Low bandwidth voip solution by asterisk. exactly like syncswitch with encryption , if you have already developed a solution i would like to see it's demo first

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.


Beceriler: Asterisk PBX, Linux, PHP, VoIP

Daha fazlasını gör: bandwith sbo, www developed by, voip web server, pc for web developers, local web developers, linux developers, bandwidth com, a solution, VoIP termination, voip bandwidth optimization, syncswitch, sip router, setup voip solution, bandwidth, bandwidth optimization, iax2 sip, asterisk bandwidth work, voip web sip, g729 sip client, sip client linux, g729 convert, asterisk send, voip server setup, iax2 asterisk, linux sip voip

İşveren Hakkında:
( 0 değerlendirme ) Germany

Proje NO: #6542744

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

in %bids___i_period_sub_35% gün içinde631%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(562 Değerlendirme)

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

in %bids___i_period_sub_35% gün içinde777%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(6 Değerlendirme)

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

in %bids___i_period_sub_35% gün içinde555%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(0 Değerlendirme)

Hi I can do this for you .............................Also looking for good reviews .................>Thanks

in %bids___i_period_sub_35% gün içinde555%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(0 Değerlendirme)

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

in %bids___i_period_sub_35% gün içinde400%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(0 Değerlendirme)

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

1 gün içinde %bids___i_sum_sub_32%%project_currencyDetails_sign_sub_33% USD
(0 Değerlendirme)