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

[Raj Jain]

On Mon, Apr 28, 2008 at 4:41 PM, Tilghman Lesher <
tilghman at mail.jeffandtilghman.com> wrote:

> On Monday 28 April 2008 14:41, Andreas Brodmann wrote:
> > 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)
>
> That will never work.  Calling the Incomplete application means that the
> dialplan will wait for more digits (and so the dialplan will restart).
>  You
> need a more exact extension in the same context (or else the Incomplete
> app
> needs to go into an included context).


May be I'm missing something basic and someone can help clarify it for me.
I've dealt with this with Cisco Call Manager 6.X and Cisco 7960 phones that
support KPML (KeyPad Markup Language; a dialplan language expressed in XML)
that support this feature. The part that escapes me is that why do we need
Asterisk dial-plan to be stateful across INVITE/484 pairs if both the SIP
phone and the SIP trunk understand the 484 semantics.

Why can't we treat 484 received on the SIP trunk as a failed call and
propagate that specific response code to the phone and eliminate that state
from the Asterisk dial-plan?  Remember that the phone sends the previous
digits in the new INVITEs after receiving 484s so that the To: or R-URI
eventually satisfies the SIP trunk provider's routing plan.

But like I said, I may be missing something basic. What would that be?

-- 
Raj Jain

mailto:rj2807 at gmail dot com
sip:rjain at iptel dot org
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