941] - chan_pjsip: Add ability to send flash events + - [ASTERISK-29820] - cli: Add command to evaluate a function + - [ASTERISK-29876] - app_queue: Add music on hold option + + Bugs fixes + - [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera available + - [ASTERISK-29638] - res_pjsip_session: No video after early media + - [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up Dahdi Call On Hold + - [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi restart + - [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with missing secrets + - [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled are always recompiled + - [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when channel is NULL + - [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max number of groups + - [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R" flags. (documentation bug) + - [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable console colorization" + - [ASTERISK-29843] - Session timers get removed on UPDATE + - [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented + - [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE + - [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if early_media already enabled + - [ASTERISK-29948] - iostream: Infinite TCP timeout writing data + - [ASTERISK-29253] - Incorrect bridging on transfer + - [ASTERISK-30006] - res_pjsip: UDP transport does not work when async_operations is greater than 1 + - [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not enabled + - [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new keyword + - [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database columns + - [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP attributes + - [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting channel for lack of RTP activity + - [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP activity in one way sessions + - [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name + - [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 + - [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context (AST_PBX_MAX_STACK - 1) + - [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't available + - [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or xmlstarlet ro be installed when it shouldn't + - [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats printout + - [ASTERISK-29939] - agi: Fix xmldoc bug with set music + - [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments displayed incorreclty + - [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived + - [ASTERISK-29674] - Adjust for 64bit time_t + - [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with SUBSCRIBE request + - [ASTERISK-29928] - logging messages truncated when using MUSL runtime + - [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file + - [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09' + + Improvements + - [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring cadences + - [ASTERISK-29940] - general: Add since tags to xmldocs + - [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP) implementation + - [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup + - [ASTERISK-29954] - app_meetme: Emit warning if conference not found + - [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk + - [ASTERISK-29976] - Should Readme include information about install_prereq script? + - [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries + - [ASTERISK-29980] - build: External binary modules don't use https + - [ASTERISK-25716] - Documentation: Document explanations and examples for possible values of DIALSTATUS + - [ASTERISK-29967] - pbx_builtins: Add missing documentation + +------------------------------------------------------------------- +Tue Apr 26 17:19:07 UTC 2022 - Michael Ströder + +- Update to release 18.11.3 + * [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not enabled + +------------------------------------------------------------------- +Fri Apr 15 08:29:49 UTC 2022 - Michael Ströder + +- Update to release 18.11.2 with security fixes for + * AST-2022-001: res_stir_shaken: resource exhaustion with large files + * AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header + * AST-2022-003: func_odbc: Possible SQL Injection +- remove unpackaged file + +------------------------------------------------------------------- +Wed Mar 30 08:16:33 UTC 2022 - Michael Ströder + +- Update to release 18.11.1 + * [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't available + * [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or xmlstarlet ro be installed when it shouldn't + +------------------------------------------------------------------- +Thu Mar 24 14:09:38 UTC 2022 - Michael Ströder + +- Updated to jansson-2.14 +- Update to release 18.11.0: + * Security bugs fixed: + - [ASTERISK-29945] - pjproject: Security fixes for things + * New Features: + - [ASTERISK-29853] - ami: Allow events to be globally disabled + - [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields + * Bugs fixed: + - [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'" error + - [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists + - [ASTERISK-29904] - RLS: Batched Notifications stop working + - [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown + - [ASTERISK-29873] - [patch] Queue Realtime load + - [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a call event. + - [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) + - [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs + - [ASTERISK-29886] - Asterisk AMI sends not-valid XML + * Improvements: + - [ASTERISK-29909] - app_queue: Add support for withdrawing a call + - [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists + - [ASTERISK-29353] - Qualify jansson 2.14 for asterisk + - [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs + - [ASTERISK-29896] - xmldocs: Add since tag + - [ASTERISK-29861] - asterisk.h: add macro for curl user agent + - [ASTERISK-29809] - curl, stir_shaken: refactor curl code + - [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox + - [ASTERISK-29925] - func_db: Warn about malformed key names + - [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions + - [ASTERISK-29866] - cli: add core dump information to core show settings + - [ASTERISK-29898] - documentation: Add default attributes to documentation + - [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility + - [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits + +------------------------------------------------------------------- +Sat Mar 5 10:04:14 UTC 2022 - Michael Ströder + +- Update to release 18.10.1 also with many bug fixes and small improvements + * Security fixes: + - AST-2022-004: pjproject: integer underflow on STUN message + - AST-2022-005: pjproject: undefined behavior after freeing a dialog set + - AST-2022-006: pjproject: unconstrained malformed multipart SIP message + * New Features 18.10.0: + - [ASTERISK-29808] cdr: allow disabling CDR by default + - [ASTERISK-29830] ami: Add AMI event for Wink + - [ASTERISK-29802] app_sf: Add full tech-agnostic SF support + - [ASTERISK-29759] app_sendtext: Add ReceiveText application + - [ASTERISK-29706] func_json: Add JSON parsing function + +------------------------------------------------------------------- +Sun Feb 6 13:36:20 UTC 2022 - Martin Hauke + +- Reenable build with support for DAHDI on supported platforms + +------------------------------------------------------------------- +Sat Jan 8 11:53:07 UTC 2022 - Michael Ströder + +- Update to release 18.9.0 + * New Features + - [ASTERISK-29720] - res_tonedetect: Add call progress tone detection + - [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to Queue Members + * Bugs fixed + - [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors. + - [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen + - [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are configured and we have a schema warning + - [ASTERISK-29776] - stir/shaken: Requires GNU designator + - [ASTERISK-29764] - chan_misdn: Fix for Doxygen + - [ASTERISK-29773] - progdocs: doxyref.h outdated + - [ASTERISK-29765] - xmldoc: Fix for Doxygen + - [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes + - [ASTERISK-29762] - channels: Fix for Doxygen + - [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in same bridge + - [ASTERISK-29754] - odbc: Fix for Doxygen + - [ASTERISK-29753] - parking: Fix for Doxygen + - [ASTERISK-29755] - frame: Fix for Doxygen + - [ASTERISK-29756] - res_ari: Fix for Doxygen + - [ASTERISK-29751] - channel: Fix for Doxygen + - [ASTERISK-29750] - stasis: Fix for Doxygen + - [ASTERISK-29752] - app: Fix for Doxygen + - [ASTERISK-29749] - res_xmpp: Fix for Doxygen + - [ASTERISK-29742] - addons: Fix for Doxygen. + - [ASTERISK-29747] - res_pjsip: Fix for Doxygen + - [ASTERISK-29737] - chan_iax2: Fix for Doxygen + - [ASTERISK-29743] - bridges: Fix for Doxygen + - [ASTERISK-29741] - tests: Fix for Doxygen + - [ASTERISK-29740] - apps: Fix for Doxygen + - [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file + - [ASTERISK-29736] - bridge_channel: Fix for Doxygen + - [ASTERISK-29735] - progdocs: Avoid multiple use of section labels + - [ASTERISK-29734] - progdocs: Use Doxygen \example correctly + - [ASTERISK-29744] - app_morsecode: Fix deadlock + - [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing + - [ASTERISK-29705] - app_read: Fix custom terminator functionality regression + - [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined. + - [ASTERISK-29702] - sig_analog: Fix truncated buffer copy + - [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from dynamic hints + - [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup + - [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions. + - [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable dispatched. + - [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized + - [ASTERISK-29713] - GCC 11.2: two stringop-overread + - [ASTERISK-29682] - Squash compiler issues generated by gcc 11 + - [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile + - [ASTERISK-27816] - func_talkdetect's logic is completely broken + - [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request + - [ASTERISK-26497] - make install downloads x86_32 variants of external modules on non Intel architectures + * Improvements + - [ASTERISK-29777] - documentation: Standardize example syntax + - [ASTERISK-29715] - app_voicemail: Refactor email generation functions + - [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent + - [ASTERISK-29714] - Spelling errors + - [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at dial time + - [ASTERISK-29662] - Add mix option to Playback application for say and filename + +------------------------------------------------------------------- +Thu Sep 9 21:40:15 UTC 2021 - Jan Engelhardt + +- Update to release 18.6.0 + * AST-2021-009 - pjproject-bundled: Avoid crash during + handshake for TLS + * app_reload: New Reload application + * app_waitforcond: New application + * app_dtmfstore: New application to store digits + * AST-2021-008 - chan_iax2: remote crash on unsupported + media format + +------------------------------------------------------------------- +Thu May 13 21:04:27 UTC 2021 - Diederik de Groot + +- Bug + + * Category: Applications/app_queue + ASTERISK-28356: app_queue: CLI set ringinuse for realtime member not + working + Reported by: Michael + * [35302efe73] Sean Bright -- app_queue: Add alembic migration to add + ringinuse to queue_members. + ASTERISK-24631: Incorrect description of option "context" in + queues.conf.sample + Reported by: Etienne Lessard + * [31364fa4c8] Sean Bright -- queues.conf.sample: Correct 'context' + documentation. + ASTERISK-26614: app_queue: updatecdr option in queues.conf does + effectively nothing + Reported by: Alexander Gonchiy + * [e27fa9eceb] Sean Bright -- app_queue.c: Remove dead 'updatecdr' code. + ASTERISK-27542: app_queue: When "queue show" CLI command is executed a + crash occurs + Reported by: Miguel Sanz + * [4393207751] Sean Bright -- app_queue.c: Don't crash when realtime + queue name is empty. + ASTERISK-29355: app_queue: Queue member status message sent even if status + doesn't change + Reported by: Roman Pertsev + * [55c467eab1] Joshua C. Colp -- app_queue: Only send QueueMemberStatus + if status changes. + + * Category: Bridges/bridge_simple + ASTERISK-29379: Segfault - ast_channel_is_multistream (chan=0x0) at + channel_internal_api.c:1590 + Reported by: Ross Beer + * [88aec107df] George Joseph -- bridge_channel_write_frame: Check for + NULL channel + + * Category: Channels/chan_local + ASTERISK-29035: chan_local: Multistream support breaks T.38 faxing + Reported by: Matthias Hensler + * [ed2f637b47] Joshua C. Colp -- core_unreal: Fix deadlock with T.38 + control frames. + + * Category: Core/BuildSystem + ASTERISK-29348: menuselect doesn't return errors in many cases + Reported by: George Joseph + * [f47c5cbdf9] Jaco Kroon -- menuselect: exit non-zero in case of + failure on --enable|disable options. + + * Category: Core/CodecInterface + ASTERISK-29328: translate.c: possible buffer overflow when upsampling + Reported by: Jean Aunis - Prescom + * [dec44306cf] Jean Aunis -- translate.c: Take sampling rate into + account when checking codec's buffer size + + * Category: Core/Stasis + ASTERISK-29355: app_queue: Queue member status message sent even if status + doesn't change + Reported by: Roman Pertsev + * [55c467eab1] Joshua C. Colp -- app_queue: Only send QueueMemberStatus + if status changes. + + * Category: Documentation + ASTERISK-24434: Fix differing usage of assignment operators in + modules.conf + Reported by: Rusty Newton + * [be3153346b] Sean Bright -- modules.conf: Fix more differing usages of + assignment operators. + ASTERISK-24631: Incorrect description of option "context" in + queues.conf.sample + Reported by: Etienne Lessard + * [31364fa4c8] Sean Bright -- queues.conf.sample: Correct 'context' + documentation. + ASTERISK-25358: dateformat not read from logger.conf by remote console + Reported by: Igor Liferenko + * [a0009c807e] Mark Murawski -- logger: Console sessions will now + respect logger.conf dateformat= option + + * Category: Resources/General + ASTERISK-29130: prometheus: Crash when scraping bridge + Reported by: Francisco Correia + * [19eef2a6dc] George Joseph -- res_prometheus: Clone containers before + iterating + + * Category: Resources/res_pjsip + ASTERISK-29354: res_pjsip: Allow partial reloading of transports + Reported by: Joshua C. Colp + * [f213833514] Joshua C. Colp -- res_pjsip: Add support for partial + transport reload. + + * Category: Resources/res_pjsip_session + ASTERISK-29215: res_pjsip_session: NULL active_media_state topology caused + asterisk crash + Reported by: sungtae kim + * [c78d0ce429] George Joseph -- res_pjsip_session: Make + reschedule_reinvite check for NULL topologies + + * Category: Resources/res_rtp_asterisk + ASTERISK-29364: res_rtp_asterisk: standard deviation miscalculation + Reported by: Kevin Harwell + * [17c86dcfaa] Kevin Harwell -- res_rtp_asterisk: Fix standard deviation + calculation + ASTERISK-29373: res_rtp_asterisk: Flash events are duplicated + Reported by: N A + * [b0d828f14a] Joshua C. Colp -- res_rtp_asterisk: Only raise flash + control frame on end. + ASTERISK-29352: res_rtp_asterisk: Fix frame delivery time when SSRC + changes + Reported by: Joshua C. Colp + * [2e7fc84398] Joshua C. Colp -- res_rtp_asterisk: Force resync on SSRC + change. + +- Improvement + + * Category: Core/General + ASTERISK-29339: loader: Let's output warnings for deprecated modules! + Reported by: Joshua C. Colp + * [a9a9864478] Joshua C. Colp -- loader: Output warnings for deprecated + modules. + ASTERISK-29337: menuselect: Add ability to set deprecated in and removed + in versions for modules + Reported by: Joshua C. Colp + * [6aac148d59] Joshua C. Colp -- menuselect: Add ability to set + deprecated and removed versions. + * [60fb559ccc] Joshua C. Colp -- xml: Allow deprecated_in and removed_in + for MODULEINFO. + ASTERISK-29335: xml: Embed module information into core XML documentation. + Reported by: Joshua C. Colp + * [60800b038a] Joshua C. Colp -- xml: Embed module information into core + XML documentation. + + * Category: Documentation + ASTERISK-29336: documentation: Fix inconsistent support levels + Reported by: Joshua C. Colp + * [be3e469f98] Joshua C. Colp -- documentation: Fix non-matching module + support levels. + ASTERISK-29335: xml: Embed module information into core XML documentation. + Reported by: Joshua C. Colp + * [60800b038a] Joshua C. Colp -- xml: Embed module information into core + XML documentation. + +------------------------------------------------------------------- +Fri Mar 26 12:25:27 UTC 2021 - Michael Ströder + +- update to 18.3.0 + * app_mixmonitor now sends manager events MixMonitorStart, MixMonitorStop and + MixMonitorMute when the channel monitoring is started, stopped and muted (or + unmuted) respectively. + * chan_iax2: You can now specify a default "auth" method in the + [general] section of iax.conf + * chan_pjsip, app_transfer: Added TRANSFERSTATUSPROTOCOL variable. + performing a REFER. + * Introduce an ARGC variable for func_odbc functions, along with a minargs + per-function configuration option. + * SRTP replay protection has been added to res_srtp and + a new configuration option "srtpreplayprotection" has + been added to the rtp.conf config file. + +------------------------------------------------------------------- +Sun Mar 14 22:20:25 UTC 2021 - Jan Engelhardt + +- Update to release 18.2.2 + * AST-2021-006 - res_pjsip_t38.c: Check for + session_media on reinvite. + +------------------------------------------------------------------- +Thu Feb 18 19:38:20 UTC 2021 - Michael Ströder + +- Update to 18.2.1 with security fixes: + * AST-2021-001: Remote crash in res_pjsip_diversion + * AST-2021-002: Remote crash possible when negotiating T.38 + * AST-2021-003: Remote attacker could prematurely tear down SRTP calls + * AST-2021-004: An unsuspecting user could crash Asterisk with multiple + * AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver + +------------------------------------------------------------------- +Sat Feb 13 19:58:22 UTC 2021 - Jan Engelhardt + +- Cut build recipe parts for platforms older than SLE/Leap 15 + +------------------------------------------------------------------- +Tue Feb 13 09:38:39 UTC 2021 - Asterisk Team + +- update to 18.2.0: + * Security + - [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains + * Bug fixes + - [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event + - [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy + - [ASTERISK-29155] - app_queue: Deadlock between queues container and individual queues + - [ASTERISK-29161] - Incorrect setup of recall channels + - [ASTERISK-29168] - Asterisk crashes during call transfer + - [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable + - [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses + - [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses + - [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped + - [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if registration can't be send + - [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the Transfer result + - [ASTERISK-29210] - res_pjsip: Crash when examining transport + - [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts + - [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are accepted. + - [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled. + - [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled user-agent. + - [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. + - [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are accepted. + - [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled. + - [ASTERISK-29209] - Debug messages printed by scope trace might be missing newlines + - [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels spuriously + - [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend + - [ASTERISK-29188] - null media causing the Asterisk crash + - [ASTERISK-29173] - Media cache URL requests allow infinite redirects + - [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without entries + - [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS response + - [ASTERISK-29191] - tel: URI in Diversion header causes crash + - [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered + - [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN instead of a channel variable + - [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all subscribers when using generic subscription + - [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description + - [ASTERISK-29191] - tel: URI in Diversion header causes crash + - [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set + * Improvements + - [ASTERISK-29118] - VoiceMail() should have an option to play greetings as Early Media + - [ASTERISK-28549] - Two repeated 183 + - [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other newer linux versions + - [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in /tmp + - [ASTERISK-28549] - Two repeated 183 + +------------------------------------------------------------------- +Tue Dec 22 09:58:39 2020 - Torrey Searle + +- Update for 18.1.1: + * Security bugs fixed: + - [AST-2020-001] - res_pjsip: Return dialog locked and referenced + - [AST-2020-002] - res_pjsip: Stop sending INVITEs after challenge limit. + +------------------------------------------------------------------- +Thu Nov 19 14:01:31 UTC 2020 - Michael Ströder + +- update to 17.9.0: + * Security bugs fixed: + - [ASTERISK-29057] - pjsip: Crash on call rejection during high load + * Improvements: + - [ASTERISK-29055] - Create a Bridge with video_single mode + - [ASTERISK-29056] - Increase reg_server column size for ps_contacts table realtime + * many more bug fixes + +------------------------------------------------------------------- +Fri Nov 6 09:08:03 UTC 2020 - Michael Ströder + +- update to 17.8.1 with security fixes: + * AST-2020-001: Remote crash in res_pjsip_session + * AST-2020-002: Outbound INVITE loop on challenge with different nonce. + +------------------------------------------------------------------- +Fri Oct 23 08:26:25 UTC 2020 - Hans-Peter Jansen + +- Asterisk 17.8.0 + * ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY + events (Reported by Ove Aursand) + * ASTERISK-29043 - app_queue: Leave empty sometimes not + recorded as abandoned (Reported by Kfir Itzhak) + * ASTERISK-29042 - res_parking: Parker UUID is no longer + copied (Reported by Misha Vodsedalek) + * ASTERISK-28878 - chan_pjsip: PJSIP_MEDIA_OFFER Broken + asterisk 16 (Reported by Joseph Ades) + * ASTERISK-29046 - pbx: Deadlock when doing a reload, while + simultaneously doing an ExtensionState on a pattern match hint + that ends up adding an extension (Reported by Ramarajan) + * ASTERISK-29040 - res_speech: Assertion on format + (Reported by Nickolay V. Shmyrev) + * ASTERISK-29001 - chan_pjsip does not process or forward 181 + responses (Reported by Torrey Searle) + * ASTERISK-29034 - Lastpause of realtime members is reseting + (Reported by Evandro César Arruda) + * ASTERISK-27273 - app_voicemail: When a voicemail is marked as + "Urgent", it is not sent by email/processed by the mailcmd + command (Reported by Leandro Dardini) + * ASTERISK-29033 - res_pjsip_session: Aggressively terminates + session on failed re-INVITE (Reported by Joshua C. Colp) + * ASTERISK-28974 - res_rtp_asterisk: T.140 messages have + appended RTP string to each message block. + (Reported by Thomas Johnson) + +------------------------------------------------------------------- +Mon Oct 19 15:55:34 UTC 2020 - Hans-Peter Jansen + +- Add dahdi build conditional + dahdi-linux is bitrotten, and TW kernel is moving too fast to catch up +- Use proper gmime dependency +- Add full asterisk include folder + +------------------------------------------------------------------- +Thu Sep 3 09:30:15 UTC 2020 - Michael Ströder + +- Update to release 17.7.0 + * [ASTERISK-29042] - res_parking: Parker UUID is no longer copied + * [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously + doing an ExtensionState on a pattern match hint that ends up adding an extension + * [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload + * [ASTERISK-29021] - Fix VERSION(ASTERISK_VERSION_NUM) on certified versions + * [ASTERISK-28927] - Asterisk crash in music on hold + * [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered + INVITE when NAT is active (UDP transport with external_media_address) + * [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured contacts is not correct + * [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info + * [ASTERISK-28978] - acl: named_acl rule misconfiguration results in + segfault on reading rule from realtime + * [ASTERISK-28975] - res_http_websocket: + Text payload data doesn't necessary include trailing zero + +------------------------------------------------------------------- +Fri Jul 17 16:06:36 UTC 2020 - Diederik de Groot + +- Update to release 17.6.0 + * AMI: You can now specify an optional 'Content-Type' as an argument + for the Asterisk SendText manager action. + * res_pjsip: Added a new PJSIP system setting called disable_rport. + * res_sorcery_memory_cache: The SorceryMemoryCacheExpireObject AMI + action and CLI command allow expiring of a specific object within + the sorcery memory cache. + * res_ari_channels: When creating a channel in ARI using the create + call you can now specify dialplan variables to be set as part of the + same operation. + * res_pjsip_logger: The PJSIP packet logger now has the following CLI + commands: + +------------------------------------------------------------------- +Sat Jun 6 07:00:36 UTC 2020 - Jan Engelhardt + +- Update to release 17.4.0 + * ARI: Application event filtering is now supported. An + application can now specify an "allowed" and/or "disallowed" + list(s) of event types. + * AttendedTransfer: A new application, this will queue up + attended transfer to the given extension. + * BlindTransfer: A new application, this will redirect all + channels currently bridged to the caller channel to the + specified destination. + * ConfBridge: Add "average_all", "highest_all", and + "lowest_all" values for the remb_behavior option. These + values operate on a bridge level instead of a per-source + level. + * Dial: Add RINGTIME and RINGTIME_MS variables containing + respectively seconds and milliseconds between creation of the + dialing channel and receiving the first RINGING signal. + * Dial: Add PROGRESSTIME and PROGRESSTIME_MS variables + analogous to the above with respect to the PROGRESS signal. + Shorter of these two times should be equivalent to the PDD + (Post Dial Delay) value. + * Dial: Add DIALEDTIME_MS and ANSWEREDTIME_MS variables to get + millisecond resolution versions of DIALEDTIME and + ANSWEREDTIME. + +------------------------------------------------------------------- +Thu Mar 12 18:37:36 UTC 2020 - Hans-Peter Jansen + +- Update to new upstream release 16.8.0 + + Bugs fixed in this release: + * ASTERISK-28766 - PJSIP blind transfer not completed after + using Proceeding() (Reported by lvl) + * ASTERISK-28685 - check_expr2: linking (when hardening) and + cross-compiling troubles (Reported by Sebastian Kemper) + * ASTERISK-28764 - res_rtp_asterisk: Improve NACK support and + seqno handling (Reported by Joshua C. Colp) + * ASTERISK-28755 - SIP/Stasis: SIP headers not transmitted in + the "variables" field (Reported by Jean Aunis - Prescom) + * ASTERISK-28754 - ASTERISK-28738 Causes Audio Issue After + Hold (Reported by Ross Beer) + * ASTERISK-28697 - res_pjsip: Named ACL does not update on + reload if changed (Reported by Timothy Vanderaerden) + * ASTERISK-28746 - res_pjsip_outbound_registration keeps + retrying the first entry in a SRV record set + (Reported by George Joseph) + * ASTERISK-28716 - ICE: pjnath shouldn't wait for ICE to + complete before allowing sending + (Reported by Benjamin Keith Ford) + * ASTERISK-28738 - Incorrect state machine used when + MOH_PASSTHRU is used (Reported by Torrey Searle) + * ASTERISK-28742 - res_rtp_asterisk: static for audio due to + incomplete dtls/srtp setup (Reported by Kevin Harwell) + * ASTERISK-28735 - Realtime MoH Unknown format '' -- + defaulting to SLIN (Reported by Ross Beer) + * ASTERISK-28730 - res_pjsip_session: Fix out of order + session refreshes (Reported by Joshua C. Colp) + * ASTERISK-28718 - chan_sip: Returns 403 if RTP ports are + depleted, should return 503 (Reported by Walter Doekes) + * ASTERISK-28719 - Cannot remove defaultrule from queue using + realtime queues (Reported by EDV O-TON) + * ASTERISK-28713 - res_stasis_playback: Error building JSON + (Reported by Sébastien Duthil) + * ASTERISK-28714 - REGRESSION: Feature + subscription_persistence_recreate (ASTERISK-27759) Causes + Segfaults (Reported by Ross Beer) + * ASTERISK-26082 - res_pjsip_messaging: MessageSend + Content-Type can't be changed (Reported by Alex) + * ASTERISK-28423 - ARI causes STASIS Deadlock + (Reported by Ross Beer) + * ASTERISK-28679 - stasis application is destroyed after its + creation (Reported by Francois Blackburn) + * ASTERISK-25421 - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS + in spite of the error when sending + (Reported by Dmitriy Serov) + * ASTERISK-28686 - chan_sip strictrtp=yes fails when media + source is changed: no audio (Reported by Walter Doekes) + * ASTERISK-28139 - RTP Stream Incorrect Payload Type Causes + Asterisk To Drop Calls (Reported by Paul Brooks) + * ASTERISK-26955 - pjsip: SIP Packets with Via "received=" + Containing IPv6 Address Delimited by "[]" Rejected + (Reported by Peter Sokolov) + + Improvements made in this release: + * ASTERISK-28750 - TLS/SSL Key too small error + (Reported by Martin Zeh) + * ASTERISK-28733 - stream: Add support for adding/removing + streams during SFU/calls (Reported by Joshua C. Colp) + * ASTERISK-24798 - Documentation - Clarify That Format Is Set + By File Name Extension In MixMonitor (Reported by xrobau) + * ASTERISK-28726 - install_prereq script uses the interactive + mode when installing aptitude (Reported by Sylvain Afchain) + +------------------------------------------------------------------- +Wed Feb 5 19:39:12 UTC 2020 - Hans-Peter Jansen + +- Update to new upstream release 16.8.0 + + New Features made in this release: + * ASTERISK-17491 - CURLOPT() needs a "followlocation" + parameter / "maxredirs" doesn't do anything + (Reported by candrews) + * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add + ability to match on source port + (Reported by Sean Bright) + + Bugs fixed in this release: + * ASTERISK-28679 - stasis application is destroyed after its + creation (Reported by Francois Blackburn) + * ASTERISK-28423 - ARI causes STASIS Deadlock + (Reported by Ross Beer) + * ASTERISK-28714 - REGRESSION: Feature + subscription_persistence_recreate (ASTERISK-27759) Causes + Segfaults (Reported by Ross Beer) + * ASTERISK-28677 - CDR billsec is always 0 for transferred + calls (Reported by Maciej Michno) + * ASTERISK-28702 - chan_dahdi: holding a channel via flash to + dialtone times out after 0:16:40 (Reported by Andrew Siplas) + * ASTERISK-28706 - silk 24hHz doesn't show up in 'core show + translation' output (Reported by Sean Bright) + * ASTERISK-24484 - Update documentation for statsd module - + usage requirements unclear (Reported by Dan Jenkins) + * ASTERISK-28695 - core: minmemfree watermark uses free RAM, + not available RAM (Reported by Kevin Flyn) + * ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a + whitespace appears empty in the dialplan + (Reported by Frank Matano) + * ASTERISK-23739 - [patch]Segfault forwarding voicemail with + ODBC storage enabled and realtime voicemail_data is used + (Reported by Stas Kobzar) + * ASTERISK-27622 - empty voicemail.conf required for ARA + (realtime) voicemail to leave message + (Reported by Jim Van Meggelen) + * ASTERISK-28349 - Pause reason not reported in QueueMember + AMI event (Reported by Niksa Baldun) + * ASTERISK-21794 - CLI command 'realtime update2' syntax + failure when using according to usage help + (Reported by Cedric BASSAGET) + * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document + support for hostnames (Reported by Joshua C. Colp) + * ASTERISK-27775 - res_pjsip_notify: Multiple Event headers + can be present instead of just one (Reported by AvayaXAsterisk) + * ASTERISK-28682 - app_record: Lack of `beep` audio file + causes application to return error and hangup + (Reported by Corey Farrell) + * ASTERISK-28507 - Wiki docs missing for MessageWaiting + (Reported by David M. Lee) + * ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence + does not preserve XML version number + (Reported by Bryan Nelson) + * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios + with concurrent command pri show span X + (Reported by Dirk Wendland) + * ASTERISK-28633 - stasis bridge topic leak + (Reported by Joeran Vinzens) + * ASTERISK-28492 - pjsip reload not reloading wizard + endpoint/pickup_group endpoint/call_group + (Reported by Jean-Denis Girard) + * ASTERISK-28562 - SIP WSS message not processed until next + frame arrives (Reported by Robert Sutton) + * ASTERISK-27243 - contrib: valgrind.supp doesn't suppress + what it's supposed to due to invalid syntax + (Reported by Richard Kenner) + * ASTERISK-28497 - func_odbc: truncating Unicode string on + readsql (Reported by Boris P. Korzun) + * ASTERISK-28647 - chan_sip: RTP frames not transmitted after + emitting a COLP (Reported by Jean Aunis - Prescom) + * ASTERISK-28667 - Asterisk ignores parsing of config files + if a Byte order mark is present (Reported by Robin Leffmann) + * ASTERISK-28664 - "trustrpid" is misspelled in + sip_to_pjsip.py (Reported by Pascal Cadotte Michaud) + * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql + don't build on 17.0.0 (Reported by George Joseph) + * ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes + non-existent media stream if codecs create additional streams + and offer does not have them (Reported by nappsoft) + * ASTERISK-28660 - res_fax: wrap Asterisk initiated + negotiation with config option (Reported by Kevin Harwell) + * ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes + fails to deactivate CDR. (Reported by Frederic LE FOLL) + * ASTERISK-28626 - Missing arguments in PJSIP_CONTACT + function documentation (Reported by Pascal Cadotte Michaud) + * ASTERISK-28609 - Memory Leak in res_rtp_asterisk.c + (Reported by Ted G) + * ASTERISK-28651 - chan_sip logs errors on tx to non-existent + TCP connections (Reported by Jaco Kroon) + * ASTERISK-28502 - chan_pjsip incorrectly re-writes REGISTER + 200 Response Contact (Reported by Ross Beer) + * ASTERISK-28625 - Playback of local files impacted by large + media cache (Reported by Kevin Reeves) + + Improvements made in this release: + * ASTERISK-28710 - Should be able to disable the /httpstatus + URI in the built-in HTTP server (Reported by Sean Bright) + * ASTERISK-28638 - Simplify dialplan for Dial, Page, and + ChanIsAvail (Reported by cmaj) + * ASTERISK-28673 - GET FULL VARIABLE documentation + clarification (Reported by Jonathan Harris) + * ASTERISK-28658 - app_confbridge: Add support for setting + maximum sample rate (Reported by Joshua C. Colp) + +------------------------------------------------------------------- +Wed Dec 25 13:06:01 UTC 2019 - Hans-Peter Jansen + +- Update to new upstream release 16.7.0 + + Security bugs fixed in this release: + * ASTERISK-28589 - chan_sip: Depending on configuration an + INVITE can alter Addr of a peer (Reported by Andrey V.T.) + * ASTERISK-28580 - Bypass SYSTEM write permission in manager + action allows system commands execution + (Reported by Eliel Sardañons) + + Improvements made in this release: + * ASTERISK-28602 - res_pjsip_outbound_registration: Maximum + retries reached (Reported by Daniel) + * ASTERISK-28586 - Typo in README-SERIOUSLY.bestpractices.md + (Reported by Sam Banks) + * ASTERISK-22192 - [patch] Allow voicemail forwards with ODBC + backend when format differs from attachfmt column + (Reported by cmaj) + * ASTERISK-28567 - Problem with ASTERISK-20207: Asterisk + should clear out any .lock files in the voice mail directory on + startup. (Reported by Michael) + * ASTERISK-28542 - [patch] add the ability for asterisk to + generate on-hold re-invites (Reported by Torrey Searle) + * ASTERISK-28512 - Add pass-through support for H.265 (HEVC) + codec (Reported by Florian Floimair) + + Bugs fixed in this release: + * ASTERISK-28609 - Memory Leak in res_rtp_asterisk.c + (Reported by Ted G) + * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql + don't build on 17.0.0 (Reported by George Joseph) + * ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes + non-existent media stream if codecs create additional streams + and offer does not have them + (Reported by nappsoft) + * ASTERISK-28641 - res_pjsip Segfaults when realtime + configuration to an AOR points to a not existent AOR + (Reported by Ross Beer) + * ASTERISK-28644 - Stale comment in app_queue about + ring_entry exception (Reported by Walter Doekes) + * ASTERISK-28445 - res_pjsip_session: ast_json_vpack: Invalid + UTF-8 string on hangup when TEST_FRAMEWORK enabled + (Reported by Bernhard Schmidt) + * ASTERISK-28637 - chan_sip+native_bridge_rtp: directmedia + compatibility check failure when negociated ptime is not + default ptime. (Reported by Frederic LE FOLL) + * ASTERISK-28631 - res_parking: Doesn't park when parkee and + parker are the same (Reported by Ross Beer) + * ASTERISK-28621 - Enforce T.38 error correction mode at 200 + ok received (Reported by Salah Ahmed) + * ASTERISK-28624 - res_pjsip_outbound_registration: add SRV + failover (Reported by Kevin Harwell) + * ASTERISK-28608 - app_amd: Use time calculation to calculate + timeout (Reported by Michael Cargile) + * ASTERISK-28615 - chan_dahdi: PRI span status may stay + "Down, Active" after a short alarm + (Reported by Frederic LE FOLL) + * ASTERISK-28576 - res_rtp_asterisk: ICE Completion Crash + when sent packet length doesn't match + (Reported by Joshua Elson) + * ASTERISK-26481 - FILE function grabs garbage along with + read data when target line has no newline + (Reported by Jonathan Harris) + * ASTERISK-28618 - bridge_softmix: hold not cleared when + joining a softmix bridge (Reported by Kevin Harwell) + * ASTERISK-28616 - parking: Deadlock when multi call parking + (Reported by Joshua C. Colp) + * ASTERISK-28423 - ARI causes STASIS Deadlock + (Reported by Ross Beer) + * ASTERISK-28572 - Memory leaks in res_calendar_exchange and + res_calendar_icalendar (Reported by Yoooooo Ha) + * ASTERISK-28585 - ari/resource_events: Crash in event + session cleanup (Reported by Kevin Harwell) + * ASTERISK-28590 - utils.c throws repeated warnings + "pthread_attr_setstacksize: Invalid argument" + (Reported by Speed Dial Dave) + * ASTERISK-28578 - race condition on pjsip channelstats + command (Reported by Salah Ahmed) + * ASTERISK-28571 - cdr_pgsql: accesses obsolete (and finally + removed) column (Reported by Christoph Moench-Tegeder) + * ASTERISK-28575 - MWI Send Notify Crash on 16.6 + (Reported by Joshua Elson) + * ASTERISK-28574 - pjproject fails to build on 16.6.0, works + on 16.5 (Reported by Niklas Larsson) + * ASTERISK-28561 - Asterisk Deadlocks + (Reported by Aheliotech) + * ASTERISK-28552 - res_pjsip_mwi: Frack during unload on + unsolicited_mwi container (Reported by Kevin Harwell) + * ASTERISK-28566 - CDR backend unload problem during active + call(s) (Reported by Marian Piater) + * ASTERISK-28553 - stasis.c: Crash during unload + (Reported by Kevin Harwell) + * ASTERISK-28086 - chan_pjsip: Crash when initiating PlayDTMF + over AMI (Reported by Jeremiah Gadd) + * ASTERISK-28544 - Wrong contact representation in ipv6 mode + (Reported by Jørgen H) + * ASTERISK-28534 - Segmentation fault when there is no + priority for an extension (Reported by Timothy Vanderaerden) + * ASTERISK-28463 - res_pjsip_path: Crash when invalid contact + is configured (Reported by Juan Martin) + * ASTERISK-28521 - pjsip: Memory Leak + (Reported by Mark) + * ASTERISK-28523 - Asterisk 16.5.0 Memory leak + (Reported by Cyril Ramière) + * ASTERISK-28538 - chan_pjsip: Deadlock on fax detection + (Reported by Joshua C. Colp) + * ASTERISK-28536 - Asterisk release candidates fail to build + on FreeBSD (Reported by Guido Falsi) + * ASTERISK-23756 - setvar directive when used in template and + a child of said template, results in duplicate variable names + (Reported by Michael Goryainov) + + New Features made in this release: + * ASTERISK-28614 - app_senddtmf: Allow "receiving" DTMF with + PlayDTMF instead of only "sending" (Reported by lvl) + * ASTERISK-28613 - func_curl: CURLOPT cannot set Content-Type + header (Reported by Martin Tomec) + * ASTERISK-28533 - func_jitterbuffer: Add support for video + synchronization (Reported by Joshua C. Colp) + +------------------------------------------------------------------- +Fri Nov 22 10:34:03 UTC 2019 - Hans-Peter Jansen + +- Update to new upstream release 16.6.2 + * ASTERISK-28580 + manager.c: Prevent the Originate action from running the Originate app + If an AMI user without the "system" authorization calls the + Originate AMI command with the Originate application, + the second Originate could run the "System" command. + Action: Originate + Channel: Local/1111 + Application: Originate + Data: Local/2222,app,System,touch /tmp/owned + If the "system" authorization isn't set, we now block the + Originate app as well as the System, Exec, etc. apps. + + * ASTERISK-28589 #close + chan_sip.c: Prevent address change on unauthenticated SIP request. + If the name of a peer is known and a SIP request is sent using that + peer's name, the address of the peer will change even if the request + fails the authentication challenge. This means that an endpoint can + be altered and even rendered unusuable, even if it was in a working + state previously. This can only occur when the nat option is set to the + default, or auto_force_rport. + This change checks the result of authentication first to ensure it is + successful before setting the address and the nat option. + +- Update to new upstream release 16.6.1 + * ASTERISK-28574 + pjproject_bundled: Replace earlier reverts with official fixes. + Issues in pjproject 2.9 caused us to revert some of their changes + as a work around. This introduced another issue where pjproject + wouldn't build with older gcc versions such as that found on + CentOS 6. This commit replaces the reverts with the official + fixes for the original issues and allows pjproject to be built + on CentOS 6 again. + + * ASTERISK-28575 + res_pjsip_mwi: potential double unref, and potential unwanted double link + When creating an unsolicited MWI aggregate subscription it was possible for + the subscription object to be double unref'ed. This patch removes the explicit + unref as it is not needed since the RAII_VAR will handle it at function end. + Less concerning there was also a bug that could potentially allow the aggregate + subscription object to be added to the unsolicited container twice. This patch + ensures it is added only once. + +------------------------------------------------------------------- +Sun Oct 13 15:40:20 UTC 2019 - Hans-Peter Jansen + +- Update to new upstream release 16.6.0 + - Security bugs fixed in this release: + * [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with + declined stream causes crash (Reported by Alexei Gradinari) + - Bugs fixed in this release: + * [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark) + * [ASTERISK-28523] - Asterisk 16.5.0 Memory leak + (Reported by Cyril Ramière) + * [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection + (Reported by Joshua C. Colp) + * [ASTERISK-28536] - Asterisk release candidates fail to build + on FreeBSD (Reported by Guido Falsi) + * [ASTERISK-28511] - codec_resample: Bad sound quality when up + sampling from SLIN16 to SLIN32 (Reported by Ruddy G) + * [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when + a PRI channel hangs up (Reported by Frederic LE FOLL) + * [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes + is set in cdr.conf (Reported by Frederic LE FOLL) + * [ASTERISK-28499] - translate: Crash when frame does not have a + "src" field set (Reported by Gregory Massel) + * [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized + type not at end of a struct (Reported by Alexander Traud) + * [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register + (Reported by Chris Savinovich) + * [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 + characters, NEC only supports up to 32 characters + (Reported by Dan Cropp) + * [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail + because not checking mailstream (Reported by Alexei Gradinari) + * [ASTERISK-28487] - compile menuselect on gentoo + (Reported by Kilburn) + * [ASTERISK-28472] - Asterisk occasionally passes a NULL as + srtp->session to srtp_protect/unprotect causing SEGV + (Reported by Jonas Swiatek) + * [ASTERISK-28498] - cel / cdr: Event times may be incorrect + (Reported by Joshua C. Colp) + * [ASTERISK-28480] - json integer overflow in ssrc and timestamp + (Reported by Salah Ahmed) + * [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double + entries (Reported by Ian Jones) + * [ASTERISK-28483] - packet lost on UDPTL wrap around + (Reported by Torrey Searle) + * [ASTERISK-28477] - Crash when not specifying "dbfile" in + res_config_sqlite3.conf (Reported by Dennis) + * [ASTERISK-28478] - Crash performing "core reload" with modified + res_config_sqlite3.conf (Reported by Dennis) + * [ASTERISK-26968] - chan_pjsip: Transfer() does not result in + TRANSFERSTATUS reflecting SIP response to transfer + (Reported by Dan Cropp) + * [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self + deadlocks (in chan_sip) (Reported by Walter Doekes) + - New Features made in this release: + * [ASTERISK-17808] - [patch] Unregister a realtime moh class + (Reported by Byron Clark) + * [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip + to setup From header URI domain (Reported by Stas Kobzar) + +------------------------------------------------------------------- +Fri Sep 20 08:53:46 UTC 2019 - Hans-Peter Jansen + +- Update to new upstream release 16.5.1 + - Security bugs fixed in this release: + * AST-2019-005 - translate: Don't assume all frames will have a src. + This change removes the assumption that a frame will always have + a src set on it. This assumption is incorrect. + * AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media + After receiving a 200 OK with a declined stream in response to a T.38 + initiated re-invite Asterisk would crash when attempting to dereference + a NULL session media object. + +------------------------------------------------------------------- +Tue Aug 13 07:04:15 UTC 2019 - Hans-Peter Jansen + +- Update to new upstream release 16.5.0 + - Security bugs fixed in this release: + * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with + no body causes crash (Reported by Gil Richard) + * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38 + reINVITE (Reported by Francesco Castellano) + - Bugs fixed in this release: + * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit + systems caused by ASTERISK-28317 (Reported by abelbeck) + * ASTERISK-28458 - res_pjsip_sdp_rtp: Remove unused variable + (Reported by Michael Maier) + * ASTERISK-26006 - Show offending IP for TLS setup failures in + logs (Reported by Oleksandr Natalenko) + * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors + not logged (Reported by Bernhard Schmidt) + * ASTERISK-28419 - app_amd: Does not work with silence + suppression (Reported by Nasir Iqbal) + * ASTERISK-28018 - IP Fragmentation happening instead of DTLS + fragmentation on handshake server hello certificate + (Reported by vijay kumar) + * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when + Asterisk attempts to generate hangup event + (Reported by Abhay Gupta) + * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work + (Reported by Dmitry Svyatogorov) + * ASTERISK-27981 - res_fax: Fax session leak with fax + gatewaying (Reported by pasandev) + * ASTERISK-28427 - new mwi.h include missing from some dahdi + source files, causes build failure + (Reported by Guido Falsi) + * ASTERISK-28421 - Wrong type used for timestamp in + res_rtp_asterisk (Reported by Morten Tryfoss) + * ASTERISK-27994 - PJSIP: Early media ringback not indicated + after Progress() (Reported by Gregory Massel) + - Improvements made in this release: + * ASTERISK-28234 - pbx_dundi: Add IPv4/IPv6 dual bind support + for DUNDi (Reported by Kirsty Tyerman) + + https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.0 + +- Update bundled pjproject tarball to 2.9 + +------------------------------------------------------------------- +Thu Jul 25 16:13:06 UTC 2019 - Hans-Peter Jansen + +- Add postgresql-server-devel dependency for Factory + +------------------------------------------------------------------- +Fri Jul 12 09:41:03 UTC 2019 - Hans-Peter Jansen + +- Update to new upstream release 16.4.1 + * AST-2019-002: Remote crash vulnerability with MESSAGE messages + A specially crafted SIP in-dialog MESSAGE message can cause Asterisk + to crash. + + * AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver + When T.38 faxing is done in Asterisk a T.38 reinv ite may be sent to an + endpoint to switch it to T.38. If the endpoint responds with an improperly + formatted SDP answer including both a T.38 UDPTL stream and an audio or video + stream containing only codecs not allowed on the SIP peer or user a crash will + occur. The code incorrectly assumes that there will be at least one common + codec when T.38 is also in the SDP answer. Fixes CVE-2019-13161. + + https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1 + +- Update to new upstream release 16.4.0 + https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.0 + +- Update to new upstream release 16.3.0 + https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.3.0 + +- Remove dependency on jansson (locally supplied) +------------------------------------------------------------------- +Fri Mar 1 10:11:49 UTC 2019 - Adam Majer + +- Update to new upstream release 16.2.1 + * Fix remote crash vulnerability in SDP protocol violation + (CVE-2019-7251) + +------------------------------------------------------------------- +Sat Feb 23 05:53:11 UTC 2019 - seanlew@opensuse.org + +- Update to new upstream release 16.2.0 + +------------------------------------------------------------------- +Wed Dec 26 22:51:36 UTC 2018 - Michael Ströder + +- Update to new upstream release 16.1.1 + * Fix for Regression: MWI polling no longer works + +------------------------------------------------------------------- +Wed Dec 12 22:59:05 UTC 2018 - Jan Engelhardt + +- Update to new upstream release 16.1.0 + * Fix a buffer overflow for DNS SRV/NAPTR records + * Fix a crash when reading HTTP Upgrade request + +------------------------------------------------------------------- +Thu Nov 22 19:47:07 UTC 2018 - Hans-Peter Jansen + +- Enable app_macro build + deprecated now, but missing it will break too many diaplans + +------------------------------------------------------------------- +Tue Nov 20 17:32:52 UTC 2018 - Jan Engelhardt + +- Update to new upstream release 16.0.1 + * webrtc: Both REMB and NACK are now supported. + * Text messages sent through a conference bridge using + ConfBridge will now be relayed to the other participants. + * app_originate: The 'a' option has been added which + asynchronously places calls. The application will return + immediately instead of waiting for the originated channel + to answer. + * app_queue: A wrapup time can now be configured on a per-member + basis instead of on a per-queue basis for static members as + defined in the configuration file. + * app_queue: Predial handler support has also been added so that + subroutines can be invoked on the callee or caller channels. + +------------------------------------------------------------------- +Thu Oct 25 10:45:46 UTC 2018 - Hans-Peter Jansen + +- Add missing /usr/share/asterisk/keys directory for res_crypto +- Adjust permissions of /var/lib/asterisk/phoneprov/* + +------------------------------------------------------------------- +Wed Oct 24 17:01:01 UTC 2018 - Hans-Peter Jansen + +- Improve systemd unit to wait for network online state + +------------------------------------------------------------------- +Sun Oct 21 10:56:40 UTC 2018 - Hans-Peter Jansen + +- Don't install /etc/init.d script, if systemd driven + +------------------------------------------------------------------- +Mon Oct 15 15:19:39 UTC 2018 - Hans-Peter Jansen + +- Make adjusting asterisk.conf actually work, and prevent that perl + expression from failing unnoticed ever again. + +------------------------------------------------------------------- +Fri Sep 21 10:32:43 UTC 2018 - Michael Ströder + +- Update to new upstream release 15.6.1 + * AST-2018-009: Remote crash vulnerability in HTTP websocket upgrade + +------------------------------------------------------------------- +Thu Jul 12 19:27:24 UTC 2018 - michael@stroeder.com + +- Update to new upstream release 15.5.0 with following + security fixes: + * [ASTERISK-27818] + Username bruteforce is possible when using ACL with PJSIP + * [ASTERISK-27807] + iostreams: Potential DoS when client connection closed prematurely + +------------------------------------------------------------------- +Tue Jun 19 10:04:31 UTC 2018 - adam.majer@suse.de + +- drop pwlib-devel from BR as it is not going to be ported to + OpenSSL 1.1 (boo#1074796) + +------------------------------------------------------------------- +Tue Jun 12 06:32:13 UTC 2018 - michael@stroeder.com + +- Update to new upstream release 15.4.1 with following + security fixes: + * AST-2018-007: Infinite loop when reading iostreams + * AST-2018-008: PJSIP endpoint presence disclosure when using ACL + +------------------------------------------------------------------- +Sat May 26 14:32:42 UTC 2018 - dev@stellardeath.org + +- Switch to bundled pjproject to avoid segmentation faults when + using channel PJSIP + +------------------------------------------------------------------- +Mon May 14 10:07:59 UTC 2018 - michael@stroeder.com + +- Update to new upstream release 15.4.0 with following + security fixes: + * [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS + * [ASTERISK-27583] - Segmentation fault occurs in asterisk with an + invalid SDP fmtp attribute + * [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an + invalid SDP media format description + * [ASTERISK-27618] - Crash occurs when sending a repeated number of + INVITE messages over TCP or TLS transport + * [ASTERISK-27640] - SUBSCRIBE message with a large Accept value + causes stack corruption + +------------------------------------------------------------------- +Sun May 6 14:44:26 UTC 2018 - dev@stellardeath.org + +- Remove sqlite2-devel as BuildRequires, is no longer available + +------------------------------------------------------------------- +Sun Mar 4 14:25:11 UTC 2018 - jengelh@inai.de + +- Update to new upstream release 15.2.2 + * The "Data Retrieval API" has been removed. This API was not + actively maintained, was not added to new modules (such as + res_pjsip), and there exist better alternatives to acquire + the same information, such as the ARI. As a result, the + "DataGet" AMI action as well as the "data get" CLI command + have been removed. + * Streams, as a new concept for media flows, have been + introduced. + * To simplify configuration for users a new option, webrtc, has + been created which controls configuration options that are + required for WebRTC. + +------------------------------------------------------------------- +Wed Feb 21 23:40:10 UTC 2018 - michael@stroeder.com + +- Update to new upstream release 14.7.6 + * AST-2018-001: Crash when receiving unnegotiated dynamic payload + * AST-2018-002: Crash when given an invalid SDP media format description + * AST-2018-003: Crash with an invalid SDP fmtp attribute + * AST-2018-004: Crash when receiving SUBSCRIBE request + * AST-2018-005: Crash when large numbers of TCP connections are closed suddenly + * AST-2018-006: WebSocket frames with 0 sized payload causes DoS + +------------------------------------------------------------------- +Sat Dec 23 09:41:36 UTC 2017 - michael@stroeder.com + +- Update to new upstream release 14.7.5 + * AST-2017-014: Crash in PJSIP resource when missing a contact header + +------------------------------------------------------------------- +Thu Dec 14 22:06:50 UTC 2017 - michael@stroeder.com + +- Update to new upstream release 14.7.4 + AST-2017-012: Remote Crash Vulnerability in RTCP Stack + +------------------------------------------------------------------- +Fri Dec 1 21:40:51 UTC 2017 - michael@stroeder.com + +- Update to new upstream release 14.7.3 + * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny + +------------------------------------------------------------------- +Tue Sep 19 19:23:13 UTC 2017 - michael@stroeder.com + +- Update to new upstream release 14.6.2 with security fix: + AST-2017-008: RTP/RTCP information leak +- HTTPS source download links + +------------------------------------------------------------------- +Wed Sep 6 13:06:14 UTC 2017 - michael@stroeder.com + +- Update to new upstream release 14.6.1 with security fixes: + * AST-2017-007: Remote Crash Vulerability in res_pjsip + * AST-2017-006: Shell access command injection in app_minivm + * AST-2017-005: Media takeover in RTP stack + +------------------------------------------------------------------- +Wed Jul 12 14:07:26 UTC 2017 - michael@stroeder.com + +- Update to new upstream release 14.6.0 + +------------------------------------------------------------------- +Tue May 30 18:44:30 UTC 2017 - michael@stroeder.com + +- Update to new upstream release 14.5.0 + +------------------------------------------------------------------- +Mon May 22 19:58:24 UTC 2017 - hpj@urpla.net + +- separate doc package (with ~26 MB) + +------------------------------------------------------------------- +Sat May 20 07:58:46 UTC 2017 - michael@stroeder.com + +- Update to new upstream release 14.4.1 with security fixes: + * AST-2017-002: Ensure transaction key buffer is large enough + * AST-2017-003: Handle zero-length body parts correctly + * AST-2017-004: chan_skinny: Add EOF check in skinny_session + +------------------------------------------------------------------- +Fri Apr 7 17:22:26 UTC 2017 - michael@stroeder.com + +- Update to new upstream release 14.4.0 + +------------------------------------------------------------------- +Tue Feb 21 22:13:30 UTC 2017 - jengelh@inai.de + +- asterisk won't start without the sound directory + (so make sure it is always there) + +------------------------------------------------------------------- +Mon Feb 13 22:47:10 UTC 2017 - michael@stroeder.com + +- Update to new upstream release 14.3.0 + +------------------------------------------------------------------- +Tue Jan 17 13:39:09 UTC 2017 - jengelh@inai.de + +- Enable app_meetme + +------------------------------------------------------------------- +Thu Dec 8 21:59:57 UTC 2016 - michael@stroeder.com + +- Update to new upstream release 14.2.1 with security fixes: + * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus + * AST-2016-009: Remote unauthenticated sessions in chan_sip + +------------------------------------------------------------------- +Wed Nov 23 18:25:45 UTC 2016 - michael@stroeder.com + +- Update to new upstream release 14.2.0 + +------------------------------------------------------------------- +Thu Nov 10 20:53:36 UTC 2016 - michael@stroeder.com + +- Update to new upstream release 14.1.2 + +------------------------------------------------------------------- +Wed Oct 26 18:21:07 UTC 2016 - michael@stroeder.com + +- Update to new upstream release 14.1.1 + +------------------------------------------------------------------- +Sat Oct 1 15:24:14 UTC 2016 - jengelh@inai.de + +- Add asterisk-cflags.diff to drop -march=native again + [boo#1002419] + +------------------------------------------------------------------- +Sat Oct 1 00:14:18 UTC 2016 - michael@stroeder.com + +- Update to new upstream release 14.0.2 + +------------------------------------------------------------------- +Tue Sep 27 19:56:20 UTC 2016 - michael@stroeder.com + +- Update to new upstream release 14.0.1 + +------------------------------------------------------------------- +Mon Sep 26 20:01:56 UTC 2016 - jengelh@inai.de + +- Update to new upstream release 14.0.0 + +------------------------------------------------------------------- +Fri May 13 19:58:36 UTC 2016 - michael@stroeder.com + +- Update to new upstream maintenance release 13.9.1 + +------------------------------------------------------------------- +Wed Mar 30 06:30:17 UTC 2016 - michael@stroeder.com + +- Update to new upstream maintenance release 13.9.0 + +------------------------------------------------------------------- +Sat Feb 6 22:11:13 UTC 2016 - jengelh@inai.de + +- Update to new upstream maintenance release 13.7.2 + +------------------------------------------------------------------- +Fri Feb 5 18:33:22 UTC 2016 - michael@stroeder.com + +- Update to new upstream maintenance release 13.7.1 + with security fixes: + * AST-2016-001: BEAST vulnerability in HTTP server + * AST-2016-002: File descriptor exhaustion in chan_sip + * AST-2016-003: Remote crash vulnerability when receiving UDPTL FAX data. +- Added build dependencies: + * libv4l-devel + * libSDL2-devel + +------------------------------------------------------------------- +Thu Dec 10 12:15:00 UTC 2015 - zawel1@gmail.com + +- Update to new upstream maintenance release 13.6.0 + +------------------------------------------------------------------- +Thu Aug 13 15:16:00 UTC 2015 - jengelh@inai.de + +- Update to new upstream maintenance release 13.5.0 + +------------------------------------------------------------------- +Mon Jun 8 22:30:50 UTC 2015 - jengelh@inai.de + +- Update to new upstream maintenenace release 13.4.0 + +------------------------------------------------------------------- +Thu Apr 9 10:14:52 UTC 2015 - jengelh@inai.de + +- Update to new upstream maintenance release 13.3.2 +* fix for CVE-2015-3008 asterisk: TLS Certificate Common name NULL + byte exploit + +------------------------------------------------------------------- +Mon Mar 16 20:53:29 UTC 2015 - jengelh@inai.de + +- Update to new upstream maintenance release 13.2 + +------------------------------------------------------------------- +Sat Jan 3 15:43:55 UTC 2015 - jengelh@inai.de + +- Update to new upstream maintenance release 13.1 + +------------------------------------------------------------------- +Thu Nov 20 23:22:45 UTC 2014 - joop.boonen@opensuse.org + +- Build version 13.0.1 + +------------------------------------------------------------------- +Thu Nov 20 21:13:00 UTC 2014 - joop.boonen@opensuse.org + +- Corrected the file paths +- Added missing files +- Added excludes + +------------------------------------------------------------------- +Mon Nov 17 20:51:45 UTC 2014 - jengelh@inai.de + +- Update to new upstream release 13 +* Asterisk security events are now provided via AMI, allowing end + users to monitor their Asterisk system in real time for security + related issues. +* Both AMI and ARI now allow external systems to control the state + of a mailbox. Using AMI actions or ARI resources, external + systems can programmatically trigger Message Waiting Indicators + (MWI) on subscribed phones. This is of particular use to those + who want to build their own VoiceMail application using ARI. +* ARI now supports the reception/transmission of out of call text + messages using any supported channel driver/protocol stack + through ARI. Users receive out of call text messages as JSON + events over the ARI websocket connection, and can send out of + call text messages using HTTP requests. +* The PJSIP stack now supports RFC 4662 Resource Lists, allowing + Asterisk to act as a Resource List Server. This includes defining lists of presence state, mailbox state, or lists of presence state/mailbox state; managing subscriptions to lists; and batched delivery of NOTIFY requests to subscribers. +* The PJSIP stack can now be used as a means of distributing device + state or mailbox state via PUBLISH requests to other Asterisk + instances. This is analogous to Asterisk's clustering support + using XMPP or Corosync; unlike existing clustering mechanisms, + using the PJSIP stack to perform the distribution of state does + not rely on another daemon or server to perform the work. + +------------------------------------------------------------------- +Fri Aug 22 10:29:26 UTC 2014 - jengelh@inai.de + +- Update to new upstream maintenance release 12.5.0 +* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.5.0-summary.txt + +------------------------------------------------------------------- +Sun Jul 13 00:07:28 UTC 2014 - jengelh@inai.de + +- Update to new upstream release 12.4.0 +* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.4.0-summary.txt +- Reenable SS7 support in chan_dahdi (for libss7-2.0) + +------------------------------------------------------------------- +Fri Jun 27 23:49:07 UTC 2014 - jengelh@inai.de + +- Update to new upstream release 12.3.2 +* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.3.2-summary.txt + +------------------------------------------------------------------- +Wed Apr 23 17:16:00 UTC 2014 - marcelloceschia@users.sourceforge.net + +- Update to new upstream release 12.2.0 +* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.2.0-summary.txt + +------------------------------------------------------------------- +Thu Mar 22 22:32:00 UTC 2014 - marcelloceschia@users.sourceforge.net + +- Update to new upstream release 12.1.1 (security release) +* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.1.1-summary.txt + +------------------------------------------------------------------- +Thu Mar 6 07:34:00 UTC 2014 - jengelh@inai.de + +- Update to new upstream release 12.1.0 (bugfix release) +* http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12.1.0-summary.txt + +------------------------------------------------------------------- +Tue Dec 24 10:33:13 UTC 2013 - jengelh@inai.de + +- Update to new upstream release 12.0.0 +* A more flexible bridging core based on the Bridging API +* A new internal message bus, Stasis +* Major standardization and consistency improvements to AMI +* Addition of the Asterisk REST Interface (ARI) +* A new SIP channel driver, chan_pjsip +* https://wiki.asterisk.org/wiki/display/AST/New+in+12 + +------------------------------------------------------------------- +Tue Dec 24 10:29:52 UTC 2013 - jengelh@inai.de + +- Update to new upstream release 11.7.0 (bugfix release) +* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.7.0-summary.html + for details + +------------------------------------------------------------------- +Sun Nov 24 11:57:30 UTC 2013 - jengelh@inai.de + +- Update to new upstream release 11.6.0 (bugfix release) +* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.6.0-summary.html + for details + +------------------------------------------------------------------- +Thu Aug 15 14:56:19 UTC 2013 - jengelh@inai.de + +- Use libuuid to reenable res_rtp_asterisk + +------------------------------------------------------------------- +Thu Aug 8 11:32:21 UTC 2013 - jengelh@inai.de + +- Update to new upstream release 11.5.0 (bugfix release) +* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.5.0-summary.html + for details + +------------------------------------------------------------------- +Sun Jun 2 23:05:18 UTC 2013 - jengelh@inai.de + +- Update to new upstream release 11.4.0 (bugfix release) +* See http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.4.0-summary.html + for details + +------------------------------------------------------------------- +Sun Feb 17 09:27:46 UTC 2013 - jengelh@inai.de + +- Enable building res_corosync (replaces res_ais from asterisk-10) +- Order asterisk after network (bnc#796148) + +------------------------------------------------------------------- +Sat Feb 16 02:32:13 UTC 2013 - jengelh@inai.de + +- Enable building chan_ooh323 +- Put config sample files into their respective subpackages +- Split off asterisk-freetds +- Make libasteriskssl.so symlink point to actual file +- Call ldconfig for libasteriskssl1 + +------------------------------------------------------------------- +Thu Jan 24 18:02:07 UTC 2013 - jengelh@inai.de + +- Update to new upstream release 11.2.1 (bugfix release) +* Fixed stuck DTMF when using ChannelRedirect to split a two + channel bridge +* Asterisk deadlocked during startup with mutex errors +* Resolved segfault in chan_sip while performing connected line + update + +------------------------------------------------------------------- +Fri Dec 21 22:50:50 UTC 2012 - joop.boonen@opensuse.org + +- Update to new upstream release 11.1.0 +* chan_local: Fix local_pvt ref leak in local_devicestate(). +* Fix a SIP request memory leak with TLS connections. +* Fix a bug where our Motif ICE candidates were not quite proper, + and make us more forgiving. + +------------------------------------------------------------------- +Wed Dec 5 11:07:31 UTC 2012 - joop.boonen@opensuse.org + +- Update to new upstream release 11.0.1 +* Fix a bug which made ConfBridge not record conferences when the + record command was initiated from AMI/CLI commands +* Fix a bug causing SIP reloads to remove all entries from the registry +* Fix an issue with res_http_websocket where the chan_sip WebSocket + handler could not be registered. + +------------------------------------------------------------------- +Sat Nov 3 02:33:18 UTC 2012 - jengelh@inai.de + +- Update to new upstream release 11.0.0 +* WebRTC Support with WebSocket transport over SIP. +* DTLS-SRTP - A secure transport for RTP media streams used by + WebRTC and SIP endpoints. +* ICE, STUN and TURN – A set of related technologies for + establishing live media streams between software agents running + behind network address translators (NATs) and firewalls. ICE, + STUN and TURN have been incorporated into the Asterisk RTP engine. + +------------------------------------------------------------------- +Sun Apr 8 18:44:34 UTC 2012 - jengelh@medozas.de + +- Update to new upstream release 10.3.0 +* http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-10.3.0-summary.html +- Make /var/lib/asterisk writable, so that the sqlite db can + be automatically created +- Replace init script by something less convoluted; + also add a systemd service file (bnc#750762, bnc#750763) + +------------------------------------------------------------------- +Fri Mar 16 19:28:25 UTC 2012 - jengelh@medozas.de + +- Update to new upstream release 10.2.1 +* Fix AST-2012-002, AST-2012-003 security vulnerabilities + +------------------------------------------------------------------- +Sun Mar 11 21:54:37 UTC 2012 - jengelh@medozas.de + +- Update to new upstream release 10.2.0 +* http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-10.2.0-summary.html +- Restore spandsp support (bnc#731943) +- Set permissions on files (bnc#750761) + +------------------------------------------------------------------- +Wed Feb 1 15:10:07 UTC 2012 - jengelh@medozas.de + +- Update to new upstream release 10.1.0 +* http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/asterisk-10.1.0-summary.html +- Add autotools BuildRequires for factory/12.2 + +------------------------------------------------------------------- +Fri Dec 16 00:06:31 UTC 2011 - jengelh@medozas.de + +- Update to final 10.0.0 + +------------------------------------------------------------------- +Sat Oct 8 22:12:16 UTC 2011 - jengelh@medozas.de + +- New package, for a change list see + https://wiki.asterisk.org/wiki/display/AST/New+in+10 diff --git a/asterisk.init b/asterisk.init new file mode 100644 index 0000000..a77280b --- /dev/null +++ b/asterisk.init @@ -0,0 +1,44 @@ +#!/bin/sh +### BEGIN INIT INFO +# Provides: asterisk +# Required-Start: $network $syslog $named $local_fs $remote_fs +# Required-Stop: $network $syslog $named $local_fs $remote_fs +# Should-Start: dahdi misdn lcr wanrouter mysql postgresql +# Should-Stop: dahdi misdn lcr wanrouter mysql postgresql +# Default-Start: 3 5 +# Default-Stop: 0 1 2 6 +# Short-Description: Asterisk PBX +# Description: the Asterisk Open Source PBX +### END INIT INFO +. /etc/rc.status +pid="/var/run/asterisk/asterisk.pid"; +dmn="/usr/sbin/asterisk"; +case "$1" in +start) + echo -en "Starting asterisk "; + startproc -sp "$pid" "$dmn" -fn; + rc_status -v;; +stop) + echo -en "Stopping asterisk "; + killproc -p "$pid" "$dmn"; + rc_status -v;; +status) + echo -en "Checking for asterisk "; + checkproc -p "$pid" "$dmn"; + rc_status -v;; +reload) + echo -en "Reloading asterisk "; + killproc -p "$pid" -HUP "$dmn"; + rc_status -v;; +restart) + "$0" stop; + "$0" start; + rc_status;; +try-restart) + "$0" status >/dev/null && "$0" restart; + rc_status;; +*) + echo "Usage: $0 {start|stop|status|reload|restart|try-restart}"; + exit 1;; +esac; +rc_exit; diff --git a/asterisk.keyring b/asterisk.keyring new file mode 100644 index 0000000..45ca1f8 --- /dev/null +++ b/asterisk.keyring @@ -0,0 +1,265 @@ +pub 1024D/F1DDCF73 2011-11-15 +uid Matthew Jordan +sub 1024g/51E145E3 2011-11-15 + +pub 1024D/2D557FFB 2012-01-24 +uid Kinsey Moore +sub 1024g/BFB53077 2012-01-24 + +pub 1024D/861FFB7D 2012-09-11 +uid Rusty Newton +sub 1024g/A8B3A5BB 2012-09-11 + +pub 1024D/7BD982D8 2010-08-30 +uid Richard Mudgett +sub 1024g/253C5733 2010-08-30 + +pub 1024D/E8F94236 2010-09-15 +uid Erin Spiceland +sub 1024g/C759348A 2010-09-15 + +pub 3072R/96A4EC20 2010-09-23 +uid David M. Lee (Digium) +sub 3072R/BC44A064 2010-09-23 + +pub 1024D/777DCC45 2013-03-27 +uid Kevin Harwell +sub 1024g/D520EE46 2013-03-27 + +pub 1024D/CD5D508E 2011-06-28 +uid Jonathan Rose +sub 1024g/338B8924 2011-06-28 + +pub 1024D/EA9A7B8A 2013-08-27 +uid Jonathan Rose +sub 1024g/7DB7048C 2013-08-27 + +pub 2048R/6B940F89 2013-12-16 +uid Joshua Colp +uid Joshua Colp +sub 2048R/8424242D 2013-12-16 + +-----BEGIN PGP PUBLIC KEY BLOCK----- +Version: GnuPG v2.0.22 (GNU/Linux) + +mQGiBE7Cs1wRBACKAIjLS7px9u5GaxPD6EliEEs9UfVPu9BalptM4UhYcYlxX8lY +9QUA7nB1pTMLsajxsvoJnA2m3yICRL0QpAWSyTJs98x6jtMUZWgZDhR1CWKgE8kW +O6wCCInXqH8T7AQ9f4NqyhaYhw41XP/tVmvd1FzERfXPFstoL8NpXEUsgwCgwZKv +odmpcpjnWGLOiQQlUP1QZG8D/A9il4Wsmwli6aSpIgR+xgYa/SigxgXf09Bv3r4p +7q8sMNsgH/AwKfWaHLyERU7kTgChvd38ACY6EAkGeI/x6IyOO8CoP+a17QHBgP7+ +tvd4TuRndI3u0tWiX56iAc3A5JnAn/H/7Gk8jas98+QVCrDo7zKf+4zQmmbOmeOU +Y8gwA/93wRyufhjlsH8Q1hkY+M8P9/R1T/8wFv9o3AwaI1rer3tzdtIG4vjyNRre +gZinERkfRRKTn05O9n77DbQ5n5Fh/6EPMkvGiOJGZiBeo/giDxa/JlPvsCqDXY/l 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+=LdPV +-----END PGP PUBLIC KEY BLOCK----- diff --git a/asterisk.service b/asterisk.service new file mode 100644 index 0000000..fcc42eb --- /dev/null +++ b/asterisk.service @@ -0,0 +1,14 @@ +[Unit] +Description=Asterisk PBX +Wants=nss-lookup.target +Wants=network-online.target +After=network-online.target + +[Service] +PIDFile=/run/asterisk/asterisk.pid +ExecStart=/usr/sbin/asterisk -fn +ExecReload=/bin/kill -HUP $MAINPID + +[Install] +WantedBy=basic.target +Also=systemd-networkd-wait-online.service diff --git a/asterisk.spec b/asterisk.spec new file mode 100644 index 0000000..13a310c --- /dev/null +++ b/asterisk.spec @@ -0,0 +1,671 @@ +# +# spec file for package asterisk +# +# Copyright (c) 2019 SUSE LINUX Products GmbH, Nuernberg, Germany. +# Copyright (c) 2017-2020 LISA GmbH, Bingen, Germany. +# +# All modifications and additions to the file contributed by third parties +# remain the property of their copyright owners, unless otherwise agreed +# upon. The license for this file, and modifications and additions to the +# file, is the same license as for the pristine package itself (unless the +# license for the pristine package is not an Open Source License, in which +# case the license is the MIT License). An "Open Source License" is a +# license that conforms to the Open Source Definition (Version 1.9) +# published by the Open Source Initiative. + +# Please submit bugfixes or comments via http://bugs.opensuse.org/ +# + +%ifarch ppc ppc64 ppc64le +%bcond_with dahdi +%else +%bcond_without dahdi +%endif + +Name: asterisk +Version: 18.12.1 +Release: 0 +URL: http://asterisk.org/ +Summary: The Asterisk Open Source PBX +License: BSD-2-Clause and BSD-4-Clause or GPL-2.0 +Group: Productivity/Telephony/Servers + +#DL-URL: http://downloads.asterisk.org/pub/telephony/asterisk/ +Source: https://downloads.asterisk.org/pub/telephony/asterisk/releases/%name-%version.tar.gz +Source2: https://downloads.asterisk.org/pub/telephony/asterisk/releases/%name-%version.tar.gz.asc +Source3: %name.keyring +Source4: asterisk.service +Source5: asterisk.init +Source6: https://raw.githubusercontent.com/asterisk/third-party/master/pjproject/2.12/pjproject-2.12.tar.bz2 +Source8: https://raw.githubusercontent.com/asterisk/third-party/master/jansson/2.14/jansson-2.14.tar.bz2 +Patch1: asterisk-init.diff +Patch2: asterisk-configure-paths.diff +Patch3: asterisk-cflags.diff +Patch4: asterisk-rundir.diff +BuildRequires: autoconf >= 2.60 +BuildRequires: automake >= 1.9 +BuildRequires: wget +BuildRequires: bison +%if %{with dahdi} +BuildRequires: dahdi-linux-devel +BuildRequires: libtonezone-devel +%endif +BuildRequires: fdupes +BuildRequires: flex +BuildRequires: freetds-devel +BuildRequires: gcc-c++ +BuildRequires: graphviz +BuildRequires: imap-devel +BuildRequires: libedit-devel +BuildRequires: libgsm-devel +BuildRequires: libmysqlclient-devel +BuildRequires: libpri-devel +BuildRequires: libsrtp-devel +BuildRequires: libss7-devel +BuildRequires: libtool +BuildRequires: mISDNuser-devel +BuildRequires: ncurses-devel +BuildRequires: net-snmp-devel +BuildRequires: openldap2-devel +BuildRequires: openr2-devel +# It seems to also want the CLI interface of openssl +BuildRequires: openssl +BuildRequires: osp-toolkit-devel +BuildRequires: pkgconfig +BuildRequires: popt-devel +BuildRequires: postgresql-devel +%if (0%{?sle_version} >= 150200 && 0%{?is_opensuse}) || 0%{?suse_version} > 1500 +BuildRequires: postgresql-server-devel +%endif +BuildRequires: libv4l-devel +BuildRequires: rubber +BuildRequires: termcap +BuildRequires: texlive +BuildRequires: xz +BuildRequires: pkgconfig(liburiparser) +%define with_corosync 1 +BuildRequires: pkgconfig(SDL_image) +BuildRequires: pkgconfig(alsa) +BuildRequires: pkgconfig(bluez) +BuildRequires: pkgconfig(gmime-3.0) +BuildRequires: pkgconfig(gtk+-2.0) +BuildRequires: pkgconfig(iksemel) +BuildRequires: pkgconfig(jack) +BuildRequires: pkgconfig(libavcodec) +BuildRequires: pkgconfig(libcfg) +BuildRequires: pkgconfig(libcpg) +BuildRequires: pkgconfig(libcrypto) +BuildRequires: pkgconfig(libcurl) +BuildRequires: pkgconfig(libical) +BuildRequires: pkgconfig(libiodbc) +BuildRequires: pkgconfig(libnewt) +BuildRequires: pkgconfig(libusb) +BuildRequires: pkgconfig(libxml-2.0) +BuildRequires: pkgconfig(libxslt) +BuildRequires: pkgconfig(lua) +BuildRequires: pkgconfig(neon) +BuildRequires: pkgconfig(ogg) +BuildRequires: pkgconfig(openssl) +BuildRequires: pkgconfig(portaudio-2.0) +BuildRequires: pkgconfig(sdl) +BuildRequires: pkgconfig(sdl2) +BuildRequires: pkgconfig(spandsp) +BuildRequires: pkgconfig(speex) +BuildRequires: pkgconfig(speexdsp) +BuildRequires: pkgconfig(sqlite3) +BuildRequires: pkgconfig(uuid) +BuildRequires: pkgconfig(vorbis) +BuildRequires: pkgconfig(x11) +BuildRequires: pkgconfig(zlib) +BuildRequires: systemd-rpm-macros +Requires: asterisk-sounds-base +Requires(pre): pwdutils +Provides: bundled(libjansson) = 2.14 +Provides: bundled(pjproject) = 2.10 +Provides: bundled(pjsip) = 2.10 +# addons are included in main package since 1.8.x +Provides: asterisk-addons = %version-%release +Suggests: asterisk-doc = %version + +%define scdperm %%attr(0751,root,asterisk) %%verify(not mode) +%define scfperm %%attr(0640,root,asterisk) %%verify(not mode) %%config(noreplace) + +%description +Asterisk is a complete PBX in software. It provides all of the +features you would expect from a PBX and more. Asterisk does voice +over IP in three protocols and can interoperate with almost all +standards-based telephony equipment using relatively inexpensive +hardware. + +Asterisk provides voicemail services with directory, call +conferencing, interactive voice response, and call queuing. It has +support for three-way calling, caller ID services, ADSI, SIP, and +H.323 (as both client and gateway). + +Documentation is available on the Asterisk home page +(http://asterisk.org) and on the Asterisk wiki +(http://voip-info.org/wiki-Asterisk). + +%package -n libasteriskssl1 +Summary: Asterisk SSL initialization wrapper +Group: System/Libraries + +%description -n libasteriskssl1 +This Asterisk library is used to wrap various initialization and +shutdown functions from the libssl and libcrypto libraries provided +by OpenSSL. This is done so that Asterisk can ensure that these +functions do *not* get called by any modules that are loaded into +Asterisk, since they should only be called once in any single +process. + +%package -n libasteriskpj2 +Summary: Asterisk PJSIP library +Group: System/Libraries + +%description -n libasteriskpj2 +Asterisk-bundled version of the PJSIP library + +%package doc +Summary: Asterisk Open Source PBX documentation +Group: Productivity/Telephony/Servers + +%description doc +The Asterisk Open Source PBX documentation. + +%package alsa +Summary: Soundcard module for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description alsa +This package allows Asterisk to use a soundcard supported by ALSA as +a telephone. + +%package calendar +Summary: Calendar resource modules for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description calendar +The Asterisk Calendaring API aims to be a generic interface for +integrating Asterisk with various calendaring technologies. The goal +is to be able to support reading and writing of calendar events as +well as allowing notification of pending events through the Asterisk +dialplan. + +There are three calendaring modules that ship with Asterisk that +provide support for iCalendar, CalDAV, and Microsoft Exchange Server +calendars. All three modules support event notification. Both CalDAV +and Exchange support reading and writing calendars, while iCalendar +is a read-only format. + +%package console +Summary: Console channel module for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description console +The Asterisk Command Line Interface. + +In addition to being the console for Asterisk, the CLI also sports +several features that make it very helpful to use for obtaining +information and affecting system configuration. + +%package corosync +Summary: Corosync module for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description corosync +Corosync is an open source group messaging system typically used in +clusters, cloud computing, and other high availability environments. + +Using Corosync together with res_corosync allows events to be shared +amongst a local cluster of Asterisk servers. Specifically, the types +of events that may be shared include: + +* Device state +* Message Waiting Indication, or MWI (to allow voicemail to live on a + server that is different from where the phones are registered) + +%package dahdi +Summary: Telephony Hardware Module for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description dahdi +This module allows Asterisk to use telephony hardware that is +supported by the dahdi kernel drivers. Supported hardware ranges from +FXO and FXS cards over ISDN BRI cards to T1, and E1 cards with up to +four interfaces. + +See the dahdi package for details. + +%package devel +Summary: Development files for Asterisk +Group: Development/Libraries/C and C++ +Requires: asterisk = %version +Requires: libasteriskssl1 = %version + +%description devel +This package contains the header files and libraries necessary for +developing applications which use asterisk. + +Install asterisk-devel if you want to develop applications which +depend on asterisk. + +%package freetds +Summary: FreeTDS Call Detail Recording and Call Event Logging for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description freetds +This package allows Asterisk to use a FreeTDS database for CDR and +CEL. + +%package mobile +Summary: Mobile (BlueTooth) channel for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description mobile +Mobile (BlueTooth) channel for Asterisk. + +%package moh-base +Summary: MOH audio files for Asterisk +Group: Productivity/Telephony/Servers +BuildArch: noarch + +%description moh-base +MusicOnHold: Plays hold music specified by class. If omitted, the +default music source for the channel will be used. + +%package odbc +Summary: Database Module for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description odbc +This package allows Asterisk to use read configuration data from, and +write call logs to ODBC databases. + +%package pgsql +Summary: Database Module for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description pgsql +This package allows Asterisk to use read configuration data from, and +write call logs to PostgreSQL databases. + +%package mysql +Summary: Mysql Database Module for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description mysql +This package allows Asterisk to use read configuration data from, and +write call logs to MySQL databases. + +%package radius +Summary: Radius modules for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description radius +The RADIUS CEL Backend. It needs: +* FreeRADIUS server +* Radiusclient-ng library +* Asterisk PBX + +For more information, see the wiki at +https://wiki.asterisk.org/wiki/display/AST/RADIUS+CEL+Backend + +%package snmp +Summary: SNMP resource module for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description snmp +Rudimentary support for SNMP access to Asterisk is available. + +%package sounds-base +Summary: Sound files for Asterisk +License: CC-BY-SA-3.0 +Group: Productivity/Telephony/Servers +BuildArch: noarch + +%description sounds-base +The set of core Asterisk sounds. + +%package spandsp +Summary: SpanDSP plugin for Asterisk +Group: Productivity/Telephony/Servers +Requires: asterisk = %version + +%description spandsp +Spandsp is a library for Digital Signal Processing (DSP). DSP is used +to turn images into audio and back. In Asterisk, spandsp, is required +for sending and receiving faxes. + +%prep +%autosetup -p1 + +%build +# activate privilege separation +# activate languageprefix (needed by sounds package) +# the sad story about perl: -p doesn't consider missing files as failures +perl -i -pe ' + s{^;\s*runuser\s*=\s*\S+}{runuser = asterisk}; + s{^;\s*rungroup\s*=\s*\S+}{rungroup = asterisk}; + s{^;\s*languageprefix\s*=\s*\S+}{languageprefix = yes}; +' configs/samples/asterisk.conf.sample 2>&1 | grep -i 'no such file' && false + +./bootstrap.sh +# +# NB: +# [ 27s] checking for -march=native support... user CFLAGS present +# This is exactly what we want. +# +%configure \ + --with-pjproject-bundled=yes \ + --with-jansson-bundled=yes \ + NOISY_BUILD=yes \ + EXTERNALS_CACHE_DIR="%_sourcedir" +make menuselect.makeopts +menuselect/menuselect \ + --enable app_macro \ + --enable app_meetme \ + --enable app_mysql \ + --enable res_config_mysql \ + --enable cdr_mysql \ + --enable smsq \ + --enable chan_mobile +make NOISY_BUILD=1 # %{?_smp_mflags} + +%install +b="%buildroot" +%make_install DOWNLOAD="/bin/true" +make install-headers DESTDIR="$b" +make samples DESTDIR="$b" + +# F!@#$d up DESTDIR support. Do proper symlink. +rm -f "$b/%_libdir/libasteriskssl.so" +ln -s libasteriskssl.so.1 "$b/%_libdir/libasteriskssl.so" + +mkdir -p "$b/%_includedir" +install -pm0644 include/asterisk.h "$b/%_includedir/" +cp -rp include/asterisk "$b/%_includedir/" +rm "$b/%_includedir/asterisk/.gitignore" +rm "$b/%_includedir/asterisk/autoconfig.h.in" +rm -f "$b/%_includedir/asterisk/autoconfig.h.in~" + +# service +mkdir -p "$b/%_sbindir" +mkdir -p "$b/%_unitdir" +install -pm0644 "%{S:4}" "$b/%_unitdir/" +ln -sf /sbin/service "$b/%_sbindir/rcasterisk" + +mkdir -p "$b/%_datadir/%name" +cp -a configs "$b/%_datadir/%name/" + +fl_asterisk="$PWD/asterisk.files" +>"$fl_asterisk" +pushd "$b/%_sysconfdir/%name/" +for i in *; do + case "$i" in + (ais.conf|alsa.conf|calendar.conf|console.conf) :;; + (*_corosync.conf|*_dahdi.conf) :;; + (*_odbc.conf|*_pgsql.conf|*_mysql.conf|*_snmp.conf) :;; + (*_tds.conf) :;; + (*) echo "%%scfperm %%_sysconfdir/%%name/$i" >>"$fl_asterisk";; + esac +done +popd +pushd "$b/%_libdir/%name/modules/" +for i in *; do + case "$i" in + (*_ais.so|*_alsa.so|*_calendar*.so|*_console.so) :;; + (*_corosync.so|*_dahdi*.so) :;; + (*_odbc.so|*_pgsql.so|*_mysql.so|*_radius.so|*_snmp.so) :;; + (*_spandsp.so|*_tds.so) :;; + (*) echo "%_libdir/%name/modules/$i" >>"$fl_asterisk";; + esac +done +popd +pushd "$b/%_datadir/%name/configs/" +for i in *; do + j="${i%.sample}" + case "$j" in + (ais.conf|alsa.conf|calendar.conf|console.conf) :;; + (*_corosync.conf|*_dahdi.conf) :;; + (*_odbc.conf|*_pgsql.conf|*_mysql.conf|*_snmp.conf) :;; + (*_tds.conf) :;; + (*) echo "%_datadir/%name/configs/$i" >>"$fl_asterisk";; + esac +done +popd + +%fdupes %buildroot/%_prefix + +# don't package man files twice +rm doc/*.8 + +# avoid stray dahdi config +%if ! %{with dahdi} +rm %buildroot/%_sysconfdir/%name/chan_dahdi.conf +%endif + +%pre +%_bindir/getent group asterisk >/dev/null || \ + %_sbindir/groupadd -r asterisk || : +%_bindir/getent passwd asterisk >/dev/null || \ + %_sbindir/useradd -r -s /sbin/false -d /var/lib/asterisk \ + -M -c "Asterisk PBX" -g asterisk asterisk || : +%service_add_pre asterisk.service + +%post +%service_add_post asterisk.service + +%preun +%service_del_preun asterisk.service + +%postun +%service_del_postun asterisk.service + +%post -n libasteriskssl1 -p /sbin/ldconfig +%postun -n libasteriskssl1 -p /sbin/ldconfig +%post -n libasteriskpj2 -p /sbin/ldconfig +%postun -n libasteriskpj2 -p /sbin/ldconfig + +%files -f asterisk.files +%license COPYING LICENSE +%doc BUGS CHANGES ChangeLog README.md sample.call *.txt doc/README.txt +%scdperm %dir %_sysconfdir/asterisk +%_unitdir/asterisk.service +%_sbindir/astcanary +%_sbindir/astdb2bdb +%_sbindir/astdb2sqlite3 +%_sbindir/asterisk +%_sbindir/astgenkey +%_sbindir/astversion +%_sbindir/autosupport +%_sbindir/rasterisk +%_sbindir/rcasterisk +%_sbindir/safe_asterisk +%_sbindir/smsq +%dir %_libdir/asterisk +%dir %_libdir/asterisk/modules +%dir %_datadir/%name +%dir %_datadir/%name/configs +%dir %_datadir/%name/keys +%_datadir/%name/documentation +%_datadir/%name/firmware +%_datadir/%name/images +%_datadir/%name/phoneprov/ +%_datadir/%name/rest-api/ +%_datadir/%name/scripts/ +%_datadir/%name/static-http +%exclude %_datadir/%name/configs/samples/alsa.conf.sample +%exclude %_datadir/%name/configs/samples/calendar.conf.sample +%exclude %_datadir/%name/configs/samples/console.conf.sample +%exclude %_datadir/%name/configs/samples/res_corosync.conf.sample +%exclude %_datadir/%name/configs/samples/chan_dahdi.conf.sample +%exclude %_datadir/%name/configs/samples/*_tds.conf.sample +%exclude %_datadir/%name/configs/samples/*_odbc.conf.sample +%exclude %_datadir/%name/configs/samples/*_pgsql.conf.sample +%exclude %_datadir/%name/configs/samples/*_mysql.conf.sample +%exclude %_datadir/%name/configs/samples/*_snmp.conf.sample +%exclude %{_sysconfdir}/asterisk/chan_mobile.conf +%exclude %{_libdir}/asterisk/modules/*_mobile.so +%attr(0750,asterisk,asterisk) %_localstatedir/lib/%name +%attr(0750,asterisk,asterisk) %_localstatedir/spool/%name +%attr(0750,asterisk,asterisk) %dir %_localstatedir/log/asterisk +%attr(0750,asterisk,asterisk) %dir %_localstatedir/log/asterisk/cdr-csv +%attr(0750,asterisk,asterisk) %dir %_localstatedir/log/asterisk/cdr-custom +%attr(0750,asterisk,asterisk) %dir %_localstatedir/log/asterisk/cel-custom +%_mandir/man8/* + +%files doc +%doc doc/* *.txt *.html + +%files -n libasteriskssl1 +%_libdir/libasteriskssl.so.* + +%files -n libasteriskpj2 +%_libdir/libasteriskpj.so.* + +%files alsa +%scdperm %dir %_sysconfdir/%name +%scfperm %_sysconfdir/%name/alsa.conf +%dir %_libdir/%name +%dir %_libdir/%name/modules +%_libdir/%name/modules/*_alsa.so +%dir %_datadir/%name +%dir %_datadir/%name/configs/samples +%_datadir/%name/configs/samples/alsa.conf.sample + +%files calendar +%scdperm %dir %_sysconfdir/%name +%scfperm %_sysconfdir/%name/calendar.conf +%dir %_libdir/%name +%dir %_libdir/%name/modules +%_libdir/%name/modules/*_calendar*.so +%dir %_datadir/%name +%dir %_datadir/%name/configs/samples +%_datadir/%name/configs/samples/calendar.conf.sample + +%files console +%scdperm %dir %_sysconfdir/%name +%scfperm %_sysconfdir/%name/console.conf +%dir %_libdir/%name +%dir %_libdir/%name/modules +%_libdir/%name/modules/*_console.so +%dir %_datadir/%name +%dir %_datadir/%name/configs/samples +%_datadir/%name/configs/samples/console.conf.sample + +%if 0%{?with_corosync} +%files corosync +%scdperm %dir %_sysconfdir/%name +%scfperm %_sysconfdir/%name/res_corosync.conf +%dir %_libdir/%name +%dir %_libdir/%name/modules +#_libdir/name/modules/res_corosync.so +%dir %_datadir/%name +%dir %_datadir/%name/configs/samples +%_datadir/%name/configs/samples/res_corosync.conf.sample +%endif + +%if %{with dahdi} +%files dahdi +%scdperm %dir %_sysconfdir/%name +%scfperm %_sysconfdir/%name/chan_dahdi.conf +%dir %_libdir/%name +%dir %_libdir/%name/modules +%_libdir/%name/modules/*_dahdi* +%dir %_datadir/dahdi +%dir %_datadir/dahdi/span_config.d +%_datadir/dahdi/span_config.d/40-asterisk +%dir %_datadir/%name +%dir %_datadir/%name/configs/samples +%_datadir/%name/configs/samples/chan_dahdi.conf.sample +%endif + +%files devel +%_includedir/%{name}.h +%dir %_includedir/%{name} +%_includedir/%{name}/*.h +%dir %_includedir/%{name}/doxygen +%_includedir/%{name}/doxygen/*.h +%_libdir/libasteriskssl.so +%_libdir/libasteriskpj.so + +%files freetds +%scdperm %dir %_sysconfdir/%name +%scfperm %_sysconfdir/%name/*_tds.conf +%dir %_libdir/%name +%dir %_libdir/%name/modules +%_libdir/%name/modules/*_tds.so +%dir %_datadir/%name +%dir %_datadir/%name/configs/samples +%_datadir/%name/configs/samples/*_tds.conf.sample + +%files mobile +%attr(0640,asterisk,asterisk) %config(noreplace) %_sysconfdir/asterisk/chan_mobile.conf +%_libdir/asterisk/modules/*_mobile.so + +%files moh-base +%dir %_datadir/%name +%_datadir/%name/moh + +%files odbc +%scdperm %dir %_sysconfdir/%name +%scfperm %_sysconfdir/%name/*_odbc.conf +%dir %_libdir/%name +%dir %_libdir/%name/modules +%_libdir/%name/modules/*_odbc.so +%dir %_datadir/%name +%dir %_datadir/%name/configs/samples +%_datadir/%name/configs/samples/*_odbc.conf.sample + +%files pgsql +%scdperm %dir %_sysconfdir/%name +%scfperm %_sysconfdir/%name/*_pgsql.conf +%dir %_libdir/%name +%dir %_libdir/%name/modules +%_libdir/%name/modules/*_pgsql.so +%dir %_datadir/%name +%dir %_datadir/%name/configs/samples +%_datadir/%name/configs/samples/*_pgsql.conf.sample + +%files mysql +%scdperm %dir %_sysconfdir/%name +%scfperm %_sysconfdir/%name/*_mysql.conf +%dir %_libdir/%name +%dir %_libdir/%name/modules +%_libdir/%name/modules/*_mysql.so +%dir %_datadir/%name +%dir %_datadir/%name/configs/samples +%_datadir/%name/configs/samples/*_mysql.conf.sample + +%files sounds-base +%dir %_datadir/%name +%_datadir/%name/sounds + +%files snmp +%scdperm %dir %_sysconfdir/%name +%scfperm %_sysconfdir/%name/*_snmp.conf +%dir %_libdir/%name +%dir %_libdir/%name/modules +%_libdir/%name/modules/*_snmp.so +%dir %_datadir/%name +%dir %_datadir/%name/configs/samples +%_datadir/%name/configs/samples/*_snmp.conf.sample + +%files spandsp +%dir %_libdir/%name +%dir %_libdir/%name/modules +%_libdir/%name/modules/*_spandsp.so + +%changelog diff --git a/jansson-2.14.tar.bz2 b/jansson-2.14.tar.bz2 new file mode 100644 index 0000000..7cfb604 Binary files /dev/null and b/jansson-2.14.tar.bz2 differ diff --git a/pjproject-2.12.tar.bz2 b/pjproject-2.12.tar.bz2 new file mode 100644 index 0000000..c4637bf Binary files /dev/null and b/pjproject-2.12.tar.bz2 differ