Neil,<br><br>Howdy! Long time no chat. :) <br><br>Regarding SIP/VoIP: A SIP call will not come in over a PRI circuit. SIP is VoIP and PRI is TDM - two totally different methods for carrying telephone calls. My guess is that you might have a bit of confusion about where and how calls are coming in. Let&#39;s start with the basics: Do you have a SIP/VoIP provider? If so, did they supply you with phone number(s) for dialing in to your system? Have you gone through the configuration process to make sure your FS box is properly communicating with the provider?<br>
<br>If you need some help you can contact me off list. Of course, this means you&#39;ll have to take me to that Mexican food place the next time I&#39;m in Frisco! :)<br><br>-MC<br><br><div class="gmail_quote">On Sat, Mar 24, 2012 at 11:18 AM, Neil Patel <span dir="ltr">&lt;<a href="mailto:neilp@cs.stanford.edu">neilp@cs.stanford.edu</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I have not modified the sip_profiles directory from standard. There is no variable set for dtmf-type in either internal.xml or external.xml.<div>
<br></div><div>I did a trace using:</div><div>sofia global siptrace on</div><div>


<br></div><div>There wer no SIP packets logged. If the SIP call is going to a PRI profile (freetdm), will there still be SIP traffic?</div><div><br></div><div>Thanks,</div><div>Neil<div><div class="h5"><br><br><div class="gmail_quote">
On Sat, Mar 24, 2012 at 7:22 PM, Anita Hall <span dir="ltr">&lt;<a href="mailto:anita.hall@simmortel.com" target="_blank">anita.hall@simmortel.com</a>&gt;</span> wrote:<br>


<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">What is your config for DTMF in sip profile ?<br><br><a href="http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#dtmf-type" target="_blank">http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#dtmf-type</a><br>



<br>Use tshark to generate a SIP trace and paste it on <a href="http://pastebin.freeswitch.org/" target="_blank">http://pastebin.freeswitch.org/</a> and give the link here. <br>
<br clear="all">regards,<br>Anita<br><br>
<br><br><div class="gmail_quote"><div><div>On Sat, Mar 24, 2012 at 2:43 PM, Neil Patel <span dir="ltr">&lt;<a href="mailto:neilp@cs.stanford.edu" target="_blank">neilp@cs.stanford.edu</a>&gt;</span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>
Hi All,<div><br></div><div>I have a basic IVR application in Lua connected to a PRI line. It currently is not responding to DTMF input given from any VoIP call (e.g. Skype). However, it accepts input from local mobile or landline calls just fine.</div>






<div><br></div><div>I recently pulled latest from git and built; before that, DTMF detection from both were working. Is there something I need to configure to allow DTMF for VoIP? </div><div><br></div><div>Thanks,</div><div>






Neil<br>
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