<div dir="ltr">Hi list can i ask U <div>does this <span style="font-size:12.8000001907349px">release solved my problem</span></div><div><span style="font-size:12.8000001907349px"><a href="https://github.com/versatica/JsSIP/issues/311">https://github.com/versatica/JsSIP/issues/311</a></span><br></div><div><span style="font-size:12.8000001907349px">(already try on a last FREEPBX --same issue)</span></div><div><span style="font-size:12.8000001907349px">Regards</span></div></div><div class="gmail_extra"><br><div class="gmail_quote">2015-04-01 22:01 GMT+03:00 Asterisk Development Team <span dir="ltr"><<a href="mailto:asteriskteam@digium.com" target="_blank">asteriskteam@digium.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">The Asterisk Development Team has announced the release of Asterisk 11.17.0.<br>
This release is available for immediate download at<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk</a><br>
<br>
The release of Asterisk 11.17.0 resolves several issues reported by the<br>
community and would have not been possible without your participation.<br>
Thank you!<br>
<br>
The following are the issues resolved in this release:<br>
<br>
New Features made in this release:<br>
-----------------------------------<br>
* ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation<br>
(Reported by Dwayne Hubbard)<br>
<br>
Bugs fixed in this release:<br>
-----------------------------------<br>
* ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in<br>
res_odbc (Reported by ibercom)<br>
* ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE<br>
with replaces (Reported by Eelco Brolman)<br>
* ASTERISK-24479 - Enable REF_DEBUG for module references<br>
(Reported by Corey Farrell)<br>
* ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to<br>
fully disconnect underlying socket, leading to events being<br>
dropped with no additional information (Reported by Matt Jordan)<br>
* ASTERISK-24772 - ODBC error in realtime sippeers when device<br>
unregisters under MariaDB (Reported by Richard Miller)<br>
* ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove<br>
(Reported by Corey Farrell)<br>
* ASTERISK-24799 - [patch] make fails with undefined reference to<br>
SSLv3_client_method (Reported by Alexander Traud)<br>
* ASTERISK-24787 - [patch] - Microsoft exchange incompatibility<br>
for playing back messages stored in IMAP - play_message: No<br>
origtime (Reported by Graham Barnett)<br>
* ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc<br>
OSX with 64 bit integers (Reported by Corey Farrell)<br>
* ASTERISK-24796 - Codecs and bucket schema's prevent module<br>
unload (Reported by Corey Farrell)<br>
* ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML<br>
(Reported by Ashley Sanders)<br>
* ASTERISK-24797 - bridge_softmix: G.729 codec license held<br>
(Reported by Kevin Harwell)<br>
* ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid<br>
thread ID being passed to pthread_kill (Reported by JoshE)<br>
* ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime<br>
fail (Reported by Terry Wilson)<br>
* ASTERISK-23214 - chan_sip WARNING message 'We are requesting<br>
SRTP for audio, but they responded without it' is ambiguous and<br>
wrong in some cases (Reported by Rusty Newton)<br>
* ASTERISK-15434 - [patch] When ast_pbx_start failed, both an<br>
error response and BYE are sent to the caller (Reported by<br>
Makoto Dei)<br>
* ASTERISK-18105 - most of asterisk modules are unbuildable in<br>
cygwin environment (Reported by feyfre)<br>
* ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)<br>
* ASTERISK-24838 - chan_sip: Locking inversion occurs when<br>
building a peer causes a peer poke during request handling<br>
(Reported by Richard Mudgett)<br>
* ASTERISK-24825 - Caller ID not recognized using<br>
Centrex/Distinctive dialing (Reported by Richard Mudgett)<br>
* ASTERISK-24739 - [patch] - Out of files -- call fails --<br>
numerous files with inodes from under /usr/share/zoneinfo,<br>
mostly posixrules (Reported by Ed Hynan)<br>
* ASTERISK-23390 - NewExten Event with application AGI shows up<br>
before and after AGI runs (Reported by Benjamin Keith Ford)<br>
* ASTERISK-24786 - [patch] - Asterisk terminates when playing a<br>
voicemail stored in LDAP (Reported by Graham Barnett)<br>
* ASTERISK-24808 - res_config_odbc: Improper escaping of<br>
backslashes occurs with MySQL (Reported by Javier Acosta)<br>
* ASTERISK-20850 - [patch]Nested functions aren't portable.<br>
Adapting RAII_VAR to use clang/llvm blocks to get the<br>
same/similar functionality. (Reported by Diederik de Groot)<br>
* ASTERISK-19470 - Documentation on app_amd is incorrect (Reported<br>
by Frank DiGennaro)<br>
* ASTERISK-21038 - Bad command completion of "core set debug<br>
channel" (Reported by Richard Kenner)<br>
* ASTERISK-18708 - func_curl hangs channel under load (Reported by<br>
Dave Cabot)<br>
* ASTERISK-16779 - Cannot disallow unknown format '' (Reported by<br>
Atis Lezdins)<br>
* ASTERISK-24876 - Investigate reference leaks from<br>
tests/channels/local/local_optimize_away (Reported by Corey<br>
Farrell)<br>
* ASTERISK-24817 - init_logger_chain: unreachable code block<br>
(Reported by Corey Farrell)<br>
* ASTERISK-24880 - [patch]Compilation under OpenBSD (Reported by<br>
snuffy)<br>
* ASTERISK-24879 - [patch]Compilation fails due to 64bit time<br>
under OpenBSD (Reported by snuffy)<br>
<br>
Improvements made in this release:<br>
-----------------------------------<br>
* ASTERISK-24790 - Reduce spurious noise in logs from voicemail -<br>
Couldn't find mailbox %s in context (Reported by Graham Barnett)<br>
<br>
For a full list of changes in this release, please see the ChangeLog:<br>
<br>
<a href="http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0" target="_blank">http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0</a><br>
<br>
Thank you for your continued support of Asterisk!<br>
<span class="HOEnZb"><font color="#888888"><br>
--<br>
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</font></span></blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature"><div>Best regards<br>Antony<br></div><div>моб (066) 919-75-33</div><div>моб (063) 656-43-40</div><div><a href="mailto:mail%3Asatskiy.a@gmail.com" target="_blank">satskiy.a@gmail.com</a></div></div>
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