asterisk users - Nov 2003 - ISDN debugging and SIP dial-in issue

Hi,

my setup is quite simple: an asterix CVS of 2003-11-15 on a
2.4.21-debian-5 GNU/Linux box in an internal network (192.168.1.0/24,
asterisk is 192.168.1.10).

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

   - with an ISDN AVM c4 i4l card on an ISDN connection with 2 channels.

I try to:

   - dial-in from ISDN, then transfer to the SIP phone: works very well.

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

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

the very good Asterisk basic demos (echo, IAX) work very well.

Details:
   - ISDN dial-out:
    -- Executing Dial("Modem[i4l]/ttyI0",
"Modem/g1:079xxxxxxx|60|r") in
       new stack
       DEBUG[15376]: File app_dial.c, Line 392 (dial_exec): SIMPLE DIAL (NO
       URL)
    -- Called g1:079xxxxxxx

     (xxx are from me)

   - 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:

        DEBUG[5126]: File chan_sip.c, Line 3369 (build_route): build_route:
        Contact hop: <sip:17476691152@192.168.1.190>
        -- Executing Playback("SIP/17476691152-7158",
        "extbusy|skip") in new stack
        -- Timeout on SIP/17476691152-7158
        == CDR updated on SIP/17476691152-7158
        -- Executing Hangup("SIP/17476691152-7158", "") in
new stack
        == Spawn extension (localphones, t, 1) exited non-zero on
           'SIP/17476691152-7158'
        DEBUG[15376]: File chan_sip.c, Line 1068 (sip_hangup):
        find_user(17476691152) - decrement inUse counter
        DEBUG[5126]: File chan_sip.c, Line 565 (__sip_ack): Stopping
        retransmission on
'75057cca-9ad7-2fdf-39af-8774a2a01abf@192.168.1.190'
        of Response 33505: Found

My specific questions are:
   - 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)

   - what should I do for this SIP dial-in issue ?  Specifically how can
     I debug this ?

Thank you very much for any ideas/pointers.

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

Hi Marc,
>    - with a SIP phone configured as 192.168.1.190, and with its SIP
>      server being 192.168.1.190
> 
>    - with an ISDN AVM c4 i4l card on an ISDN connection with 2 channels.
[..]
>    - ISDN dial-out:
>     -- Executing Dial("Modem[i4l]/ttyI0",
"Modem/g1:079xxxxxxx|60|r") in
>        new stack
>        DEBUG[15376]: File app_dial.c, Line 392 (dial_exec): SIMPLE DIAL (NO
>        URL)
>     -- Called g1:079xxxxxxx

What is your reason to use i4l instead of the chan_capi driver
(http://www.junghanns.net/asterisk/)? Did you try both, and found i4l
perform better?

I have the same card as you do, and am not very satisfied by the sound
quality.
--
Best regards

Peer Oliver Schmidt
the internet company