<p>I think its not inbound call its outgoing, and during call progress the remote end events are not passing back to sip.</p>
<p>Mitul</p>
<div class="gmail_quote">On Jul 27, 2012 10:36 PM, "Raj Mathur (राज माथुर)" <<a href="mailto:raju@linux-delhi.org">raju@linux-delhi.org</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
On Friday 27 Jul 2012, Tim Nelson wrote:<br>
> Another mystery for the list, hopefully someone has ideas on a fix...<br>
> :)<br>
><br>
> I've got an Asterisk 1.8.12.0 system connected to a CAS T1 (ESF/B8ZS,<br>
> fractional 1-8). Outbound dialing works correctly, but while the<br>
> call is in progress, there is no 'ringing' heard by the end user.<br>
> So, on a SIP phone connected to this system, I dial a number, that<br>
> call goes out DAHDI via the CAS T1, and the remote side is actually<br>
> ringing (my cell phone for example), but the SIP phone remains<br>
> silent. If I answer my cell phone, full 2-way audio is present.<br>
<br>
Do you Answer() the SIP phone before dialling DAHDI?<br>
<br>
Regards,<br>
<br>
-- Raj<br>
--<br>
Raj Mathur || <a href="mailto:raju@kandalaya.org">raju@kandalaya.org</a> || GPG:<br>
<a href="http://otheronepercent.blogspot.com" target="_blank">http://otheronepercent.blogspot.com</a> || <a href="http://kandalaya.org" target="_blank">http://kandalaya.org</a> || CC68<br>
It is the mind that moves || <a href="http://schizoid.in" target="_blank">http://schizoid.in</a> || D17F<br>
<br>
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