[asterisk-users] SIP trunk with > 250 lines

[Danny Nicholas]

Here are a few "look outs";  Using conference rooms will increase your
bandwidth requirements.  On board Network controllers will affect
performance in this "high-use" scenario.  250 simultaneous calls will use
about 7.5Mb of bandwidth depending on the codec(s) you use.

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Cary Fitch
Sent: Tuesday, March 24, 2009 10:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP trunk with > 250 lines

 

First Issue to be addressed is how many simultaneous calls and bandwidth
availability.

 

Number of "lines" (numbers) is not a limitation in it self unless they are
all in use.

 

Cary Fitch

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Christian
Victor
Sent: Tuesday, March 24, 2009 10:10 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] SIP trunk with > 250 lines

 

Hi!

A customer of mine wants to connect an asterisk system with 240 to 480 lines
to a PSTN switch. To save the costs for E1 cards and the corresponding E1
mainlines he wants to connect the system to the switch by a SIP trunk.

Phones will be connected to the server through the same SIP trunk as this
will be some kind of a "hosted pbx".

Given he finds a provider wich has this much SIP capacity and IP bandwith
and no codec conversion is needed - do you think this is possible with pure
asterisk on a decent system? Is there anything I shoudl watch out for?

Your help is much appreciated!

Chris

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090324/43c21da5/attachment.htm