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<div class="moz-cite-prefix">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 ?<br>
<br>
Denis<br>
<br>
Le 12.11.2015 17:05, Sam Basan a écrit :<br>
</div>
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<div class="WordSection1">
<p class="MsoNormal"><a moz-do-not-send="true"
name="_MailEndCompose"><span
style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D">Snom
default configuration is SRTP enabled.<o:p></o:p></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<o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"><o:p> </o:p></span></p>
<p class="MsoNormal"><b><span
style="font-size:14.0pt;font-family:"Brush Script
MT";color:#003366"><o:p> </o:p></span></b></p>
<p class="MsoNormal"><b><span
style="font-size:14.0pt;font-family:"Brush Script
MT";color:#003366"><o:p> </o:p></span></b></p>
<p class="MsoNormal"><b><span
style="font-size:14.0pt;font-family:"Brush Script
MT";color:#003366">Sincerely,<o:p></o:p></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 id="Picture_x0020_4"
src="cid:part2.03050604.03020003@hsolutions.ch"
alt="cid:image001.jpg@01D0D5C4.27A0CBA0" height="56"
width="87"><o:p></o:p></span></p>
<p class="MsoNormal"><b><span
style="font-size:14.0pt;font-family:"Brush Script
MT";color:#003366">Sam Basan</span></b><span
style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"><o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"><img
id="Picture_x0020_2"
src="cid:part3.08060604.05070309@hsolutions.ch"
alt="cid:image003.png@01C918DA.6B3E4530" height="48"
width="397"><o:p></o:p></span></p>
<p class="MsoNormal"><span
style="font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D"><o:p> </o:p></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 class="moz-txt-link-freetext" href="mailto:mitul@enterux.in">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 class="moz-txt-link-rfc2396E" href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a><br>
<b>Subject:</b> Re: [asterisk-users] No sound with internal
calls depending on which phones<o:p></o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p>You might have to disable srtp negotiations inside the phone
web ui options. <o:p></o:p></p>
<p>Mitul <o:p></o:p></p>
<div>
<p class="MsoNormal">On Nov 12, 2015 8:53 PM, "(lists) Denis
BUCHER" <<a moz-do-not-send="true"
href="mailto:dbucherml@hsolutions.ch">dbucherml@hsolutions.ch</a>>
wrote:<o:p></o:p></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 :<o:p></o:p></p>
<ul type="disc">
<li class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l1
level1 lfo1">external calls work perfectly,<o:p></o:p></li>
<li class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l1
level1 lfo1">internal calls between some phones too,<o:p></o:p></li>
<li class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l1
level1 lfo1">but internal call between two similar
phones don't work !!! (Snom 710)<o:p></o:p></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 :<o:p></o:p></p>
<ul type="disc">
<li class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l2
level1 lfo2">[Nov 10 17:51:47] ERROR[21480]:
chan_sip.c:28306 setup_srtp: No SRTP module loaded,
can't setup SRTP session.<o:p></o:p></li>
</ul>
<p class="MsoNormal">This is a working internal call :<br>
<br>
<o:p></o:p></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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
href="http://192.168.128.99:49646" target="_blank">192.168.128.99:49646</a>'<br>
Sent RTP P2P packet to <a moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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'<o:p></o:p></p>
</blockquote>
<p class="MsoNormal">This is a non-working call :<br>
<br>
<o:p></o:p></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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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 moz-do-not-send="true"
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'<o:p></o:p></p>
</blockquote>
<p class="MsoNormal">I tried many options to disable SRTP
but without success :<o:p></o:p></p>
<ul type="disc">
<li class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0
level1 lfo3">canreinvite = no<o:p></o:p></li>
<li class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0
level1 lfo3">canreinvite = nonat<o:p></o:p></li>
<li class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0
level1 lfo3">srtpcapable=no<o:p></o:p></li>
<li class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0
level1 lfo3">encryption=no<o:p></o:p></li>
<li class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0
level1 lfo3">directmedia=nonat<o:p></o:p></li>
<li class="MsoNormal"
style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto;mso-list:l0
level1 lfo3">...or noload => res_srtp.so in
modules.conf<o:p></o:p></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)<o:p></o:p></p>
</div>
<p class="MsoNormal"><br>
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