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><br><div class="gmail_quote">On Wed, Feb 16, 2011 at 10:30 AM, Faisal Hanif <span dir="ltr"><<a href="mailto:faisal@vopium.com">faisal@vopium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div link="blue" vlink="purple" lang="EN-US"><div><p class="MsoNormal"><span style="font-size: 10pt; color: rgb(31, 73, 125);">Check if dtmfmode is properly set on SIP trunk ask with your carrier which dmtfmode they support.</span></p>
<p class="MsoNormal"><span style="font-size: 10pt; color: rgb(31, 73, 125);"> </span></p><p class="MsoNormal"><b><span style="font-size: 10pt;">From:</span></b><span style="font-size: 10pt;"> <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></p><div><div></div><div class="h5">
<p class="MsoNormal"> </p><p class="MsoNormal" style="margin-bottom: 12pt;">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</p></div></div></div></div><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><br></blockquote></div><br>