[Spce-user] multi teenat

Jon Bonilla (Manwe) jbonilla at sipwise.com
Tue Nov 6 08:41:29 EST 2012

El Mon, 5 Nov 2012 01:37:45 +0600
Tanjil Ahmed <mail at tanjil.net> escribió:

> hello
> im newbie .. i need help with build up one system if anyone help me ill b
> glad
> main goal to minimize the BW in client side with good quality of voice .
> We need some kind of bandwidth compression system ( upto 60-80% than usual
> SIP calls )from Server A to Server B.
> Server A = Asterisk server
> Server B = Asterisk Client server
> Explanation of scenario:
> 1. server A ( asterisk server, with static IP) receiving VoIP calls , with
> sip protocol, using G711,G729 and/or G723.1 codec and sending calls to
> Server B
> 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls
> from server A and sending to gateways (quintum gateway for example) or E1
> cards.
> 3. Number of Server B can be unlimited.
> 4. Number of Gateways/E1 cards per server B can be unlimited
> 5. For server B installation need easy to use ISO image that could be
> booted from USB flash drive, and those USB flash drive will be delivered to
> our Server B type client (ther termination provider)
> A. Any mini Linux distribution exam- puppy Linux , linux mint
> B. Fedora desktop distribution
> C. Centos 5.8 or 6
> 7. Server A to Server B voice traffic will be encrypted so that voice port
> blocked bandwidth can be used for termination. we will used .
> A. iax trunks in trunking mode.
> B. Open vpn static mode and dynamic mode
> C. Tnic static and dynamic mode
> 8. Asterisk web billing gui for adding gateways.
> Adding client , Prefix , dialing plan viewing active calls, billing cdr
> ,etc.
> we will provide you the Dedicated server asterisk and client asterisk
> configure IAX trunking, so we can measure the BW compression making the
> SIP-> IAX call trunking, need develop a simple WEB tool to change IAX IP
> and port (you understand that it is sensitive option when trunk is blocked
> by country border GW)
> continue building up main server with codec conversion (will install
> g729/g723 codecs) amd Install OpenVPN Server&client - at this stage we will
> test it and measure the BW compression with all kinds of options like
> codecs and openvpn compression modes;
> continue project with compiling the automated installation distribution
> (with sipwise OpenVPN, Asterisk,freeswitch  Codec conversion, IAX trunks
> config ) for client-side CentOS system, which can be distributed to may
> servers.
> continue working on project by building up WEB interface for main server
> adding Billing, and other options from Item 2 like adding GW, adding
> client, adding IAX trunks

Do you have any questions or is this a project request? I'd say you should
contact sales at sipwise.com for such requests.

Anyways, I'm not pretty sure if you really know what Sipwise products are. Our
products are not based on Asterisk software. 



More information about the Spce-user mailing list