[Asterisk-Users] New Asterisk bounty: SIP simultaneous

[Andy Powell]

On 11/07/2004 at 08:42 Paul Mahler wrote:

>You are confused about what a SIP session is and what a SIP session does.
> 
>
>SIP, session initiation protocol, controls an RTP, real time protocol,
>session between two IP endpionts. The end points have to have unique IP
>addresses for the session to run. The unique SIP registration is how *
>finds a UNIQUE endpoint. 

Sorry, but this is irrelavant... SIP allows multiple endpoints to register with the
same account details and will all ring when called. The fact that the rtp stream goes to the
first endpoint to pick up (and respond) is what's important ie, if multiple devices are registered 
with the same account they will *all* be 'spoken' to...  

Asterisk currently does not support this behaviour.


>You don't want SIP to solve your problem, you want * to solve your problem.
>You are asking for this SIP "feature" because you are confused as to how
>SIP and * work, and how they work together. 

No, the idea is to get asterisk to act like a real sip proxy. The dialplan solution is
a poor hack.


>You can easily fix your business problem with *, but not with mechanism you
>are asking for. You should spend your money on getting a copy of each of
>the
>two books that are now available and learn *. Then it will be clear to you
>that you don't really want what you are asking for. 

again, irrelavant - the whole beauty of the way SIP works is that I can add to
the list of phones that get called by simply registering more phones with the
same details. I don't need my users to mess with or make a support call to 
add to the dial plan. They can add and remove themselves.

I'd also suggest adding something like

registrationlimit=1 

for those that do not want to support multiple client registrations, I'd also like to
see the implementation of the q parameter...

I'm all for this modification to SIP, although I'd probably want to see DTMF callerid
implemented first :D

Andy