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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><a 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 width=87 height=56 id="Picture_x0020_4" src="cid:image001.jpg@01D11D74.BBBA0FC0" alt="cid:image001.jpg@01D0D5C4.27A0CBA0"><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 width=397 height=48 id="Picture_x0020_2" src="cid:image002.png@01D11D74.BBBA0FC0" alt="cid:image003.png@01C918DA.6B3E4530"><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 [mailto:mitul@enterux.in] <br><b>Sent:</b> Thursday, November 12, 2015 5:25 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><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 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 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'<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 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'<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>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></blockquote></div></div></body></html>