[Asterisk-Users] ISDN debugging and SIP dial-in issue

[Philipp von Klitzing]

Hi!

>    - with a SIP phone configured as 192.168.1.190, and with its SIP
>      server being 192.168.1.190

That doesn't look right. Do you have another "SIP server" installed on 
your client machine - shouldn't that rather be *, or did you - which I 
guess - just mistype the IP? Which SIP phone are you using 
(hardware/software, brand, version)?

>    - dial-in from ISDN, then transfer to ISDN on the secondary channel:
>      doesn't work (more details below)

I assume with "transfer" you mean that you are trying to "dial out" on 
the 2nd channel. So who are you trying to call? If you are trying to call 
yourself then you'd need three channels, and you'll get a "busy" signal 
since you only have two channels...
If that is not it: Check your context setup: The incoming call must be  
in a context that is allowed to dial out again.

>    - dial anything from the SIP phone: doesn't work (more details below)

Please provide (the relevant parts of) your extensions.conf.

>    - SIP dial in: it seems the session is initiated (SIP message from
>      Asterisk on the ethernet), and then UDP (voice?) packets are sent,
>      but no answer comes from the SIP phone and after a moment Asterisk
>      fails with:

- check rtp.conf
- any firewall (personal firewall?) or NAT in between SIP client and 
Asterisk?
- maybe also check the RTP port setup in your SIP client
- show us your sip.conf

>    - how can I see the i4l chatting dialing-out to be sure what the problem
>      is (could be a wrong MSN for example, or Asterisk interpreting the
>      0 prefix)

Not sure, but: You might want to look into the isdn4linux documentation 
and use its tools like isdnlog (?) etc.

Cheers, Philipp