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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:10.0pt;font-family:"Verdana","sans-serif";color:#1F497D'>Ask with you SIP carrier which dtmfmode they are using on their end and use same on asterisk side.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Verdana","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>asterisk asterisk<br><b>Sent:</b> Wednesday, February 16, 2011 8:58 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] DTMF not detected, time out<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>In the past it was set as auto and worked. I change to RFC2833 but did not work.<br><br>How can I check further?<br><br><br><o:p></o:p></p><div><p class=MsoNormal>On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif <<a href="mailto:faisal@vopium.com">faisal@vopium.com</a>> wrote:<o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;color:#1F497D'>Check if dtmfmode is properly set on SIP trunk ask with your carrier which dmtfmode they support.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:10.0pt'>From:</span></b><span style='font-size:10.0pt'> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>asterisk asterisk<br><b>Sent:</b> Wednesday, February 16, 2011 5:39 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> [asterisk-users] DTMF not detected, time out</span><o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'>Hi, <br><br>I encounter this problem recently after quite some months of my asterisk.<br><br>I have a SIP trunk for dial in and out.<br>When dial-in, it matches the callerid number and decides. If matched, it will either go into an a very brief IVR. The IVR allows caller to dial internal extension.<br>All along it is working well both from outside call and internal users.<br>Now for unknown reason, it fails with a timeout and hangup. It is the only message I can see at the console.<br>But internal user can do this without any problem.<br><br>Appreciate if someone can help.<br><br>CK<o:p></o:p></p></div></div></div></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></div><p class=MsoNormal><o:p> </o:p></p></div></body></html>