[asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls

[Andreas Brodmann]

2008/4/28 Tilghman Lesher <tilghman at mail.jeffandtilghman.com>:

> On Monday 28 April 2008 02:15, Klaus Darilion wrote:
> > Johansson Olle E schrieb:
> > > 25 apr 2008 kl. 17.34 skrev Tzafrir Cohen:
> > >> On Fri, Apr 25, 2008 at 05:25:33PM +0200, Andreas Brodmann wrote:
> > >>> I read the docs before writing here and I used allowoverlap=yes.
> > >>>
> > >>> What I would like to know from you is wheter this is supported for
> > >>> phones only (as the example below shows) or wheter it is supposed
> > >>> to work on sip trunks for 'outgoing' calls:
> > >>>
> > >>> 1) in sip.conf: [general] allowoverlap=yes
> > >>> 2) asterisk sends an INVITE to a carrier
> > >>> 3) carrier sends 484 back
> > >>> 4) asterisk sends congestion msg to phone.
> > >>
> > >> Can you emulate it in the dialplan?
> > >
> > > Reading the source, if we get 484 we end up here:
> > >
> > > Case 484:       /* Address incomplete */
> > >                          return AST_CAUSE_INVALID_NUMBER_FORMAT
> > >
> > > And that's an error returned to the dialplan and the dialplan will
> > > have to try again.
> >
> > Does that mean that Dial() has to be extended ?
>
> I did some work on this a couple weeks ago, to support a slightly
> different
> use of incomplete matching, so I just extended it to support SIP overlap
> dialling.  See issue #12351.


 I just had a look at your issue @ bugs.digium.com

One thing I am not sure about - would your patch also cover the following
scenario
for international numbers with unknown length?

exten => _000!,1,Incomplete
exten => _000!,n,Dial(SIP/${EXTEN:1}@carrier,120)
.
.
If so, is there a backport to 1.4.x for this? If not, would it be difficult
to create?
Basically all I want is asterisk to wait for an interdigit timeout for
international numbers,
because their length is unknown. If this combination of phones using SIP
overlap dialing
and asterisk still waiting for an interdigit timeout before sending 484 to
the phone would
be possible that opened a completely new perspective to real 'VoIP only'
environments.

-Andreas
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20080428/961581bc/attachment.htm