[Asterisk-Users] PSTN gateway implementation?

[Sebastian Nocetti]

you can do that, in my experiencie, using oh323 I could not handle more than
30 active calls, doing g729 passthru...
 
I dont know how to do IP limitation.... for restrict ip access use
iptables....
 
I did basic dialpeers like this:
 
exten -> 1305.,1,dial(OH323/xxxx)
exten -> 1305.,2,congestion
 
I am right?

  _____  

De: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] En nombre de Alejandro Sosa
Enviado el: Lunes, 19 de Julio de 2004 03:25 p.m.
Para: asterisk-users at lists.digium.com
Asunto: [Asterisk-Users] PSTN gateway implementation?



Hello,

 

I need help in creating a simple PSTN Gateway. This is the scenario:

 

-I have one client sending me VoIP traffic (they don't have asterisk, so IAX
is out of the picture for me) and I need to validate that traffic (only
accept calls coming from his IP). After that I would terminate the calls to
the PSTN network and keep logs for billing purposes.

 

-I have a TE405P board and only one T1 worth of phone lines (24) connected
to it using an Adtran TA750 channel bank.

 

Is Asterisk capable of handling multiple incoming VoIP calls arriving from
the same source (IP) or do I need to get something else to take the incoming
traffic and pass it on to Asterisk? (I've read about using SER as a SIP
proxy, but it's not clear to me wheather I need it or not). Can I use the
OpenH.323 module to take care of the incoming VoIP traffic?

 

I am totally new to all this. Any help will be really appreciated.

 

Please give me your input on which is the best implementation of a
VoIP->PSTN gateway and the some sample configuration files for the
plattforms involved (Asterisk, SER, OpenH323, etc.)

 

Thanks in advance.

 

 

Alejandro Sosa.

 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040719/49661657/attachment.htm