[Asterisk-Users] Asterisk PBX Functions via SIP phone

[Chris Shaw]

----- Original Message -----
From: "Olle E. Johansson" <oej at edvina.net>
To: <asterisk-users at lists.digium.com>
Sent: Friday, August 20, 2004 1:37 PM
Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone


> Chris Shaw wrote:
>
> >>I am suprised that one would have to create a dialplan since its an
> >
> > already built in function that works with regular POTS phones. Or is it
> > because of the way DTMF is sent via SIP?
> >
> > I don't know digium's long range plans, but looking through chan_sip.c
NONE
> > of the vertical service codes are mentioned anywhere... A quick look
through
> > chan_zap reveals all of them... So for right now it's not implemented in
> > SIP...
>
> Well, here we stumble over the SIP religion again.
>
> First, a phone connected to an RJ11 jack in a Digium card is a stupid
phone. All
> the intelligence lies in the zaptel driver and asterisk.
>
> Most SIP phones are more clever (at least expected to be much more clever
than
> the GS :-).
>
> Look at the SIPURA, where you are able to implement vertical service codes
> in the SIPura. Asterisk should not bother with DND and forwards, the SIP
phone
> does. Just send the call to the phone. Some of these phones are complete
> Linux systems with IPsec, multiple lines and a lot of routing
intelligence.
>
> There's also a discussion between Asterisk developers on whether these
> codes should be fixed in the channel or in the dial plan. At least, they
> should be configurable since there's no global standard (again).
> Or there may be, but there are still differences between countries
> and providers...
>
> * Executive summary: SIP is designed for very intelligent end-points.
> * A PBX with analogue lines is designed for central intelligence.
> * Asterisk will always be in the middle of these kind of discussions,
>    and it'll be fun each time we try to sort it out.
>
> /Olle

No, I agree completely with the way it works now, in fact I think it
SHOULDN'T be implemented in SIP myself... Doing it in the dialplan (if your
phone doesn't support it) works fine and doesn't break anything (that's the
key right there). We need some more docs on how to do different things and
I'm sure many people could contribute those, myself included... Some already
have...

The only thing is, if any of the apps you've written in your dialplan become
obsoleted or change syntax, your whole implementation will get screwed
over... I guess that's true with anything though...

    -Chris