[Asterisk-Users] Asterisk PBX Functions via SIP phone

[Olle E. Johansson]

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