Why don't you try 1.12.13? Besides fixing some security issues, it should work, I doing a SMB install this weekend (so no concern of billing/CDR really) so we didn't use notransfer=yes and it worked perfectly, even behind NAT.
<br><br>Also you should contact your ITSP, maybe they don't allow this?<br><br><div><span class="gmail_quote">On 11/5/06, <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;">Thanks for responding.<br><br>Yes I am doing pretty much exactly what you showed. When I try to
<br>dial without answering, I get a busy tone on the DiD (the local Telco<br>offers to let them notify me when it becomes available). Sometimes I<br>get half a ring on the destination cell phone b4 receiving the busy<br>
signal.<br><br>Not sure whether this sheds any light. I'm going to have to answer<br>the call anyway in order to implement an auto attendant, but it was<br>worth trying.<br><br>Cheers,<br>H<br><br><br>*CLI> -- Accepting AUTHENTICATED call from <ip>:
<br> > requested format = ulaw,<br> > requested prefs = (),<br> > actual format = g729,<br> > host prefs = (g729),<br> > priority = mine<br> -- Executing Dial("IAX2/<ITSP>-1", "IAX2/<ITSP>/6135551234") in new stack
<br> -- Called <ITSP>/6135551234<br> -- Call accepted by <ip> (format g729)<br> -- Format for call is g729<br> -- IAX2/<ITSP>-2 is making progress passing it to IAX2/<ITSP>-1<br> -- Hungup 'IAX2/<ITSP>-2'
<br> == Spawn extension (incoming-iax, 6135551234", 1) exited non-zero on<br>'IAX2/<ITSP>1'<br> -- Executing Hangup("IAX2/<ITSP>-1", "") in new stack<br> == Spawn extension (incoming-iax, h, 1) exited non-zero on 'IAX2/<ITSP>-1'
<br> -- Hungup 'IAX2/<ITSP>-1'<br><br><br><br>On 11/4/06, Andrew Joakimsen <<a href="mailto:joakimsen@gmail.com">joakimsen@gmail.com</a>> wrote:<br>> 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<br>> your ITSP configuration....
<br>><br>><br>> On 11/4/06, hugolivude <<a href="mailto:hugolivude@gmail.com">hugolivude@gmail.com</a>> wrote:<br>> ><br>> > 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
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