Andreas Brodmann wrote:
> 2008/4/28 Tilghman Lesher <tilghman at mail.jeffandtilghman.com
> <mailto: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 <http://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.
How do SIP phones with overlap dialing handle the following scenario:
1. User presses "1"
2. SIP phone sends INVITE sip:1 at ....
3. user presses "2"
4. what happens now? does the phone CANCEL sip:1 at ... and INVITE sip:12 at ..?
regards
klaus