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List:       asterisk-dev
Subject:    [asterisk-dev] Asterisk 13.19.0-rc1 Now Available
From:       "Asterisk Development Team" <asteriskteam () digium ! com>
Date:       2017-12-20 23:50:55
Message-ID: E1eRo8V-0008V8-O9 () mail ! digium ! com
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The Asterisk Development Team would like to announce the first
release candidate of Asterisk 13.19.0.
This release candidate is available for immediate download at 
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.19.0-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:

New Features made in this release:
-----------------------------------
 * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
      incoming INVITE Request-URI.
      (Reported by Richard Mudgett)
 * ASTERISK-27413 - Add cache_media_frames debugging option.
   
      (Reported by Richard Mudgett)
 * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
      endpoint identification to IP only
      (Reported by Ben
      Merrills)

Bugs fixed in this release:
-----------------------------------
 * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
      read()
      (Reported by Abhay Gupta)
 * ASTERISK-25079 - AMI bridge of channels results in MOH not
      destroyed and robotic audio on one channel
      (Reported by
      Zane Conkle)
 * ASTERISK-27490 - chan_console: 'set active' fails to work
   
      (Reported by Tzafrir Cohen)
 * ASTERISK-24756 - ConfBridge sound_muted does not work from
      CLI or AMI
      (Reported by Thomas Frederiksen)
 * ASTERISK-25649 - Transfer application does not work with
      Local channels - documentation misleading
      (Reported by
      Ivan Ullmann)
 * ASTERISK-25869 - chan_sip: "rejected because extension not
      found" should be logged as a security event
      (Reported by
      Brian J. Murrell)
 * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
      streams
      (Reported by Wim De Vlaminck)
 * ASTERISK-24329 - Music On Hold announcement cuts intro of
      music the first time it is played
      (Reported by Thomas
      Frederiksen)
 * ASTERISK-19657 - Coverity Report: Fix issues for error type
      CHAR_IO
      (Reported by Matt Jordan)
 * ASTERISK-27175 - iax.conf demo peer is invalid
     
      (Reported by Tzafrir Cohen)
 * ASTERISK-27430 - README refers to security documents that do
      not exist.
      (Reported by Corey Farrell)
 * ASTERISK-20281 - "core set verbose" behaves strangely, can't
      alias it, cli.conf example broken
      (Reported by Tim
      Ringenbach at Asteria Solutions Group)
 * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
      FXS gateway
      (Reported by Tzafrir Cohen)
 * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
      RTCP packet will write past where it should
      (Reported by
      Vitezslav Novy)
 * ASTERISK-27408 - Identify causes and fix
      pjsip/resolver/srv/failover/in_dialog/transport_tcp
     
      (Reported by Corey Farrell)
 * ASTERISK-18411 - Queue members with hints for state_interface
      get stuck in "In Use" state.
      (Reported by Steven T.
      Wheeler)
 * ASTERISK-26131 - chan_sip: Crash Asterisk (in
      sip_request_call at chan_sip.c) by making a call to a single
      character in a dot pattern match
      (Reported by Dwayne
      Hubbard)
 * ASTERISK-27475 - codec_opus requires libcurl
      (Reported
      by Samuel For)
 * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
      not applied on reload
      (Reported by John Bigelow)
 * ASTERISK-27465 - CLI Completion Not Working
      (Reported
      by Ross Beer)
 * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
      Variable CDR(amaflags)=...
      (Reported by Richard Mudgett)
 * ASTERISK-27453 - RTP: Blind transfer direct media scenario
      results in one way audio.
      (Reported by Richard Mudgett)
 * ASTERISK-20643 - SIP ICE support - remove hardcoded
      limitation on SDP size, make ICE support disabled by default in
      SIP, maybe provide a better warning message
      (Reported by
      Roy)
 * ASTERISK-26980 - pjsip: Clean up WebRTC disables
     
      (Reported by abelbeck)
 * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
      flooded with unauthenticated requests
      (Reported by George
      Joseph)
 * ASTERISK-27454 - res_http_post: Don't require
      GMIME_MAJOR_VERSION
      (Reported by Joshua Colp)
 * ASTERISK-23735 - Transcoding makes bad choice in high-rate
      translations
      (Reported by Richard Kenner)
 * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
      message.
      (Reported by Frank Durden)
 * ASTERISK-24662 - [patch] column and row headers for Signed
      Linear format variants in output of 'core show translation' are
      ambiguous
      (Reported by Rusty Newton)
 * ASTERISK-27353 - H323 audio starts with a delay of 2
      seconds.
      (Reported by Marco Giordani)
 * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
      media
      (Reported by Kevin Harwell)
 * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
      (srflx) with Dual-Stack.
      (Reported by Alexander Traud)
 * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
      IPv6 addresses.
      (Reported by Alexander Traud)
 * ASTERISK-27435 - [patch] configure:
      pjsip_evsub_set_uas_timeout not found.
      (Reported by
      Alexander Traud)
 * ASTERISK-27431 - Asterisk fails to build when openssl headers
      are not installed.
      (Reported by Corey Farrell)
 * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra

      (Reported by Ivan Larionov)
 * ASTERISK-27421 - RTP source learning not working with devices
      that have some clock issues
      (Reported by nappsoft)
 * ASTERISK-27361 - Attended transfer crashes in Asterisk
      13.17.2
      (Reported by Alessandro Pimenta)
 * ASTERISK-27238 - Bridging: Crash freeing a frame that's
      already been freed
      (Reported by Richard Kenner)
 * ASTERISK-27412 - core: Audiohook freeing interpolated frame
      when it shouldn't.
      (Reported by Mikhail)
 * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
      channel variable before closing the file
      (Reported by
      George Joseph)
 * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
      insert same ip address in "source ip address" and "destination
      ip address" fields in HEP packets
      (Reported by Max Norba)
 * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
      is equal to RemoteAddress)
      (Reported by Vasilii Rogin)
 * ASTERISK-27415 - asterisk.conf: Setting astctl without
      setting astrundir is ineffective.
      (Reported by Corey
      Farrell)
 * ASTERISK-27411 - pjsip: TCP connections may not be destroyed

      (Reported by Joshua Colp)
 * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
      responses.
      (Reported by Corey Farrell)
 * ASTERISK-27337 - chan_sip: Security vulnerability with client
      code header (revisited)
      (Reported by Richard Mudgett)
 * ASTERISK-27319 - (Security) Function in PJSIP 2.7
      miscalculates the length of an unsigned long variable in 64bit
      machines
      (Reported by Kim youngsung)
 * ASTERISK-27391 - Regression: Deadlock between AOR named lock
      and pjproject grp lock
      (Reported by shaurya jain)
 * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
      contact read from astdb or database
      (Reported by Aaron An)
 * ASTERISK-27290 - res_pjsip: PIDF contact field has
      malformed/invalid XML
      (Reported by basildane)
 * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
      values
      (Reported by seanchann.zhou)
 * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
      enabled but not configured.
      (Reported by Alexander Traud)
 * ASTERISK-26426 - format_ogg_opus: remove from source
     
      (Reported by Kevin Harwell)
 * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
    
      (Reported by Alexander Traud)
 * ASTERISK-27378 - Modules: Fix issues with CLI completion.
   
      (Reported by Corey Farrell)
 * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
      character isn't allowed any more
      (Reported by Michael
      Maier)
 * ASTERISK-27390 - Audit menuselect module dependencies
     
      (Reported by Corey Farrell)
 * ASTERISK-27389 - Optional API modules should not allow
      unload.
      (Reported by Corey Farrell)
 * ASTERISK-27369 - Bridge() dialplan application fails without
      setting BRIDGERESULT channel variable
      (Reported by James
      Terhune)
 * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
      documentation
      (Reported by Igor Goncharovsky)
 * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
      'imap_delete_old_greeting'
      (Reported by Anthony Messina)
 * ASTERISK-27194 - jitterbuffer: Does not handle case where
      translator returns null frame.
      (Reported by Joshua Elson)
 * ASTERISK-26639 - core: Disabling xmldoc support does not
      work. Also results in abort during Asterisk startup.
     
      (Reported by Mr Dini)
 * ASTERISK-27372 - ARI: Node ARI client broken in latest
      versions of 13 and 14
      (Reported by Benjamin Keith Ford)
 * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
      absence of the Expires header field with an unsubscribe action.

      (Reported by Jonathan Cloots)
 * ASTERISK-25960 - The config_hook unit test causes Asterisk to
      crash if run a second time
      (Reported by George Joseph)
 * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
      when rtp_ipv6 set to yes
      (Reported by Martin Cisárik)
 * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
      curl is loaded
      (Reported by Ronald Raikes)
 * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
      but first on SDP media level.
      (Reported by Alexander
      Traud)
 * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
      Assertion on un/re-load: mod.id == -1
      (Reported by Tzafrir
      Cohen)
 * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
      enabling with conf file option - also 'sip set debug off'
      reports debugging disabled, when it really isn't
      (Reported
      by Rusty Newton)
 * ASTERISK-27328 - Missing openssl dependencies in
      res_rtp_asterisk and tcptls
      (Reported by Tzafrir Cohen)
 * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
      (o=) contains local address.
      (Reported by Alexander Traud)
 * ASTERISK-27343 - Fails to build in FreeBSD due to
      sys/sysmacros.h not existing there
      (Reported by Guido
      Falsi)
 * ASTERISK-27340 - backtrace.c: Crash due to double-free.
     
      (Reported by Corey Farrell)
 * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
      stopping.
      (Reported by Alexander Traud)
 * ASTERISK-27333 - sip_to_pjsip not correctly handling
      disallow=all directive
      (Reported by Torrey Searle)
 * ASTERISK-27416 - Can't load res_corosync.so module on
      Asterisk 13.18.2
      (Reported by Anton Mosin)

Improvements made in this release:
-----------------------------------
 * ASTERISK-24297 - cdr.c: Minor code optimizations.
     
      (Reported by Richard Mudgett)
 * ASTERISK-27449 - [PATCH] When failing to acquire target
      during attended transfer, display wanted extension
     
      (Reported by Niklas Larsson)
 * ASTERISK-27456 - app_voicemail: Add new object for
      VoicemailUserEntry
      (Reported by sungtae kim)
 * ASTERISK-27380 - ast_coredumper: allow pointing out the
      asterisk binary explicitly
      (Reported by Tzafrir Cohen)
 * ASTERISK-23556 - Compilation warning for invert.c (array
      subscript is above array bounds)
      (Reported by Marcello
      Ceschia)
 * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
     
      (Reported by Richard Mudgett)
 * ASTERISK-27335 - CDR performance needs improvement.
     
      (Reported by Richard Mudgett)

For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0-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 13.19.0.<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 13.19.0-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>New Features 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-27478'>ASTERISK-27478</a>] - \
<td><td>PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming INVITE \
Request-URI.<br>(Reported by Richard Mudgett)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27413'>ASTERISK-27413</a>] - \
<td><td>Add cache_media_frames debugging option.<br>(Reported by Richard \
Mudgett)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27206'>ASTERISK-27206</a>] - \
<td><td>res_pjsip: No mechanism exists to limit endpoint identification to IP \
only<br>(Reported by Ben Merrills)</li></td></tr> </table>
<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-27299'>ASTERISK-27299</a>] - \
<td><td>Asterisk Hangs with Bad file descriptor on read()<br>(Reported by Abhay \
Gupta)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-25079'>ASTERISK-25079</a>] - \
<td><td>AMI bridge of channels results in MOH not destroyed and robotic audio on one \
channel<br>(Reported by Zane Conkle)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27490'>ASTERISK-27490</a>] - \
<td><td>chan_console: 'set active' fails to work<br>(Reported by Tzafrir \
Cohen)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-24756'>ASTERISK-24756</a>] - \
<td><td>ConfBridge sound_muted does not work from CLI or AMI<br>(Reported by Thomas \
Frederiksen)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-25649'>ASTERISK-25649</a>] - \
<td><td>Transfer application does not work with Local channels - documentation \
misleading<br>(Reported by Ivan Ullmann)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-25869'>ASTERISK-25869</a>] - \
<td><td>chan_sip: "rejected because extension not found" should be logged as a \
security event<br>(Reported by Brian J. Murrell)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27440'>ASTERISK-27440</a>] - \
<td><td>Strictrtp has issues to qualify video rtp streams<br>(Reported by Wim De \
Vlaminck)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-24329'>ASTERISK-24329</a>] - \
<td><td>Music On Hold announcement cuts intro of music the first time it is \
played<br>(Reported by Thomas Frederiksen)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-19657'>ASTERISK-19657</a>] - \
<td><td>Coverity Report: Fix issues for error type CHAR_IO<br>(Reported by Matt \
Jordan)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27175'>ASTERISK-27175</a>] - \
<td><td>iax.conf demo peer is invalid<br>(Reported by Tzafrir Cohen)</li></td></tr> \
<tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27430'>ASTERISK-27430</a>] - \
<td><td>README refers to security documents that do not exist.<br>(Reported by Corey \
Farrell)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-20281'>ASTERISK-20281</a>] - \
<td><td>"core set verbose" behaves strangely, can't alias it, cli.conf example \
broken<br>(Reported by Tim Ringenbach at Asteria Solutions Group)</li></td></tr> \
<tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27382'>ASTERISK-27382</a>] - \
<td><td>crash after an invalid rtcp packet from GT48 FXS gateway<br>(Reported by \
Tzafrir Cohen)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27429'>ASTERISK-27429</a>] - \
<td><td>res_rtp_asterisk: Multiple reports in an RTCP packet will write past where it \
should<br>(Reported by Vitezslav Novy)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27408'>ASTERISK-27408</a>] - \
<td><td>Identify causes and fix \
pjsip/resolver/srv/failover/in_dialog/transport_tcp<br>(Reported by Corey \
Farrell)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-18411'>ASTERISK-18411</a>] - \
<td><td>Queue members with hints for state_interface get stuck in "In Use" \
state.<br>(Reported by Steven T. Wheeler)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26131'>ASTERISK-26131</a>] - \
<td><td>chan_sip: Crash Asterisk (in sip_request_call at chan_sip.c) by making a call \
to a single character in a dot pattern match<br>(Reported by Dwayne \
Hubbard)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27475'>ASTERISK-27475</a>] - \
<td><td>codec_opus requires libcurl<br>(Reported by Samuel For)</li></td></tr> \
<tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27467'>ASTERISK-27467</a>] - \
<td><td>pjsip_options: qualify_frequency sometimes not applied on reload<br>(Reported \
by John Bigelow)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27465'>ASTERISK-27465</a>] - \
<td><td>CLI Completion Not Working<br>(Reported by Ross Beer)</li></td></tr> <tr><td \
valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27460'>ASTERISK-27460</a>] - \
<td><td>CDR: Deadlock using AMI Originate with Variable \
CDR(amaflags)=...<br>(Reported by Richard Mudgett)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27453'>ASTERISK-27453</a>] - \
<td><td>RTP: Blind transfer direct media scenario results in one way \
audio.<br>(Reported by Richard Mudgett)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-20643'>ASTERISK-20643</a>] - \
<td><td>SIP ICE support - remove hardcoded limitation on SDP size, make ICE support \
disabled by default in SIP, maybe provide a better warning message<br>(Reported by \
Roy)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26980'>ASTERISK-26980</a>] - \
<td><td>pjsip: Clean up WebRTC disables<br>(Reported by abelbeck)</li></td></tr> \
<tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27452'>ASTERISK-27452</a>] - \
<td><td>Security: chan_skinny:  Memory exhaustion if flooded with unauthenticated \
requests<br>(Reported by George Joseph)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27454'>ASTERISK-27454</a>] - \
<td><td>res_http_post: Don't require GMIME_MAJOR_VERSION<br>(Reported by Joshua \
Colp)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-23735'>ASTERISK-23735</a>] - \
<td><td>Transcoding makes bad choice in high-rate translations<br>(Reported by \
Richard Kenner)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27445'>ASTERISK-27445</a>] - \
<td><td>ARI: Updating a bridge gives wrong error message.<br>(Reported by Frank \
Durden)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-24662'>ASTERISK-24662</a>] - \
<td><td>[patch] column and row headers for Signed Linear format variants in output of \
'core show translation' are ambiguous<br>(Reported by Rusty Newton)</li></td></tr> \
<tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27353'>ASTERISK-27353</a>] - \
<td><td>H323 audio starts with a delay of 2 seconds.<br>(Reported by Marco \
Giordani)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27442'>ASTERISK-27442</a>] - \
<td><td>pjsip: 183 without To tag does not negotiate media<br>(Reported by Kevin \
Harwell)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27437'>ASTERISK-27437</a>] - \
<td><td>[patch] ICE: server-reflexive candidates (srflx) with \
Dual-Stack.<br>(Reported by Alexander Traud)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27434'>ASTERISK-27434</a>] - \
<td><td>[patch] chan_sip/ICE: Square brackets around IPv6 addresses.<br>(Reported by \
Alexander Traud)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27435'>ASTERISK-27435</a>] - \
<td><td>[patch] configure: pjsip_evsub_set_uas_timeout not found.<br>(Reported by \
Alexander Traud)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27431'>ASTERISK-27431</a>] - \
<td><td>Asterisk fails to build when openssl headers are not installed.<br>(Reported \
by Corey Farrell)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27332'>ASTERISK-27332</a>] - \
<td><td>Asterisk fails to configure on MacOS Sierra<br>(Reported by Ivan \
Larionov)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27421'>ASTERISK-27421</a>] - \
<td><td>RTP source learning not working with devices that have some clock \
issues<br>(Reported by nappsoft)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27361'>ASTERISK-27361</a>] - \
<td><td>Attended transfer crashes in Asterisk 13.17.2<br>(Reported by Alessandro \
Pimenta)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27238'>ASTERISK-27238</a>] - \
<td><td>Bridging: Crash freeing a frame that's already been freed<br>(Reported by \
Richard Kenner)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27412'>ASTERISK-27412</a>] - \
<td><td>core: Audiohook freeing interpolated frame when it shouldn't.<br>(Reported by \
Mikhail)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27423'>ASTERISK-27423</a>] - \
<td><td>app_record:  We set the RECORD_STATUS channel variable before closing the \
file<br>(Reported by George Joseph)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26758'>ASTERISK-26758</a>] - \
<td><td>res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source \
ip address" and "destination ip address" fields in HEP packets<br>(Reported by Max \
Norba)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27363'>ASTERISK-27363</a>] - \
<td><td>res_http_websocket: Wrong LocalAddress (it is equal to \
RemoteAddress)<br>(Reported by Vasilii Rogin)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27415'>ASTERISK-27415</a>] - \
<td><td>asterisk.conf: Setting astctl without setting astrundir is \
ineffective.<br>(Reported by Corey Farrell)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27411'>ASTERISK-27411</a>] - \
<td><td>pjsip: TCP connections may not be destroyed<br>(Reported by Joshua \
Colp)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27345'>ASTERISK-27345</a>] - \
<td><td>res_pjsip_session: RTP instances leak on 488 responses.<br>(Reported by Corey \
Farrell)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27337'>ASTERISK-27337</a>] - \
<td><td>chan_sip: Security vulnerability with client code header \
(revisited)<br>(Reported by Richard Mudgett)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27319'>ASTERISK-27319</a>] - \
<td><td>(Security) Function in PJSIP 2.7 miscalculates the length of an unsigned long \
variable in 64bit machines<br>(Reported by Kim youngsung)</li></td></tr> <tr><td \
valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27391'>ASTERISK-27391</a>] - \
<td><td>Regression: Deadlock between AOR named lock and pjproject grp \
lock<br>(Reported by shaurya jain)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27393'>ASTERISK-27393</a>] - \
<td><td>res_pjsip: Crash occurs when an empty contact read from astdb or \
database<br>(Reported by Aaron An)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27290'>ASTERISK-27290</a>] - \
<td><td>res_pjsip: PIDF contact field has malformed/invalid XML<br>(Reported by \
basildane)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27032'>ASTERISK-27032</a>] - \
<td><td>res_pjsip: TLS options do not handle empty values<br>(Reported by \
seanchann.zhou)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27394'>ASTERISK-27394</a>] - \
<td><td>[patch] tcptls: Print notice when TLS is enabled but not \
configured.<br>(Reported by Alexander Traud)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26426'>ASTERISK-26426</a>] - \
<td><td>format_ogg_opus: remove from source<br>(Reported by Kevin \
Harwell)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27356'>ASTERISK-27356</a>] - \
<td><td>[patch] libsrtp-2.x.x + AES-GCM support<br>(Reported by Alexander \
Traud)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27378'>ASTERISK-27378</a>] - \
<td><td>Modules: Fix issues with CLI completion.<br>(Reported by Corey \
Farrell)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27387'>ASTERISK-27387</a>] - \
<td><td>Regression: pjsip 13.18.0 - from_user - "+" character isn't allowed any \
more<br>(Reported by Michael Maier)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27390'>ASTERISK-27390</a>] - \
<td><td>Audit menuselect module dependencies<br>(Reported by Corey \
Farrell)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27389'>ASTERISK-27389</a>] - \
<tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27369'>ASTERISK-27369</a>] - \
<td><td>Bridge() dialplan application fails without setting BRIDGERESULT channel \
variable<br>(Reported by James Terhune)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27377'>ASTERISK-27377</a>] - \
<td><td>Typo in CHANNEL(dtmf_features) usage documentation<br>(Reported by Igor \
Goncharovsky)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27181'>ASTERISK-27181</a>] - \
<td><td>GCC 7 warning: app_voicemail.c: In function \
'imap_delete_old_greeting'<br>(Reported by Anthony Messina)</li></td></tr> <tr><td \
valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27194'>ASTERISK-27194</a>] - \
<td><td>jitterbuffer: Does not handle case where translator returns null \
frame.<br>(Reported by Joshua Elson)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-26639'>ASTERISK-26639</a>] - \
<td><td>core: Disabling xmldoc support does not work. Also results in abort during \
Asterisk startup.<br>(Reported by Mr Dini)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27372'>ASTERISK-27372</a>] - \
<td><td>ARI: Node ARI client broken in latest versions of 13 and 14<br>(Reported by \
Benjamin Keith Ford)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-18140'>ASTERISK-18140</a>] - \
<td><td>Expires handling in SUBSCRIBE confuses the absence of the Expires header \
field with an unsubscribe action.<br>(Reported by Jonathan Cloots)</li></td></tr> \
<tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-25960'>ASTERISK-25960</a>] - \
<td><td>The config_hook unit test causes Asterisk to crash if run a second \
time<br>(Reported by George Joseph)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27198'>ASTERISK-27198</a>] - \
<td><td>res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6 set to \
yes<br>(Reported by Martin Cisárik)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27346'>ASTERISK-27346</a>] - \
<td><td>res_xmpp: Crash if OAuth 2.0 is used before curl is loaded<br>(Reported by \
Ronald Raikes)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27365'>ASTERISK-27365</a>] - \
<td><td>[patch] chan_sip: Crypto attribute not last but first on SDP media \
level.<br>(Reported by Alexander Traud)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-24483'>ASTERISK-24483</a>] - \
<td><td>res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on un/re-load: mod.id == \
-1<br>(Reported by Tzafrir Cohen)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-23462'>ASTERISK-23462</a>] - \
<td><td>Cannot disable SIP debugging via CLI after enabling with conf file option - \
also 'sip set debug off' reports debugging disabled, when it really \
isn't<br>(Reported by Rusty Newton)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27328'>ASTERISK-27328</a>] - \
<td><td>Missing openssl dependencies in res_rtp_asterisk and tcptls<br>(Reported by \
Tzafrir Cohen)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27341'>ASTERISK-27341</a>] - \
<td><td>[patch] res_pjsip_session: SIP/SDP origin (o=) contains local \
address.<br>(Reported by Alexander Traud)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27343'>ASTERISK-27343</a>] - \
<td><td>Fails to build in FreeBSD due to sys/sysmacros.h not existing \
there<br>(Reported by Guido Falsi)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27340'>ASTERISK-27340</a>] - \
<td><td>backtrace.c: Crash due to double-free.<br>(Reported by Corey \
Farrell)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27339'>ASTERISK-27339</a>] - \
<td><td>[patch] Crash on ast_ssl_teardown when stopping.<br>(Reported by Alexander \
Traud)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27333'>ASTERISK-27333</a>] - \
<td><td>sip_to_pjsip not correctly handling disallow=all directive<br>(Reported by \
Torrey Searle)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27416'>ASTERISK-27416</a>] - \
<td><td>Can't load res_corosync.so module on Asterisk 13.18.2<br>(Reported by Anton \
Mosin)</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-24297'>ASTERISK-24297</a>] - \
<td><td>cdr.c: Minor code optimizations.<br>(Reported by Richard \
Mudgett)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27449'>ASTERISK-27449</a>] - \
<td><td>[PATCH] When failing to acquire target during attended transfer, display \
wanted extension<br>(Reported by Niklas Larsson)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27456'>ASTERISK-27456</a>] - \
<td><td>app_voicemail: Add new object for VoicemailUserEntry<br>(Reported by sungtae \
kim)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27380'>ASTERISK-27380</a>] - \
<td><td>ast_coredumper: allow pointing out the asterisk binary \
explicitly<br>(Reported by Tzafrir Cohen)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-23556'>ASTERISK-23556</a>] - \
<td><td>Compilation warning for invert.c (array subscript is above array \
bounds)<br>(Reported by Marcello Ceschia)</li></td></tr> <tr><td valign=top \
nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27355'>ASTERISK-27355</a>] - \
<td><td>Upgrade bundled PJPROJECT to 2.7<br>(Reported by Richard \
Mudgett)</li></td></tr> <tr><td valign=top nowrap='nowrap'><li>[<a \
href='https://issues.asterisk.org/jira/browse/ASTERISK-27335'>ASTERISK-27335</a>] - \
<td><td>CDR performance needs improvement.<br>(Reported by Richard \
Mudgett)</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-13.19.0-rc1'>http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.19.0-rc1</a>
 <p>
<b>Thank you for your continued support of Asterisk!</b><br>
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