[Asterisk-Users] Simple gateway SIP <--> ISDN

[Daniel Gonzalez]

Hi *,

I have a very simple setup, since this is my first test with asterisk: I have configured an
Asterisk server and a kphone client (SIP) to talk to each other. Right now, the SIP user gets
authenticated by asterisk without problems. My goal is to redirect the call to a given ISDN
telephone number.

Here are the parameters that I want to use for my setup:

- SIP user: test_sip_user
- destination ISDN telephone number: 123456
- asterisk server:
        192.168.1.100
        ISDN interface
        extension associated to the ISDN number: 100
- kphone client:
        192.168.1.200
        sound system (ALSA with OSS emulation, working)


The first problem that I have is that, even though kphone and asterisk are able to authenticate
the user, I am not sure that sound gets transmitted.

This is the first thing that I would like to achieve: to verify that sound is flowing between
kphone and asterisk. The easiest thing would be to get a dial tone in the kphone client, but I
fear that this is not possible, since SIP initiates a session with all needed parameters, and does
not need/accept a dial tone. Please, correct me (and tell me how to do it :) ) if I am wrong on
this one.

The next method to verify the flow of sound, easy enough for me to try, would be to set up a
single mailbox, with a greeting message and the possibility to record speech on the mailbox. This
should allow me to verify the flow of sound if both directions. Could you provide any hints on how
to do this? Just a very simple setup is needed.

Once I have verified that sound is flowing, I would like to make the call into the ISDN network. I
have some questions:
1) Is it actually possible to implement this scenario? I have understood that asterisk can work as
a gateway between SIP and ISDN (and between other networks, too). Is this correct?
2) I am not able to figure out what extension to use for the SIP user. The kphone sends the
following request to asterisk:

  sip:100 at 192.168.1.100:5060

I do not know how to use this in an extension specification in order to get asterisk to dial the
desired number (123456) via the ISDN interface. I have tried to setup extension 100 to playback a
sound file, like this:

exten => 100,1,Wait(1)
exten => 100,2,Playback(demo-congrats)
exten => 100,3,Hangup

but kphone complains that the session can not be established. What extension specification should
I use to match the SIP call?


And I have an aside question: kphone can (apparently) also be used for video-conferences. Is this
in any way supported by asterisk? My impression is that asterisk only provides voice services.


Thanks for your help,

Daniel Gonzalez