<div class="gmail_quote">2008/4/28 Tilghman Lesher <<a href="mailto:tilghman@mail.jeffandtilghman.com">tilghman@mail.jeffandtilghman.com</a>>:<br><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 Monday 28 April 2008 02:15, Klaus Darilion wrote:<br>
> Johansson Olle E schrieb:<br>
> > 25 apr 2008 kl. 17.34 skrev Tzafrir Cohen:<br>
> >> On Fri, Apr 25, 2008 at 05:25:33PM +0200, Andreas Brodmann wrote:<br>
> >>> I read the docs before writing here and I used allowoverlap=yes.<br>
> >>><br>
> >>> What I would like to know from you is wheter this is supported for<br>
> >>> phones only (as the example below shows) or wheter it is supposed<br>
> >>> to work on sip trunks for 'outgoing' calls:<br>
> >>><br>
> >>> 1) in sip.conf: [general] allowoverlap=yes<br>
> >>> 2) asterisk sends an INVITE to a carrier<br>
> >>> 3) carrier sends 484 back<br>
> >>> 4) asterisk sends congestion msg to phone.<br>
> >><br>
> >> Can you emulate it in the dialplan?<br>
> ><br>
> > Reading the source, if we get 484 we end up here:<br>
> ><br>
> > Case 484: /* Address incomplete */<br>
> > return AST_CAUSE_INVALID_NUMBER_FORMAT<br>
> ><br>
> > And that's an error returned to the dialplan and the dialplan will<br>
> > have to try again.<br>
><br>
> Does that mean that Dial() has to be extended ?<br>
<br>
</div>I did some work on this a couple weeks ago, to support a slightly different<br>
use of incomplete matching, so I just extended it to support SIP overlap<br>
dialling. See issue #12351.</blockquote><div><br> I just had a look at your issue @ <a href="http://bugs.digium.com">bugs.digium.com</a><br><br>One thing I am not sure about - would your patch also cover the following scenario<br>
for international numbers with unknown length?<br><br>exten => _000!,1,Incomplete<br>exten => _000!,n,Dial(SIP/${EXTEN:1}@carrier,120)<br>.<br>.<br>If so, is there a backport to 1.4.x for this? If not, would it be difficult to create?<br>
Basically all I want is asterisk to wait for an interdigit timeout for international numbers,<br>because their length is unknown. If this combination of phones using SIP overlap dialing<br>and asterisk still waiting for an interdigit timeout before sending 484 to the phone would<br>
be possible that opened a completely new perspective to real 'VoIP only' environments.<br><br>-Andreas<br><br></div></div>