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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Netstat –anp has been useful in finding this error for me in the past. A “normal” Asterisk call will have 2 or 4 udp connections to carry traffic to/from phone to pbx. On a one-way call, this will be an “odd” count. Then you can check your rtp.conf and firewall and see how the channel got blocked.<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:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>John Knight<br><b>Sent:</b> Friday, October 14, 2011 8:49 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] one way voice with IVR<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>Hi Giorgio,<br><br>This behavior usually indicates some sort of firewall issue where either inbound or outbound rtp traffic (the voice) is being blocked or not routed correctly, though the SIP traffic makes it through (as the call is being set up correctly). This could also be multiple SIP extensions attempting to register over the same port from a single location.<br><br>What kind of firewall/router is being used at the location where these Snoms are registering from? Are all the phones attempting to register over port 5060 or are you setting them up to register over unique ports to Asterisk? If you are setting them up to register over specific ports, are they registering over those ports according to 'asterisk show peers'? Also, is your asterisk box local or hosted somewhere?<br><br>Comparing IAX2 to SIP registrations is somewhat different: IAX2 tends to handle cutting through firewalls better though SIP is far better supported by everyone.<br><br><br><br><o:p></o:p></p><div><p class=MsoNormal>On Fri, Oct 14, 2011 at 6:21 AM, gincantalupo <<a href="mailto:gincantalupo@fgasoftware.com">gincantalupo@fgasoftware.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Hi all,<br><br>I'm stuck on a tricky problem.<br>I set up an Asterisk 1.4.26.2 on a box with a bunch of Snom Phones. When I call an IVR I get the damned one way voice phenomena. It is not randomic, it happens all the time.<br>I tried to upgrade the snom firmware to 7.3.30 but nothing changed.<br>If I call a phone I get a normal conversation and no problem occurs if I (blind) transfer the call.<br>If I use a IAX phone everything is fine.<br>I think it is a SIP problem but I checked the sip files and they seem ok.<br>Tones seems to pass since the caller (me) can make a choice from within the IVR menu.<br><br>Sincerely, I haven't any idea left to try...<br><br>Any hints?<br><br>Thanks<br><br>Giorgio<br><span style='color:#888888'><br><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></span><o:p></o:p></p></div><p class=MsoNormal><br><br clear=all><br>-- <br><b><span style='font-size:13.5pt;font-family:"Arial","sans-serif";color:#3366FF'><br></span></b><span style='color:#888888'><img border=0 width=158 height=65 id="_x0000_i1025" src="cid:image001.png@01CC8A4E.B567E970"><o:p></o:p></span></p><p><b><span style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:#777777'>John Knight</span></b><span style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:#777777'><br>Classic City Telco LLC<br><b>Email:</b> <a href="mailto:john@classiccitytelco.com" target="_blank">john@classiccitytelco.com</a> | <b>Main:</b> (706) 995-0200<br><b>Direct:</b> (706) 995-0201 | <b>Mobile:</b> (706) 255-9203</span><span style='color:#888888'><o:p></o:p></span></p><p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:7.5pt;font-family:"Arial","sans-serif"'><br><br></span><o:p></o:p></p></div></body></html>