<div><span class="gmail_quote">2008/4/25, Johansson Olle E <<a href="mailto:oej@edvina.net">oej@edvina.net</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br> 25 apr 2008 kl. 16.44 skrev Andreas Brodmann:<br> <br><br> > 2008/4/25, Kevin P. Fleming <<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>>: Andreas Brodmann<br> > wrote:<br> > > After chasing a problem I looked at the SIP code (1.4.19)<br>
> > with a colleague and as far as we understood it, overlapped<br> > > dialing on sip trunks for outgoing calls is not supported (yet?).<br> ><br> ><br> <br>> in sip.conf you can set 'allowoverlap=yes'<br>
><br> > If a phone is configured accordingly (early dial) it will send an<br> > INVITE request<br> > for each key a user presses. e.g. 1@asterisk, 11@asterisk and then<br> > 111@asterisk.<br> > For each incomplete INVITE asterisk will return 484 "Number<br>
> incomplete" until the<br> > client sends a number which is complete (e.g. matches a pattern).<br> ><br> > This works fine until a client tries to call a number that asterisk<br> > reaches via a sip trunk to<br>
> another pbx (or carrier), whereas the length of the number is unknown:<br> > // e.g. International Calls<br> > e.g. exten => 000!,1,Dial(SIP/${EXTEN:1}@carrier,120)<br> ><br> > in this case, asterisk will send an INVITE to the carrier after the<br>
> first 3 zeros. The answer<br> > from the carrier will be 484 (number incomplete). Instead of<br> > forwarding this response to the<br> > phone asterisk will end the call -> congestion.<br> ><br> <br>That's another issue. Outbound overlap dialling is something that is<br>
propably not implemented.<br> That will require a lot of coding I think, but other developers might<br> understand overlap dialling<br> *through* asterisk better.<br> <br> For PRI, I believe we put the call in UP state and then simply forward<br>
dtmf...</blockquote><div><br>Olle<br><br>this would mean that either you use PRIs to your carrier or you cannot<br>use overlapped dialing with sip in asterisk at all, because you cannot<br>have the phones use overlap and the sip trunk to the carrier not use<br>
overlap, right?<br><br>-> global on or global off<br><br>-Andreas</div></div><br>