On Fri, Apr 25, 2008 at 5:15 PM, Andreas Sikkema <<a href="mailto:h323@ramdyne.nl">h323@ramdyne.nl</a>> wrote:<br><div class="gmail_quote"><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="Ih2E3d">On Apr 25, 2008, at 7:16 PM, Andreas Brodmann wrote:<br>
> What if once asterisk receives the command to initiate a call it<br>
> does so,<br>
> and when receiving a 484 it tells the other side anything like 484<br>
> (you're<br>
> on the right way but you are missing figures) and drops its newly<br>
> initiated<br>
> call. This continues until asterisk receives a 200 from the sip trunk.<br>
<br>
</div>Hmm, I'd not wait for a 200OK, but maybe for a 180 RINGING or 18x<br>
Session Progress with SDP? There might even be a standard that<br>
describes this kind of dialling ;-)</blockquote><div><br><br>RFC 3578 <a href="http://www.faqs.org/rfcs/rfc3578.html">http://www.faqs.org/rfcs/rfc3578.html</a> has some text on using 484 for mapping ISUP overlap dialing to SIP. I think this should work if the 484s received on the trunk are propagated over to the phone. This can probably be done by setting the hangup cause code when tearing down the phone leg of the call.<br>
<br></div></div>-- <br>Raj Jain<br><br>mailto:<a href="mailto:rj2807">rj2807</a> at gmail dot com<br>sip:rjain at iptel dot org