[Asterisk-Users] sip to pstn gateway
Fabian Müller
mueller.fab at gmx.net
Sun Oct 17 10:33:28 MST 2004
Hello,
here is what I am able to do:
- I am able to register a SIP Phone on an Asterisk server.
- I am able to call an extension on the remote Asterisk server with my SIP
phone and hear the congratulation message.
Informations about the configuration:
- There is no phonecard (no digium card, no isdn card ...) in the
Asterisk box but only a network interface card which connects the
Asterisk server to a softswitch. This softswitch is the gateway to PSTN.
Unfortunately I do not know anything about the softswitch. Is there
something important that I should know about it?
Here is what I would like to have:
- I would like to be able to call a PSTN/ISDN phone with my SIP phone.
- That means when I take my SIP Phone and dial a telephone number that
belongs to the PSTN Asterisk must route the SIP packages to the
softswitch which in turn routes the call to the PSTN.
- When I dial another registered SIP phone Asterisk should connect the
two sip phones so that they can speak to each other.
------------- ------------ -------------- --------
| SIP phone | ---> | Asterisk | ---> | Softswitch | ---> | PSTN |
------------- ------------ -------------- --------
|
| -------------
|----> | Sip phone |
-------------
I have no idea how to configure Asterisk to accomplish this task. I
started reading documents like ftp://ftp.isi.edu/in-notes/rfc3372.txt
and the sip RFC and Mailing List articles and so on but they did not
make me able to configure Asterisk in that way.
Does anybody know where I can find documents that describe how I can
do what I would like to have? Do I have to configure a SIP Proxy (SER
for example) on the Asterisk box or does it work without a SIP proxy
as well? Do I have to register the Asterisk box on the softswitch?
(Should this be possible at all?)
Thanks very much in advance for any kind of help.
Regards,
Fabian Müller
More information about the asterisk-users
mailing list