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

[Marc SCHAEFER]

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 at 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 at 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.