[Asterisk-Users] SER Interaction: Agents and Extensions

[brett-asterisk at worldcall.net]

Michael Welter wrote:

> SER newbie here.  Why do you need Asterisk for Sip->SIP setup?  And if 
> there is a reinvite, is that for the RTP stream only or for the SIP 
> transactions as well?  Will you lose the BYE transaction if there is a 
> reinvite?
>
> Also, how many SIP registrations do you expect to maintain on each SER 
> box?
>
Good questions.
I need asterisk for SIP->SIP setups because it can do "interesting" 
things with the call. Like capture digits, play prompts,listen for key 
codes during a live coversation to transfer and other neat things. As 
far as I know (and please correct me if I am wrong!) but SER can't do 
these kinds of things since it's totally disconnected from the RTP Stream.

As for the reinvite, I'm *hoping* that it's just the establishment of 
the RTP stream and not the entire signalling path. However I could be 
wrong. Looking for feedback from anyone who knows better.

I'm expecting SER to maintain "thousands" of registrations.

To quote the SER Admin manual:
"With a $3000 dual-CPU PC, the SIP Express Router is able to power IP 
telephony services in an area as large as the Bay Area during  peak 
hours. Even on an IPAQ PDA, the server withstands 150 calls per second 
(CPS)!"

That is why you want SER.

The two together provide features + scalability.

I'm still not sure how to provide services that interact one phone with 
another phone's RTP stream. Like call pickup. How can I pickup a call on 
another asterisk server? Hmm Hmmmmmmmmm

-Brett