Suljettu

Low bandwidth voip solution by asterisk.

well ,

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.

regards,

Md. Abu Noman

1 # VOIP Call Termination Over Local IP ( VOIP Over NAT Traversal Or Firewall )
2 # VOIP Call & Quality Voice Termination In Lower Bandwidth ( 1 Call 6-8kb )
3 # Web Based Administrator, Agent & Client Control ( Easy User Interface)
4 # Easy New Gateway Add System For Client ( Add Gateway:- PC MAC Address & Gateway Local IP )
5 # Stay Alive Internet System ( As Like Skype )
6 # Easy Asterisk Billing System
7 # Easy Bootable USB Client
8 # VOIP Server Connect The Common Configuration Client Over Local IP & MAC Address.

well ,

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
05. VOIP Call Termination Over Local IP ( VOIP Over NAT Traversal Or Firewall )
06. VOIP Call & Quality Voice Termination In Lower Bandwidth ( 1 Call 6-8kb )
07. Web Based Administrator, Agent & Client Control ( Easy User Interface)
08. Easy New Gateway Add System For Client ( Add Gateway:- PC MAC Address & Gateway Local IP )
09. Stay Alive Internet System ( As Like Skype )
10. Easy Asterisk Billing System
11. Easy Bootable USB Client
12. VOIP Server Connect The Common Configuration Client Over Local IP & MAC Address. .

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,
http://www.syncswitch.com/

check the site . we need same solution.

waiting for your update.

regards,
Md. Abu Noman



Taidot: Asterisk PBX, Linux, VoIP, Windows Server

Näytä lisää: low bandwidth voip, voip low bandwidth, low bandwidth voip client, low bandwith voip, low bandwidth voip calls, asterisk voip client end server, asterisk sbo solution, sbo module asterisk, asterisk optimization, asterisk setup low bandwidth, low bandwidth solution asterisk, low bandwidth bandwidth voip, asterisk bandwidth optimization, low bandwidth asterisk, voip account low bandwith, sbo voip solution, low bandwidth voip solution, asterisk sbo, client client low bandwidth voip, sbo solution, voip solution low bandwidth, voip web server, solution n, md web, bandwidth com

Tietoa työnantajasta:
( 0 arvostelua ) Kulalaumpr, Bangladesh

Projektin tunnus: #1706124

6 freelanceria on tarjonnut keskimäärin 2333 $ tähän työhön

meral

hi. can do central server and rpm for outer servers(centos 6 setup required). bid not include website work, dialler [url removed, login to view] web for control remove servers and setup saving trunk. i work in voip fro 9 years and i am b Lisää

5000 $ USD 10 päivässä
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7.8
shakoush2001

Hi I am a CISSP,RHCE,CCNA,MCSA,Linux+ and a CEH. I do have 7+ years experience in System Administration . I have experience in a high availability environment with 100+ servers and more than 500 000+ subscribers, I kno Lisää

1500 $ USD 20 päivässä
(55 arvostelua)
6.2
ekoandriprasetyo

Ready to work.

750 $ USD 3 päivässä
(2 arvostelua)
3.1
K5Sg8h7OE

Custom software development - <b><i>Removed by Admin</i></b>

750 $ USD 1 päivässä
(0 arvostelua)
0.0
masterasterisk

Dear Hiring Manager, I have an Expertise in Voice Broadcast, VPS, Trixbox, FlashPBX, FreePBX, GoAutoDial, Asterisk, Elastix, IVR, A2billing with experience in migrating applications to the cloud, and I'm very intere Lisää

1500 $ USD 21 päivässä
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0.0
mohitshrm334

Hi, I have more than 7 tears of experience of doing the similar tasks, I can do this asap. Regards Anil

4500 $ USD 15 päivässä
(0 arvostelua)
1.8