[asterisk-users] chan_isdn / chan_sip problems

[Arik Raffael Funke]

Hi,

I am using Asterisk 1.2 with internal isdn phones connected via a hfcpci 
card in nt-mode with misdn. Bridging calls from the internal hfcpci via 
a avmfritz card (also chan_misdn) to the PSTN works flawlessly. However 
when I use a sip channel to route the outgoing call via voipstunt, it 
always rings three times and then gives me a busy indication. With my 
previous configuration, asterisk 1.0.10, zaphfc, chan_capi-cm this was 
no problem.

I thought it was a sip problem and used "sip debug" but at the moment 
when the ringing switches to busy no debug messages appear. I also tried 
a softphone - it works fine with the same config. So I figure that it 
has something to do with the chan_misdn to chan_sip bridging.

Below it the chan_misdn debug trace from the console at the moment when 
the switch from ringing to busy occurs. Does this tell anybody something 
that might help with my problem? Do I have a mistake in my misdn 
configuration?

Thanks in advance for any hints.

Best regards,
Arik

------------ console debug trace -------------
hestia*CLI>
hestia*CLI>
hestia*CLI>
P[ 1] *I IND :TIMEOUT oad:23 dad:070712976872 pid:21 state:DIALING
  P[ 1]  --> state: DIALING
  P[ 1]  I SEND:DISCONNECT oad:23 dad:070712976872 pid:21
  P[ 1]   --> bc_state:BCHAN_ACTIVATED
P[ 1] *ec_disable
  P[ 1]  I IND :RELEASE oad: dad: pid:21 state:DIALING
  P[ 1]  hangup_chan
  P[ 1]  -> queue_hangup
  P[ 1]  release_chan: bc with l3id: 10042
  P[ 1]  * RELEASING CHANNEL pid:21 ctx:macro-tsblcr dad:sip oad:23 
state: DIALIN
G
  P[ 1]  I SEND:RELEASE_COMPLETE oad: dad: pid:21
  P[ 1]   --> bc_state:BCHAN_CLEANED
  Scheduling destruction of call 
'5a497edb4781951240e675014577faa1 at 192.168.10.2'
in 32000 ms
  Reliably Transmitting (no NAT) to 80.239.235.200:5060:
CANCEL sip:+4970712976872 at sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK2079ec6b;rport
From: "Arik" <sip:arikfunke1 at 192.168.10.2>;tag=as7a95fade
To: <sip:+4970712976872 at sip.voipstunt.com>
Destroying call '5a497edb4781951240e675014577faa1 at 192.168.10.2'
12 headers, 0 lines
CReliably Transmitting (no NAT) to 80.239.235.200:5060:
OPTIONS sip:sip.voipstunt.com SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK65df4ea2;rport
From: "asterisk" <sip:asterisk at 192.168.10.2>;tag=as439face3
To: <sip:sip.voipstunt.com>
Contact: <sip:asterisk at 192.168.10.2>
Call-ID: 48a6cf253c3390fc290eeeaf0c360c4f at 192.168.10.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX.235.200:5060:
Max-Forwards: 70
Date: Fri, 22 Sep 2006 15:50:43 GMTbranch=z9hG4bK2079ec6b;rport
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0976872 at sip.voipstunt.com>
Contact: sip:+4970712976872 at 80.239.235.200:5060
Call-ID: 5a497edb4781951240e675014577faa1 at 192.168.10.2
---q: 102 CANCEL
hestia*CLI>
<-- SIP read from 80.239.235.200:5060: PTIONS
SIP/2.0 200 Ok: 0
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK65df4ea2;rport
From: "asterisk" <sip:asterisk at 192.168.10.2>;tag=as439face3
To: <sip:sip.voipstunt.com>
Contact: sip:80.239.235.200:5060
Call-ID: 48a6cf253c3390fc290eeeaf0c360c4f at 192.168.10.2
CSeq: 102 OPTIONS
Supported:
User-Agent: (Very nice Sip Registrar Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS
Accept: application/sdp
Accept-Encoding:
Accept-Language: