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

[Raj Jain]

On Fri, Apr 25, 2008 at 1:16 PM, Andreas Brodmann <
andreas.brodmann at gmail.com> wrote:

>
>
> 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
>
>

Would this work if you hangup the phone leg with a 484 when you receive a
484 on the trunk? This way you won't need to keep the call up in Asterisk
and have the dial-plan act as pass-through across INVITE/484 pairs.

-- 
Raj Jain

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