[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. 17.30 skrev 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
>
>
> I will have to clarify documentation here, because as I said, I hadn't
> thought of it from your perspective.
> We do support overlap on incoming calls, but not on outbound. My
> question, since this is the developer
> list, is how to implement this on the PBX to chan_sip interface - how
> would I know when to go into
> overlap mode on SIP. Or actually, if the sip trunk provider sent me a
> 484 - what would I return to the PBX
> to request more digits?


The goal is that if the sip trunk provider sends you a 484, the phone
which initiated the call will also receive a 484, so it knows it is on
the right track but the number is incomplete.

How to get there. I am not as familiar with asterisk's core as you are.
Normally phone A initiates a call/channel to asterisk. Asterisk will
initiate a call to another end point or to anything via a sip trunk. Once
the 2nd call setup is complete the calls/channels are bridged, right?

What if once asterisk receives the command to initiate a call it does so,
and when receiving a 484 it tells the other side anything like 484 (you're
on the right way but you are missing figures) and drops its newly initiated
call. This continues until asterisk receives a 200 from the sip trunk.

Possible like that?

-Andreas
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