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List: asterisk-dev
Subject: [asterisk-dev] Asterisk 13.17.0-rc1 Now Available
From: "Asterisk Development Team" <asteriskteam () digium ! com>
Date: 2017-07-06 14:45:27
Message-ID: E1dT823-0003FC-Q2 () mail ! digium ! com
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The Asterisk Development Team would like to announce the first
release candidate of Asterisk 13.17.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.17.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release candidate:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-25665 - Duplicate logging in queue log for EXITEMPTY
events
(Reported by Ove Aursand)
* ASTERISK-27074 - core_local: local channel data not being
properly unref'ed and unlocked
(Reported by Kevin Harwell)
* ASTERISK-27075 - bridge: stuck channel(s) after failed
attended transfer
(Reported by Kevin Harwell)
* ASTERISK-24052 - app_voicemail reloads result in leaked IMAP
sockets.
(Reported by Louis Jocelyn Paquet)
* ASTERISK-27060 - Comment typo format_g729.c
(Reported
by Matthew Fredrickson)
* ASTERISK-27026 - res_ari: Crash when no ari.conf
configuration file exists
(Reported by Ronald Raikes)
* ASTERISK-27041 - Core/PBX: [patch] Deadlock between dialplan
execution and application unregistration
(Reported by
Frederic LE FOLL)
* ASTERISK-27057 - Seg Fault in ast_sorcery_object_get_id at
sorcery.c
(Reported by Ryan Smith)
* ASTERISK-27024 - nat/external_media settings ignored in
14.4.1
(Reported by Christopher van de Sande)
* ASTERISK-27046 - res_pjsip_transport_websocket: segfault in
get_write_timeout
(Reported by Jørgen H)
* ASTERISK-27022 - res_rtp_asterisk: Incorrect SSRC change for
RTCP component
(Reported by Michael Walton)
* ASTERISK-26923 - bridging: T.38 request is lost when channels
are added to bridge
(Reported by Torrey Searle)
* ASTERISK-27053 - res_pjsip_refer/session: Calls dropped
during transfer
(Reported by Kevin Harwell)
* ASTERISK-27052 - Asterisk build process fails with flag
--with-pjproject-bundled with curl download command and slow
network
(Reported by alex)
* ASTERISK-27039 - chan_pjsip: Device state is idle when
channel from endpoint is in early media
(Reported by
Joshua Colp)
* ASTERISK-26996 - chan_pjsip: Flipping between codecs
(Reported by Michael Maier)
* ASTERISK-26281 - chan_pjsip would send INVITE to
'Unreachable' endpoints
(Reported by Jacek Konieczny)
* ASTERISK-26973 - bridge: Crash when freeing frame and
snooping
(Reported by Michel R. Vaillancourt)
* ASTERISK-19291 - Background in realtime
(Reported by
Andrew Nowrot)
* ASTERISK-27025 - channel / meetme: Fix missing parentheses
(Reported by Joshua Colp)
* ASTERISK-27021 - GET /recordings/stored returns 500 Internal
Server Error
(Reported by Tim Morgan)
* ASTERISK-24858 - [patch]Asterisk 13 PJSIP sends RTP packets
in wrong byte order on Intel platform when using slin codec
(Reported by Frankie Chin)
* ASTERISK-23951 - Asterisk attempts and fails to build
format_mp3 even if mp3lib was not downloaded
(Reported by
Tzafrir Cohen)
* ASTERISK-25294 - srtp's crypto_get_random deprecated
(Reported by Tzafrir Cohen)
* ASTERISK-23839 - AGI - RECORD FILE - documentation doesn't
describe BEEP argument
(Reported by Rusty Newton)
* ASTERISK-22432 - Async AGI crashes Asterisk when issuing "set
variable" command without args
(Reported by Antoine
Pitrou)
* ASTERISK-25662 - Malformed AGI 520 Usage response
(Reported by Tony Mountifield)
* ASTERISK-25101 - DTLS configuration can not be specified in
the general section - documentation
(Reported by Ben
Langfeld)
* ASTERISK-27008 - res_format_attr_h264: SDP parse fails if
fmtp optional parameters have a space
(Reported by John
Harris)
* ASTERISK-26399 - app_queue: Agent not called when caller is
parked
(Reported by wushumasters)
* ASTERISK-26400 - app_queue: Queue member stops being called
after AMI "Redirect" action for queues with wrapuptime
(Reported by Etienne Lessard)
* ASTERISK-26715 - app_queue: Member will not receive any new
calls after doing a transfer if wrapuptime = greater than 0 and
using Local channel
(Reported by David Brillert)
* ASTERISK-26975 - app_queue: Non-zero wrapup time can cause
agents not to receive queue calls after transfer queue call
(Reported by Lorne Gaetz)
* ASTERISK-27012 - app_confbridge: ConfBridge sometimes does
not play user name recording while leaving
(Reported by
Robert Mordec)
* ASTERISK-26979 - res_rtp_asterisk: SRTP unprotect failed with
authentication failure 10 or 110
(Reported by Javier
Riveros )
* ASTERISK-26982 - chan_sip: rtcp_mux setting may cause ice
completion failure/delay if client offers rtcp-mux as
negotiable
(Reported by Stefan Engström)
* ASTERISK-26964 - res_pjsip_session: Wrong From on reinvite
when request and To URI differ
(Reported by Yasin CANER)
* ASTERISK-26789 - Audit manipulation of channel flags without
locks
(Reported by Joshua Colp)
* ASTERISK-26333 - Problems with Blind Transfer, PJSIP (Aastra
6869i)
(Reported by Matthias Binder)
Information Requests made in this release:
-----------------------------------
* ASTERISK-26976 - libsrtp-2.x.x support
(Reported by
Alex)
Improvements made in this release:
-----------------------------------
* ASTERISK-26230 - [patch] res_pjsip_mwi: unsolicited mwi could
block PJSIP taskprocessor on startup
(Reported by Alexei
Gradinari)
* ASTERISK-27043 - Core/BuildSystem: Add defines to fix build
with LibreSSL
(Reported by Guido Falsi)
* ASTERISK-27042 - Unpatched asterisk sources fail to build on
FreeBSD due to missing crypt.h file
(Reported by Guido
Falsi)
* ASTERISK-26419 - audiohooks: Remove redundant codec
translations when using audiohooks
(Reported by Michael
Walton)
* ASTERISK-26124 - res_agi: Set audio format for EAGI audio
stream
(Reported by John Fawcett)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0-rc1
Thank you for your continued support of Asterisk!
[Attachment #5 (text/html)]
<html><head></head><body>
The Asterisk Development Team would like to announce the first
release candidate of Asterisk 13.17.0.<br>
This release candidate is available for immediate download at <br>
<a href='http://downloads.asterisk.org/pub/telephony/asterisk'>http://downloads.asterisk.org/pub/telephony/asterisk</a>
<p>
The release of Asterisk 13.17.0-rc1 resolves several issues reported by the<br>
community and would have not been possible without your participation.<br>
<p>
<b>Thank you!</b><br>
<p>
The following issues are resolved in this release candidate:<br>
<p>
<b>Bugs fixed in this release:</b><br>
-----------------------------------<br>
<table border=0 padding=3>
<tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-25665'>ASTERISK-25665</a>] - \
<td><td>Duplicate logging in queue log for EXITEMPTY events<br>(Reported by Ove \
Aursand)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27074'>ASTERISK-27074</a>] - \
<td><td>core_local: local channel data not being properly unref'ed and \
unlocked<br>(Reported by Kevin Harwell)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27075'>ASTERISK-27075</a>] - \
<td><td>bridge: stuck channel(s) after failed attended transfer<br>(Reported by Kevin \
Harwell)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-24052'>ASTERISK-24052</a>] - \
<td><td>app_voicemail reloads result in leaked IMAP sockets.<br>(Reported by Louis \
Jocelyn Paquet)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27060'>ASTERISK-27060</a>] - \
<td><td>Comment typo format_g729.c<br>(Reported by Matthew \
Fredrickson)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27026'>ASTERISK-27026</a>] - \
<td><td>res_ari: Crash when no ari.conf configuration file exists<br>(Reported by \
Ronald Raikes)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27041'>ASTERISK-27041</a>] - \
<td><td>Core/PBX: [patch] Deadlock between dialplan execution and application \
unregistration<br>(Reported by Frederic LE FOLL)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27057'>ASTERISK-27057</a>] - \
<td><td>Seg Fault in ast_sorcery_object_get_id at sorcery.c<br>(Reported by Ryan \
Smith)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27024'>ASTERISK-27024</a>] - \
<td><td>nat/external_media settings ignored in 14.4.1<br>(Reported by Christopher van \
de Sande)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27046'>ASTERISK-27046</a>] - \
<td><td>res_pjsip_transport_websocket: segfault in get_write_timeout<br>(Reported by \
Jørgen H)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27022'>ASTERISK-27022</a>] - \
<td><td>res_rtp_asterisk: Incorrect SSRC change for RTCP component<br>(Reported by \
Michael Walton)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26923'>ASTERISK-26923</a>] - \
<td><td>bridging: T.38 request is lost when channels are added to bridge<br>(Reported \
by Torrey Searle)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27053'>ASTERISK-27053</a>] - \
<td><td>res_pjsip_refer/session: Calls dropped during transfer<br>(Reported by Kevin \
Harwell)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27052'>ASTERISK-27052</a>] - \
<td><td>Asterisk build process fails with flag --with-pjproject-bundled with curl \
download command and slow network<br>(Reported by alex)</li></td></tr> <tr><td \
valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27039'>ASTERISK-27039</a>] - \
<td><td>chan_pjsip: Device state is idle when channel from endpoint is in early \
media<br>(Reported by Joshua Colp)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26996'>ASTERISK-26996</a>] - \
<td><td>chan_pjsip: Flipping between codecs<br>(Reported by Michael \
Maier)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26281'>ASTERISK-26281</a>] - \
<td><td>chan_pjsip would send INVITE to 'Unreachable' endpoints<br>(Reported by Jacek \
Konieczny)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26973'>ASTERISK-26973</a>] - \
<td><td>bridge: Crash when freeing frame and snooping<br>(Reported by Michel R. \
Vaillancourt)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-19291'>ASTERISK-19291</a>] - \
<td><td>Background in realtime<br>(Reported by Andrew Nowrot)</li></td></tr> <tr><td \
valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27025'>ASTERISK-27025</a>] - \
<td><td>channel / meetme: Fix missing parentheses<br>(Reported by Joshua \
Colp)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27021'>ASTERISK-27021</a>] - \
<td><td>GET /recordings/stored returns 500 Internal Server Error<br>(Reported by Tim \
Morgan)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-24858'>ASTERISK-24858</a>] - \
<td><td>[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel \
platform when using slin codec<br>(Reported by Frankie Chin)</li></td></tr> <tr><td \
valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-23951'>ASTERISK-23951</a>] - \
<td><td> Asterisk attempts and fails to build format_mp3 even if mp3lib was not \
downloaded<br>(Reported by Tzafrir Cohen)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-25294'>ASTERISK-25294</a>] - \
<td><td>srtp's crypto_get_random deprecated<br>(Reported by Tzafrir \
Cohen)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-23839'>ASTERISK-23839</a>] - \
<td><td>AGI - RECORD FILE - documentation doesn't describe BEEP argument<br>(Reported \
by Rusty Newton)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-22432'>ASTERISK-22432</a>] - \
<td><td>Async AGI crashes Asterisk when issuing "set variable" command without \
args<br>(Reported by Antoine Pitrou)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-25662'>ASTERISK-25662</a>] - \
<td><td>Malformed AGI 520 Usage response<br>(Reported by Tony \
Mountifield)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-25101'>ASTERISK-25101</a>] - \
<td><td>DTLS configuration can not be specified in the general section - \
documentation<br>(Reported by Ben Langfeld)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27008'>ASTERISK-27008</a>] - \
<td><td>res_format_attr_h264: SDP parse fails if fmtp optional parameters have a \
space<br>(Reported by John Harris)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26399'>ASTERISK-26399</a>] - \
<td><td>app_queue: Agent not called when caller is parked<br>(Reported by \
wushumasters)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26400'>ASTERISK-26400</a>] - \
<td><td>app_queue: Queue member stops being called after AMI "Redirect" action for \
queues with wrapuptime<br>(Reported by Etienne Lessard)</li></td></tr> <tr><td \
valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26715'>ASTERISK-26715</a>] - \
<td><td>app_queue: Member will not receive any new calls after doing a transfer if \
wrapuptime = greater than 0 and using Local channel<br>(Reported by David \
Brillert)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26975'>ASTERISK-26975</a>] - \
<td><td>app_queue: Non-zero wrapup time can cause agents not to receive queue calls \
after transfer queue call<br>(Reported by Lorne Gaetz)</li></td></tr> <tr><td \
valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27012'>ASTERISK-27012</a>] - \
<td><td>app_confbridge: ConfBridge sometimes does not play user name recording while \
leaving<br>(Reported by Robert Mordec)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26979'>ASTERISK-26979</a>] - \
<td><td>res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or \
110<br>(Reported by Javier Riveros )</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26982'>ASTERISK-26982</a>] - \
<td><td>chan_sip: rtcp_mux setting may cause ice completion failure/delay if client \
offers rtcp-mux as negotiable<br>(Reported by Stefan Engström)</li></td></tr> \
<tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26964'>ASTERISK-26964</a>] - \
<td><td>res_pjsip_session: Wrong From on reinvite when request and To URI \
differ<br>(Reported by Yasin CANER)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26789'>ASTERISK-26789</a>] - \
<td><td>Audit manipulation of channel flags without locks<br>(Reported by Joshua \
Colp)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26333'>ASTERISK-26333</a>] - \
<td><td>Problems with Blind Transfer, PJSIP (Aastra 6869i)<br>(Reported by Matthias \
Binder)</li></td></tr> </table>
<p>
<b>Information Requests made in this release:</b><br>
-----------------------------------<br>
<table border=0 padding=3>
<tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26976'>ASTERISK-26976</a>] - \
<td><td>libsrtp-2.x.x support<br>(Reported by Alex)</li></td></tr> </table>
<p>
<b>Improvements made in this release:</b><br>
-----------------------------------<br>
<table border=0 padding=3>
<tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26230'>ASTERISK-26230</a>] - \
<td><td>[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on \
startup<br>(Reported by Alexei Gradinari)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27043'>ASTERISK-27043</a>] - \
<td><td>Core/BuildSystem: Add defines to fix build with LibreSSL<br>(Reported by \
Guido Falsi)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27042'>ASTERISK-27042</a>] - \
<td><td>Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h \
file<br>(Reported by Guido Falsi)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26419'>ASTERISK-26419</a>] - \
<td><td>audiohooks: Remove redundant codec translations when using \
audiohooks<br>(Reported by Michael Walton)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26124'>ASTERISK-26124</a>] - \
<td><td>res_agi: Set audio format for EAGI audio stream<br>(Reported by John \
Fawcett)</li></td></tr> </table>
<p>
For a full list of changes in this release candidate, please see the ChangeLog:<br>
<a href='http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0-rc1'>http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.17.0-rc1</a>
<p>
<b>Thank you for your continued support of Asterisk!</b><br>
</body></html>
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