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
'[EMAIL PROTECTED]'
in 32000 ms
Reliably Transmitting (no NAT) to 80.239.235.200:5060:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK2079ec6b;rport
From: "Arik" <sip:[EMAIL PROTECTED]>;tag=as7a95fade
To: <sip:[EMAIL PROTECTED]>
Destroying call '[EMAIL PROTECTED]'
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:[EMAIL PROTECTED]>;tag=as439face3
To: <sip:sip.voipstunt.com>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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: [EMAIL PROTECTED]>
Contact: sip:[EMAIL PROTECTED]:5060
Call-ID: [EMAIL PROTECTED]
---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:[EMAIL PROTECTED]>;tag=as439face3
To: <sip:sip.voipstunt.com>
Contact: sip:80.239.235.200:5060
Call-ID: [EMAIL PROTECTED]
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:
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