When you say you answer the call, I assume you have something like this:<br><br>exten => 5551212,1,Answer<br>exten => 5551212,1,Dial(SIP/provider/10005551212)<br><br>Try to not answer the call and see if the behviour changes, it could just be your ITSP configuration....
<br><br><div><span class="gmail_quote">On 11/4/06, <b class="gmail_sendername">hugolivude</b> <<a href="mailto:hugolivude@gmail.com">hugolivude@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Asterisk 1.2.7<br>RedHat 9.0<br><br>I frequently have the need to redirect calls that come in on a DiD<br>provisioned by my ITSP, back to the ITSP so that they can terminate<br>the call on the PSTN. For example when an external call comes in, I
<br>often have to send it to a cell phone. I believe that this is<br>referred to as "hairpinning" the call.<br><br>I do this by answering the incoming call and then I use a simple<br>"dial" command to send it back to my ISTP using a SIP or IAX channel
<br>and the ITSP terminates it on the cell phone. One of my main goals<br>is to keep my Asterisk box out of the media path and let the ITSP<br>handle all the provisioning for the call. I understand that the<br>default behaviour of the "dial" command is supposed to do just that,
<br>but I've run into problems though on both SIP & IAX channels.<br><br>With IAX I use a simple dial command:<br><br> Dial(IAX2/myIAX/7775551234)<br><br>Things seem to work great, I can see the handshaking in the CLI as the
<br>call gets redirected and once both end points are connected, I can<br>actually disconnect my box from the ethernet and the call is<br>uninterruoted. Unfortuanately the call quality is terrible! Low<br>volume, choppy and so on.
<br><br>It seemed to me that since I had stepped my * box out of the network,<br>the problem must be with the ITSP. They suggested I try SIP.<br><br>With SIP I use:<br><br> Dial(SIP/7775551234@mySIP)<br><br>Unfortuantely I don't get the handshakes and the whole call ends up
<br>passing through my box, which is something I'm desperate to avoid. I<br>have canreinvite=yes as seen from my sip.conf:<br><br> [mySIP]<br><br> type=peer<br><br> auth=md5<br><br> username=<UID><br> fromuser=<UID>
<br> fromdomain=<domain><br><br> secret=<pw><br> host=<domain><br><br> port=5060<br><br> nat=yes<br><br> canreinvite=yes<br><br> qualify=no<br><br> disallow=all<br><br> allow=g729<br><br>
dtmfmode=rfc2833<br><br> insecure=very<br> context=incoming-sip<br><br><br>Now the questions:<br><br>1) Given that I can see the handshaking and I can disconnect my * box<br>during the call, I think that the IAX call quality problems are on my
<br>ITSP's end, but I could be wrong. Is there anything I can do to<br>improve call quality when using IAX this way?<br><br>2) What about SIP? Why doesn't that work? I always thought that<br>"dial" would do exactly what I'm after (hairpin/redirect the call) if
<br>I avoided options like t or T.<br><br>Any direction you can provide is highly appreciated.<br><br>Thanks,<br>H<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">
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