SIP to SIP calls
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we need basically bandwith optimization. in client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should send to specific termination .
call should pass with sip / iax2 , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip / IAx to sip and pass the calls to gateway .
01. asterisk or SBO server. which receive calls from many sip server.
02. asterisk./SBO transfer calls to local PC or router .
03. in router / pc have a module which can route calls to LAN IP .
04. in LAN IP there have a termination gateway . so calls can pass normal .
the thing is , u need to work on asterisk/ SBO server and client end PC/ router.
for your understand . here i give you a web site . as they provide .. we need same solution,
[url removed, login to view]
check the site . we need same solution.
waiting for your update.
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