4 jan 2010 kl. 14.52 skrev Antonio Goméz Soto:
Hi Daniel.
Op 04-01-10 14:11, Daniel-Constantin Mierla schreef:
I would try to do:
- partition the users per asterisk instance - as much as possible host
the users talking to each other in same asterisk
- use kamailio as central registry server - if the destination user is
not on the same asterisk instance, forward it to kamailio and from there to the right asterisk instance
It's not clear to me what this means. The phones register to Kamailio as their registrar server. And they will send INVITE's to their proxy (i.e. the asterisk server?). So only Kamailio will know the ip address of a phone (except if asterisk accesses the same database using realtime - is that what you mean?)
Well, if you need Asterisk states, you need to register with Asterisk and not with Kamailio. Or possibly both.
But if you have a set of phones that can run "simple" services, then you can let them register with Kamailio and have a gold service running on a full Asterisk PBX. SIP is designed in a way to let the phones be in control of a lot of services. Even though you're using a proxy, services like call transfer, three-way conferencing and call hold will work - if the phones support it. Most business-level IP phones does support a lot of these advanced services.
/O