[Asterisk-Users] Need configuration sample for VoIP(SIP) -> PSTN Gateway

[Alejandro Sosa]

Hello,

 

I'm very new with * and I would really appreciate some help to implement a SIP to PSTN Gateway.

My current scenario includes an * box with a TE405P board. I have a 1.5Mb connection to the outside world (using a router with firewall capabilities) and channel banks that allow me to connect the T1s coming out of the TE405 board to the PSTN network (carrier).

I need to configure * to accept calls coming over IP (SIP) and terminate them thru the Zaptel interface on the PSTN network.

Also need which parameters the SIP peer needs to know to connect to my system, other than my IP and the port (ie: compression, codecs, etc.) and where/how to configure those settings on my * box.

I have some understanding on how the configurations files work in *, what I really need is some sample implementation that works for what I described, to use them as a starting point for configuring my system.

Any help will be really appreciated.

Thanks in advance,

 

Alejandro.

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