Hello developers ,
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.
Bu iş için 5 freelancer ortalamada $495 teklif veriyor
Dear Customer, this is Yaseen, and i am glad to work for you project, We specialise in Linux Unix VirtualMin Cpanel and Asterisk. I have read the project description and i assure you 100% quality and timely response to Daha Fazla
Hi, I'm working on VoIP many years. I have develop already a solution like this and it is tested and used with success. The last months I develop a complete new solution to solve more problems for the termination Daha Fazla
Good Day We have build a similiar solution with use of asterisk - a2billing Intergrated into this is osdial thru a2billing. We have a couple of asterisk boxes premise based that connect via public or point 2 point Daha Fazla
Hi I can do this for you .............................Also looking for good reviews .................>Thanks
I already did that for a grey route in paraguay. They had 120 channels and we didn't use asterisk, we use yate. At that moment for iax2 trunking to work you needed a digium hardware to keep the sync. You can expect 280 Daha Fazla
i can install it i make [url removed, login to view] i can make ur also just need a centos server & local end need a usb & a pc or router also we provied u a web base portal u can test it now we have allrady rady