<br><br><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. 17.30 skrev Andreas Brodmann:<br> <br><br> > 2008/4/25, Johansson Olle E <<a href="mailto:oej@edvina.net">oej@edvina.net</a>>:<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<br> > 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...<br> ><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> <br>I will have to clarify documentation here, because as I said, I hadn't<br> thought of it from your perspective.<br> We do support overlap on incoming calls, but not on outbound. My<br> question, since this is the developer<br>
list, is how to implement this on the PBX to chan_sip interface - how<br> would I know when to go into<br> overlap mode on SIP. Or actually, if the sip trunk provider sent me a<br> 484 - what would I return to the PBX<br>
to request more digits?</blockquote><div><br>The goal is that if the sip trunk provider sends you a 484, the phone<br>which initiated the call will also receive a 484, so it knows it is on<br>the right track but the number is incomplete.<br>
<br>How to get there. I am not as familiar with asterisk's core as you are.<br>Normally phone A initiates a call/channel to asterisk. Asterisk will<br>initiate a call to another end point or to anything via a sip trunk. Once<br>
the 2nd call setup is complete the calls/channels are bridged, right?<br><br>What if once asterisk receives the command to initiate a call it does so,<br>and when receiving a 484 it tells the other side anything like 484 (you're<br>
on the right way but you are missing figures) and drops its newly initiated<br>call. This continues until asterisk receives a 200 from the sip trunk.<br><br>Possible like that?<br><br>-Andreas<br><br></div><br></div><br>