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

[Andreas Brodmann]

2008/4/25, Johansson Olle E <oej at edvina.net>:
>
>
> 25 apr 2008 kl. 16.44 skrev Andreas Brodmann:
>
>
> > 2008/4/25, Kevin P. Fleming <kpfleming at digium.com>: Andreas Brodmann
> > wrote:
> > > After chasing a problem I looked at the SIP code (1.4.19)
> > > with a colleague and as far as we understood it, overlapped
> > > dialing on sip trunks for outgoing calls is not supported (yet?).
> >
> >
>
> > in sip.conf you can set 'allowoverlap=yes'
> >
> > If a phone is configured accordingly (early dial) it will send an
> > INVITE request
> > for each key a user presses. e.g. 1 at asterisk, 11 at asterisk and then
> > 111 at asterisk.
> > For each incomplete INVITE asterisk will return 484 "Number
> > incomplete" until the
> > client sends a number which is complete (e.g. matches a pattern).
> >
> > This works fine until a client tries to call a number that asterisk
> > reaches via a sip trunk to
> > another pbx (or carrier), whereas the length of the number is unknown:
> > // e.g. International Calls
> > e.g. exten => 000!,1,Dial(SIP/${EXTEN:1}@carrier,120)
> >
> > in this case, asterisk will send an INVITE to the carrier after the
> > first 3 zeros. The answer
> > from the carrier will be 484 (number incomplete). Instead of
> > forwarding this response to the
> > phone asterisk will end the call -> congestion.
> >
>
> That's another issue. Outbound overlap dialling is something that is
> propably not implemented.
> That will require a lot of coding I think, but other developers might
> understand overlap dialling
> *through* asterisk better.
>
> For PRI, I believe we put the call in UP state and then simply forward
> dtmf...


Olle

this would mean that either you use PRIs to your carrier or you cannot
use overlapped dialing with sip in asterisk at all, because you cannot
have the phones use overlap and the sip trunk to the carrier not use
overlap, right?

-> global on or global off

-Andreas
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