[Asterisk-Users] How to configure Asterisk as a VoIP(SIP) to 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 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 to know what kind of 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 it as a starting point configuring my system.

Any help will be really appreciated.

Thanks in advance,

 

Alejandro.

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