[Asterisk-Users] hierarchical VoIP system

trixter aka Bret McDanel trixter at 0xdecafbad.com
Wed Nov 30 10:59:17 MST 2005


On Wed, 2005-11-30 at 17:45 +0000, Joao Pereira wrote:
> Hello
> Im managing a WAN with a lot of Universities. Some of them already 
> installed a VoIP solution based on SER (to manage SIP clients) and 
> Asterisk (for services and PSTN GW). The DNS routing provided by SER is 
> working perfectly, but we want to start routing all calls thru IP 
> transparently.
> We want our legacy PBXs (that are connected to Asterisk) to forward all 
> calls to IP. The idea is to forward all calls to a central VoIP server, 
> that has all the numbers that already are VoIP enabled, and then:
> - if the called number is VoIP enabled, he routes the call to that Univ. 
> VoIP server
> - if the called number isnt in the list, the call goes back to the PBX 
> and a PSTN call is dialed
> 

Have you considered enum for the voip enabled phones and failing through
to either realtime or extensions.conf if enum fails?  

tip I found enum is easier to manage with powerdns and the mysql backend
(although it can do postgress, isc bind, and other stuff for its
backend, it seems faster for me and many have reported a much lower
memory footprint when doing thousands of zones).

That would seem to accomplish what you want and make it easier to port
people over to voip as needed.  Infact depending on how you configure
everything, everyone could be in enum even the old legacy routes, then
its a simple matter of editing what is already there.  At least that has
been my experience.


> Now, the top of the hierarchy should be an Asterisk or SER? I dont know 
> which of the systems is the best choice for the job. Does someone has an 
> idea of what should we use?
> 

SER tends to deal with large numbers of sip registrations better than
asterisk on the same hardware.  Mostly because it is specifically
written for just that task.  realtime may change that (I havent seen any
specific studies done on load issues post realtime so I cant comment as
I havent done any personally).


-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 189 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051130/dafbaf5c/attachment.pgp


More information about the asterisk-users mailing list