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
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