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:[EMAIL PROTECTED]>
-- 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 '[EMAIL PROTECTED]'
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.
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