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 , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip 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 .
Now i connect Server to local PC over SIP to SIP
its use per call 24KB+
i need under 12 KBIf you can make it please bid
waiting for your update.
Bu iş için 3 freelancer ortalamada $724 teklif veriyor
Hi, I have lot of experience in asterisk termination and voip. I have the solution for your issue. We can chat details.
Hello, Ilan here , hope all is well on your side . Will configure environment include isolated configurations for experience performance output on back-end as well. Looking forward hearing from you . LPIC , CEH , Daha Fazla
I am an IT engineer since 2013. My major was telecommunications and networks. I have 7 years of experience into Voip solutions as Asterisk or Kamailio. I have also a certifcation in asterisk basics with a selp paced Daha Fazla