<div dir="ltr">That is correct for turning SRTP off on a Snom phone.<br><div class="gmail_extra"><br><div class="gmail_quote">On 12 November 2015 at 16:46, (lists) Denis BUCHER <span dir="ltr"><<a href="mailto:dbucherml@hsolutions.ch" target="_blank">dbucherml@hsolutions.ch</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<div>Dear Sam, dear jg, dear Mitul, dear
all,<br>
<br>
Thanks a lot for your advices! I had the same idea, but it still
doesn't work!<br>
<br>
Maybe I changed the wrong option on the GUI configuration ?<br>
I went to menu "Setup" > "Identity 1" > "RTP" > "RTP
Encryption:" > "off" on both phones.<br>
And in the configuration I see "user_srtp1!: off"<br>
<br>
Is this right ?<span class="HOEnZb"><font color="#888888"><br>
<br>
Denis</font></span><div><div class="h5"><br>
<br>
Le 12.11.2015 17:05, Sam Basan a écrit :<br>
</div></div></div><div><div class="h5">
<blockquote type="cite">
<div>
<p class="MsoNormal"><a name="150fc98aaaa0e080__MailEndCompose"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d">Snom
default configuration is SRTP enabled.<u></u><u></u></span></a></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d">You
should disable the SRTP from the phone web GUI configuration<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span><u></u> <u></u></span></b></p>
<p class="MsoNormal"><b><span><u></u> <u></u></span></b></p>
<p class="MsoNormal"><b><span>Sincerely,<u></u><u></u></span></b></p>
<p class="MsoNormal"><span style="font-size:14.0pt;font-family:"Arial",sans-serif;color:black"> </span><span style="color:black"><img src="cid:part2.03050604.03020003@hsolutions.ch" alt="cid:image001.jpg@01D0D5C4.27A0CBA0" height="56" width="87"><u></u><u></u></span></p>
<p class="MsoNormal"><b><span>Sam Basan</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d"><img src="cid:part3.08060604.05070309@hsolutions.ch" alt="cid:image003.png@01C918DA.6B3E4530" height="48" width="397"><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">From:</span></b><span style="font-size:11.0pt;font-family:"Calibri",sans-serif">
Mitul Limbani [<a href="mailto:mitul@enterux.in" target="_blank">mailto:mitul@enterux.in</a>] <br>
<b>Sent:</b> Thursday, November 12, 2015 5:25 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial
Discussion <a href="mailto:asterisk-users@lists.digium.com" target="_blank"><asterisk-users@lists.digium.com></a><br>
<b>Subject:</b> Re: [asterisk-users] No sound with internal
calls depending on which phones<u></u><u></u></span></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p>You might have to disable srtp negotiations inside the phone
web ui options. <u></u><u></u></p>
<p>Mitul <u></u><u></u></p>
<div>
<p class="MsoNormal">On Nov 12, 2015 8:53 PM, "(lists) Denis
BUCHER" <<a href="mailto:dbucherml@hsolutions.ch" target="_blank">dbucherml@hsolutions.ch</a>>
wrote:<u></u><u></u></p>
<blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm">
<div>
<p class="MsoNormal">Dear all,<br>
<br>
I have a very strange problem :<u></u><u></u></p>
<ul type="disc">
<li class="MsoNormal">external calls work perfectly,<u></u><u></u></li>
<li class="MsoNormal">internal calls between some phones too,<u></u><u></u></li>
<li class="MsoNormal">but internal call between two similar
phones don't work !!! (Snom 710)<u></u><u></u></li>
</ul>
<p class="MsoNormal">When we have sound, there are no
errors in asterisk. When we do not have sound, there is
the following error :<u></u><u></u></p>
<ul type="disc">
<li class="MsoNormal">[Nov 10 17:51:47] ERROR[21480]:
chan_sip.c:28306 setup_srtp: No SRTP module loaded,
can't setup SRTP session.<u></u><u></u></li>
</ul>
<p class="MsoNormal">This is a working internal call :<br>
<br>
<u></u><u></u></p>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal"> == Using SIP RTP CoS mark 5<br>
-- Executing [301@local:1]
Dial("SIP/dbucher-00000000", "SIP/phone1") in new
stack<br>
== Using SIP RTP CoS mark 5<br>
-- Called phone1<br>
-- SIP/phone1-00000001 is ringing<br>
-- SIP/phone1-00000001 is ringing<br>
-- SIP/phone1-00000001 is ringing<br>
-- SIP/phone1-00000001 is ringing<br>
-- SIP/phone1-00000001 is ringing<br>
-- SIP/phone1-00000001 answered
SIP/dbucher-00000000<br>
-- Remotely bridging SIP/dbucher-00000000 and
SIP/phone1-00000001<br>
Sent RTP P2P packet to <a href="http://192.168.128.99:49646" target="_blank">192.168.128.99:49646</a>
(type 00, len 000160)<br>
Sent RTP P2P packet to <a href="http://192.168.128.99:49646" target="_blank">192.168.128.99:49646</a>
(type 00, len 000160)<br>
Sent RTP P2P packet to <a href="http://192.168.128.99:49646" target="_blank">192.168.128.99:49646</a>
(type 00, len 000160)<br>
Got RTP packet from <a href="http://192.168.128.99:49646" target="_blank">192.168.128.99:49646</a>
(type 126, seq 031575, ts 000001, len 000000)<br>
[Nov 10 17:50:50] NOTICE[21513]:
res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP
codec 126 received from '<a href="http://192.168.128.99:49646" target="_blank">192.168.128.99:49646</a>'<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:57818" target="_blank">192.168.128.231:57818</a>
(type 00, len 000160)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:57818" target="_blank">192.168.128.231:57818</a>
(type 00, len 000160)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:57818" target="_blank">192.168.128.231:57818</a>
(type 00, len 000160)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:57818" target="_blank">192.168.128.231:57818</a>
(type 00, len 000160)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:57818" target="_blank">192.168.128.231:57818</a>
(type 00, len 000160)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:57818" target="_blank">192.168.128.231:57818</a>
(type 00, len 000160)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:57818" target="_blank">192.168.128.231:57818</a>
(type 00, len 000160)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:57818" target="_blank">192.168.128.231:57818</a>
(type 00, len 000160)<br>
== Spawn extension (local, 301, 1) exited non-zero
on 'SIP/dbucher-00000000'<u></u><u></u></p>
</blockquote>
<p class="MsoNormal">This is a non-working call :<br>
<br>
<u></u><u></u></p>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal"> == Using SIP RTP CoS mark 5<br>
[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306
setup_srtp: No SRTP module loaded, can't setup SRTP
session.<br>
-- Executing [301@local:1]
Dial("SIP/hsolutionspf5-00000002", "SIP/phone1") in
new stack<br>
== Using SIP RTP CoS mark 5<br>
-- Called phone1<br>
-- SIP/phone1-00000003 is ringing<br>
-- SIP/phone1-00000003 is ringing<br>
-- SIP/phone1-00000003 is ringing<br>
-- SIP/phone1-00000003 is ringing<br>
-- SIP/phone1-00000003 is ringing<br>
-- SIP/phone1-00000003 answered
SIP/hsolutionspf5-00000002<br>
-- Remotely bridging SIP/hsolutionspf5-00000002
and SIP/phone1-00000003<br>
Sent RTP P2P packet to <a href="http://192.168.128.228:65494" target="_blank">192.168.128.228:65494</a>
(type 00, len 000160)<br>
Sent RTP P2P packet to <a href="http://192.168.128.228:65494" target="_blank">192.168.128.228:65494</a>
(type 00, len 000160)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:51350" target="_blank">192.168.128.231:51350</a>
(type 03, len 000033)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:51350" target="_blank">192.168.128.231:51350</a>
(type 03, len 000033)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:51350" target="_blank">192.168.128.231:51350</a>
(type 03, len 000033)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:51350" target="_blank">192.168.128.231:51350</a>
(type 03, len 000033)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:51350" target="_blank">192.168.128.231:51350</a>
(type 03, len 000033)<br>
Sent RTP P2P packet to <a href="http://192.168.128.231:51350" target="_blank">192.168.128.231:51350</a>
(type 03, len 000033)<br>
== Spawn extension (local, 301, 1) exited non-zero
on 'SIP/hsolutionspf5-00000002'<u></u><u></u></p>
</blockquote>
<p class="MsoNormal">I tried many options to disable SRTP
but without success :<u></u><u></u></p>
<ul type="disc">
<li class="MsoNormal">canreinvite = no<u></u><u></u></li>
<li class="MsoNormal">canreinvite = nonat<u></u><u></u></li>
<li class="MsoNormal">srtpcapable=no<u></u><u></u></li>
<li class="MsoNormal">encryption=no<u></u><u></u></li>
<li class="MsoNormal">directmedia=nonat<u></u><u></u></li>
<li class="MsoNormal">...or noload => res_srtp.so in
modules.conf<u></u><u></u></li>
</ul>
<p class="MsoNormal" style="margin-bottom:12.0pt"><br>
Any help would be GREATLY appreciated !<br>
<br>
Denis<br>
<br>
P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11
(Final)<u></u><u></u></p>
</div>
<p class="MsoNormal"><br>
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</blockquote>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
</blockquote>
<br>
</div></div></div>
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Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: <a href="mailto:ish@pack-net.co.uk" target="_blank">ish@pack-net.co.uk</a>
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COMPANY REG NO. 04920552
</pre></div></div></div></div>
</div></div>