jcollie pushed to asterisk (f22). "13.3.0"
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Mon Apr 6 15:33:27 UTC 2015
>From c82fbef34261a82f378d2ca69cb96c831dd39d29 Mon Sep 17 00:00:00 2001
From: "Jeffrey C. Ollie" <jeff at ocjtech.us>
Date: Wed, 1 Apr 2015 14:51:24 -0500
Subject: 13.3.0
diff --git a/asterisk.spec b/asterisk.spec
index 961f353..6d98647 100644
--- a/asterisk.spec
+++ b/asterisk.spec
@@ -48,7 +48,7 @@
Summary: The Open Source PBX
Name: asterisk
-Version: 13.1.1
+Version: 13.3.0
Release: 1%{?_rc:.rc%{_rc}}%{?_beta:.beta%{_beta}}%{?dist}
License: GPLv2
Group: Applications/Internet
@@ -1603,6 +1603,350 @@ fi
%{_libdir}/asterisk/modules/res_xmpp.so
%changelog
+* Wed Apr 1 2015 Jeffrey C. Ollie <jeff at ocjtech.us> - 13.3.0-1:
+- The Asterisk Development Team has announced the release of Asterisk 13.3.0.
+- This release is available for immediate download at
+- http://downloads.asterisk.org/pub/telephony/asterisk
+-
+- The release of Asterisk 13.3.0 resolves several issues reported by the
+- community and would have not been possible without your participation.
+- Thank you!
+-
+- The following are the issues resolved in this release:
+-
+- New Features made in this release:
+- -----------------------------------
+- * ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a
+- channel (Reported by Matt Jordan)
+- * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
+- (Reported by Dwayne Hubbard)
+-
+- Bugs fixed in this release:
+- -----------------------------------
+- * ASTERISK-24616 - Crash in res_format_attr_h264 due to invalid
+- string copy (Reported by Yura Kocyuba)
+- * ASTERISK-24748 - res_pjsip: If wizards explicitly configured in
+- sorcery.conf false ERROR messages may occur (Reported by Joshua
+- Colp)
+- * ASTERISK-24769 - res_pjsip_sdp_rtp: Local ICE candidates leaked
+- (Reported by Matt Jordan)
+- * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
+- res_odbc (Reported by ibercom)
+- * ASTERISK-24479 - Enable REF_DEBUG for module references
+- (Reported by Corey Farrell)
+- * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
+- fully disconnect underlying socket, leading to events being
+- dropped with no additional information (Reported by Matt Jordan)
+- * ASTERISK-24772 - ODBC error in realtime sippeers when device
+- unregisters under MariaDB (Reported by Richard Miller)
+- * ASTERISK-24752 - Crash in bridge_manager_service_req when bridge
+- is destroyed by ARI during shutdown (Reported by Richard
+- Mudgett)
+- * ASTERISK-24741 - dtls_handler causes Asterisk to crash (Reported
+- by Zane Conkle)
+- * ASTERISK-24015 - app_transfer fails with PJSIP channels
+- (Reported by Private Name)
+- * ASTERISK-24727 - PJSIP: Crash experienced during multi-Asterisk
+- transfer scenario. (Reported by Mark Michelson)
+- * ASTERISK-24771 - ${CHANNEL(pjsip)} - segfault (Reported by
+- Niklas Larsson)
+- * ASTERISK-24716 - Improve pjsip log messages for presence
+- subscription failure (Reported by Rusty Newton)
+- * ASTERISK-24612 - res_pjsip: No information if a required sorcery
+- wizard is not loaded (Reported by Joshua Colp)
+- * ASTERISK-24768 - res_timing_pthread: file descriptor leak
+- (Reported by Matthias Urlichs)
+- * ASTERISK-24685 - "pjsip show version" CLI command (Reported by
+- Joshua Colp)
+- * ASTERISK-24632 - install_prereq script installs pjproject
+- without IPv6 support (Reported by Rusty Newton)
+- * ASTERISK-24085 - Documentation - We should remove or further
+- document the 'contact' section in pjsip.conf (Reported by Rusty
+- Newton)
+- * ASTERISK-24791 - Crash in ast_rtcp_write_report (Reported by
+- JoshE)
+- * ASTERISK-24700 - CRASH: NULL channel is being passed to
+- ast_bridge_transfer_attended() (Reported by Zane Conkle)
+- * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
+- (Reported by Corey Farrell)
+- * ASTERISK-24799 - [patch] make fails with undefined reference to
+- SSLv3_client_method (Reported by Alexander Traud)
+- * ASTERISK-22670 - Asterisk crashes when processing ISDN AoC
+- Events (Reported by klaus3000)
+- * ASTERISK-24689 - Segfault on hangup after outgoing PRI-Euroisdn
+- call (Reported by Marcel Manz)
+- * ASTERISK-24740 - [patch]Segmentation fault on aoc-e event
+- (Reported by Panos Gkikakis)
+- * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
+- for playing back messages stored in IMAP - play_message: No
+- origtime (Reported by Graham Barnett)
+- * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
+- OSX with 64 bit integers (Reported by Corey Farrell)
+- * ASTERISK-24796 - Codecs and bucket schema's prevent module
+- unload (Reported by Corey Farrell)
+- * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
+- (Reported by Ashley Sanders)
+- * ASTERISK-24499 - Need more explicit debug when PJSIP dialstring
+- is invalid (Reported by Rusty Newton)
+- * ASTERISK-24785 - 'Expires' header missing from 200 OK on
+- REGISTER (Reported by Ross Beer)
+- * ASTERISK-24677 - ARI GET variable on channel provides unhelpful
+- response on non-existent variable (Reported by Joshua Colp)
+- * ASTERISK-24797 - bridge_softmix: G.729 codec license held
+- (Reported by Kevin Harwell)
+- * ASTERISK-24812 - ARI: Creating channels through /channels
+- resource always uses SLIN, which results in unneeded transcoding
+- (Reported by Matt Jordan)
+- * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
+- thread ID being passed to pthread_kill (Reported by JoshE)
+- * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
+- fail (Reported by Terry Wilson)
+- * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
+- SRTP for audio, but they responded without it' is ambiguous and
+- wrong in some cases (Reported by Rusty Newton)
+- * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
+- error response and BYE are sent to the caller (Reported by
+- Makoto Dei)
+- * ASTERISK-18105 - most of asterisk modules are unbuildable in
+- cygwin environment (Reported by feyfre)
+- * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
+- * ASTERISK-24751 - Integer values in json payload to ARI cause
+- asterisk to crash (Reported by jeffrey putnam)
+- * ASTERISK-24838 - chan_sip: Locking inversion occurs when
+- building a peer causes a peer poke during request handling
+- (Reported by Richard Mudgett)
+- * ASTERISK-24825 - Caller ID not recognized using
+- Centrex/Distinctive dialing (Reported by Richard Mudgett)
+- * ASTERISK-24830 - res_rtp_asterisk.c checks USE_PJPROJECT not
+- HAVE_PJPROJECT (Reported by Stefan Engström)
+- * ASTERISK-24840 - res_pjsip: conflicting endpoint identifiers
+- (Reported by Kevin Harwell)
+- * ASTERISK-24755 - Asterisk sends unexpected early BYE to
+- transferrer during attended transfer when using a Stasis bridge
+- (Reported by John Bigelow)
+- * ASTERISK-24739 - [patch] - Out of files -- call fails --
+- numerous files with inodes from under /usr/share/zoneinfo,
+- mostly posixrules (Reported by Ed Hynan)
+- * ASTERISK-23390 - NewExten Event with application AGI shows up
+- before and after AGI runs (Reported by Benjamin Keith Ford)
+- * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
+- voicemail stored in LDAP (Reported by Graham Barnett)
+- * ASTERISK-24808 - res_config_odbc: Improper escaping of
+- backslashes occurs with MySQL (Reported by Javier Acosta)
+- * ASTERISK-24807 - Missing mandatory field Max-Forwards (Reported
+- by Anatoli)
+- * ASTERISK-20850 - [patch]Nested functions aren't portable.
+- Adapting RAII_VAR to use clang/llvm blocks to get the
+- same/similar functionality. (Reported by Diederik de Groot)
+- * ASTERISK-24872 - [patch] AMI PJSIPShowEndpoint closes AMI
+- connection on error (Reported by Dmitriy Serov)
+- * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
+- by Frank DiGennaro)
+- * ASTERISK-21038 - Bad command completion of "core set debug
+- channel" (Reported by Richard Kenner)
+- * ASTERISK-18708 - func_curl hangs channel under load (Reported by
+- Dave Cabot)
+- * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
+- Atis Lezdins)
+- * ASTERISK-24876 - Investigate reference leaks from
+- tests/channels/local/local_optimize_away (Reported by Corey
+- Farrell)
+- * ASTERISK-24882 - chan_sip: Improve usage of REF_DEBUG (Reported
+- by Corey Farrell)
+- * ASTERISK-24817 - init_logger_chain: unreachable code block
+- (Reported by Corey Farrell)
+- * ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by
+- snuffy)
+- * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
+- under OpenBSD (Reported by snuffy)
+-
+- Improvements made in this release:
+- -----------------------------------
+- * ASTERISK-24745 - [patch]Add no_answer to ARI hangup causes
+- (Reported by Ben Merrills)
+- * ASTERISK-24811 - asterisk-publication sorcery object does not
+- use realtime (Reported by Matt Hoskins)
+- * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
+- Couldn't find mailbox %s in context (Reported by Graham Barnett)
+-
+- For a full list of changes in this release, please see the ChangeLog:
+-
+- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.3.0
+
+* Wed Apr 1 2015 Jeffrey C. Ollie <jeff at ocjtech.us> - 13.2.0-1:
+- The Asterisk Development Team has announced the release of Asterisk 13.2.0.
+- This release is available for immediate download at
+- http://downloads.asterisk.org/pub/telephony/asterisk
+-
+- The release of Asterisk 13.2.0 resolves several issues reported by the
+- community and would have not been possible without your participation.
+- Thank you!
+-
+- The following are the issues resolved in this release:
+-
+- Bugs fixed in this release:
+- -----------------------------------
+- * ASTERISK-24342 - PJSIP: Qualifying endpoints attempts to do them
+- all at the same time. (Reported by Richard Mudgett)
+- * ASTERISK-24514 - res_pjsip_outbound_registration: stack overflow
+- when using non-default sorcery wizard (Reported by Kevin
+- Harwell)
+- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
+- from JSSIP (Reported by Badalian Vyacheslav)
+- * ASTERISK-24607 - res_pjsip_session: re-INVITE with declined
+- media streams results in 488 (Reported by Matt Jordan)
+- * ASTERISK-24563 - Direct Media calls within private network
+- sometimes get one way audio (Reported by Kevin Harwell)
+- * ASTERISK-24604 - res_rtp_asterisk: Crash during restart due to
+- race condition in accessing codec in stored ast_frame and codec
+- core (Reported by Matt Jordan)
+- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
+- enabled (Reported by Richard Mudgett)
+- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
+- enabled (Reported by Andreas Steinmetz)
+- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
+- casts char to unsigned int (Reported by Walter Doekes)
+- * ASTERISK-24536 - AMI redirect with PJSIP fails to move extra
+- channel (Reported by Niklas Larsson)
+- * ASTERISK-24459 - bridge_native_rtp: Native RTP bridging is
+- chosen for RTP compatible channels when the DTMF mode is not
+- compatible (Reported by Yaniv Simhi)
+- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
+- level - 'Remote address is null, most likely RTP has been
+- stopped' (Reported by Rusty Newton)
+- * ASTERISK-24513 - Local channel apparently leaked in off-nominal
+- DTMF attended transfer (Reported by Mark Michelson)
+- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
+- on startup (Reported by Richard Kenner)
+- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
+- destination when 'sendrpid=yes' (in proxy environment) (Reported
+- by Karsten Wemheuer)
+- * ASTERISK-23841 - DTMF atxfer doesn't set CallerID for the recall
+- calls to the transferrer. (Reported by Richard Mudgett)
+- * ASTERISK-24376 - res_pjsip_refer: REFER request for remote
+- session attempts to direct channel to external_replaces
+- extension instead of context, without providing for the
+- Referred-To SIP URI (Reported by Matt Jordan)
+- * ASTERISK-24591 - Stasis() side of an ARI originated channel
+- cannot be Redirected (Reported by Kinsey Moore)
+- * ASTERISK-24049 - Asterisk Manager Interface: A number of list
+- type responses aren't using astman_send_listack (Reported by
+- Jonathan Rose)
+- * ASTERISK-24637 - Channel re-enters Stasis() when it should not
+- (Reported by John Bigelow)
+- * ASTERISK-24474 - sip_to_pjsip.py lacks documentation and does
+- not function (Reported by John Kiniston)
+- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
+- (Reported by Kristian Høgh)
+- * ASTERISK-20744 - [patch] Security event logging does not work
+- over syslog (Reported by Michael Keuter)
+- * ASTERISK-24665 - Configure check required for
+- pjsip_get_dest_info() (Reported by Mark Michelson)
+- * ASTERISK-23850 - Park Application does not respect Return
+- Context Priority (Reported by Andrew Nagy)
+- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
+- in the CFlags returned (Reported by Diederik de Groot)
+- * ASTERISK-24655 - res_pjsip_outbound_publish: Hang on shutdown
+- while attempting to publish (Reported by Kevin Harwell)
+- * ASTERISK-24485 - res_pjsip cannot be unloaded or shutdown
+- (Reported by Corey Farrell)
+- * ASTERISK-24663 - [patch] Unnamed semaphore autoconf check fails
+- on cross compilation (Reported by abelbeck)
+- * ASTERISK-24624 - Transfer to invalid extension results in hung
+- channel. (Reported by Zane Conkle)
+- * ASTERISK-24615 - When Multiple Transports Exist in pjsip.conf,
+- Incorrect External Addresses is Used in SIP Packets When
+- Responding to INVITE (Reported by David Justl)
+- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
+- voicemail is not deleted after review, hangup (Reported by LEI
+- FU)
+- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
+- 32-bit packages on 64-bit hosts (Reported by Ben Klang)
+- * ASTERISK-24600 - Stuck IAX channels, Asterisk stops responding
+- to most traffic, potential deadlock (Reported by Jeff Collell)
+- * ASTERISK-24560 - Creating a named ARI bridge twice causes a
+- crash (Reported by Kinsey Moore)
+- * ASTERISK-24682 - app_dial: Multiple DialEnd events emitted when
+- MACRO_RESULT or GOSUB_RESULT are an unexpected value (Reported
+- by Matt Jordan)
+- * ASTERISK-24640 - Registration pending stays forever after sip
+- reload (Reported by Max Man)
+- * ASTERISK-24673 - outgoing sip registers cannot be removed or
+- modified without doing restart (or doing module unload
+- chan_sip.so) (Reported by Stefan Engström)
+- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
+- m() option does not queue an MWI event (Reported by Gareth
+- Palmer)
+- * ASTERISK-24649 - Pushing of channel into bridge fails; Stasis
+- fails to get app name (Reported by John Bigelow)
+- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
+- column comparison for 'defaultuser' (Reported by
+- HZMI8gkCvPpom0tM)
+- * ASTERISK-24693 - Investigate and fix memory leaks in Asterisk
+- (Reported by Kevin Harwell)
+- * ASTERISK-24626 - Voicemail passwords not being stored in ARA
+- (Reported by Paddy Grice)
+- * ASTERISK-24539 - Compile fails on OSX because of sem_timedwait
+- in bridge_channel.c (Reported by George Joseph)
+- * ASTERISK-24544 - Compile fails on OSX Yosemite because of
+- incorrect detection of htonll and ntohll (Reported by George
+- Joseph)
+- * ASTERISK-24723 - confbridge: CLI command 'confbridge list XXXX'
+- no longer displays user menus (Reported by Matt Jordan)
+- * ASTERISK-24721 - manager: ModuleLoad action incorrectly reports
+- 'module not found' during a Reload operation (Reported by Matt
+- Jordan)
+- * ASTERISK-24719 - ConfBridge recording channels get stuck when
+- recording started/stopped more than once (Reported by Richard
+- Mudgett)
+- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
+- by Kevin Harwell)
+- * ASTERISK-24728 - tcptls: Bad file descriptor error when
+- reloading chan_sip (Reported by Kevin Harwell)
+- * ASTERISK-24729 - Outbound registration not occuring on new
+- registrations after reload. (Reported by Richard Mudgett)
+- * ASTERISK-24676 - Security Vulnerability: URL request injection
+- in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
+- * ASTERISK-24666 - Security Vulnerability: RTP not closed after
+- sip call using unsupported codec (Reported by Y Ateya)
+- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
+- versions (Reported by Jared Biel)
+- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
+- Stephan Eisvogel)
+- * ASTERISK-24736 - Memory Leak Fixes (Reported by Mark Michelson)
+- * ASTERISK-24635 - PJSIP outbound PUBLISH crashes when no response
+- is ever received (Reported by Marco Paland)
+- * ASTERISK-24737 - When agent not logged in, agent status shows
+- unavailable, queue status shows agent invalid (Reported by
+- Richard Mudgett)
+-
+- Improvements made in this release:
+- -----------------------------------
+- * ASTERISK-24552 - ARI: Allow associating a channel as an
+- initiator of an Origination for record keeping purposes
+- (Reported by Matt Jordan)
+- * ASTERISK-24553 - ARI/AMI: Include language in standard channel
+- snapshot output (Reported by Matt Jordan)
+- * ASTERISK-24643 - res_pjsip: Add user=phone option (Reported by
+- Matt Jordan)
+- * ASTERISK-24644 - res_pjsip_keepalive: Add keepalive module for
+- connection-oriented transports. (Reported by Matt Jordan)
+- * ASTERISK-24412 - [patch]Incomplete channel originate/continue
+- handling with ARI (Reported by Nir Simionovich (GreenfieldTech -
+- Israel))
+- * ASTERISK-24678 - [PATCH] Added atxfer* settings to
+- features.conf.sample (Reported by Niklas Larsson)
+- * ASTERISK-24575 - [patch]Make capath work for res_pjsip (Reported
+- by cloos)
+- * ASTERISK-24671 - Missing docs for the CDR AMI Event (Reported by
+- Dan Jenkins)
+- * ASTERISK-24316 - For httpd server, need option to define server
+- name for security purposes (Reported by Andrew Nagy)
+-
+- For a full list of changes in this release, please see the ChangeLog:
+-
+- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.2.0
+
* Fri Jan 30 2015 Jeffrey C. Ollie <jeff at ocjtech.us> - 13.1.1-1:
- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
diff --git a/sources b/sources
index 3ee52d7..427560a 100644
--- a/sources
+++ b/sources
@@ -1,2 +1,2 @@
-a2781693a67e008d2a3c60b756d4d4ab asterisk-13.1.1.tar.gz
-dd4ad67d0be884750b2757455d62b938 asterisk-13.1.1.tar.gz.asc
+95b4850668b8880e6c4bfefae4fb427c asterisk-13.3.0.tar.gz
+e281295e59bb243b14e126e64e7a0810 asterisk-13.3.0.tar.gz.asc
--
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