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@192.168.10.2' in 32000 ms Reliably Transmitting (no NAT) to 80.239.235.200:5060: CANCEL sip:+4970712976872@sip.voipstunt.com SIP/2.0 Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK2079ec6b;rport From: "Arik" <sip:arikfunke1@192.168.10.2>;tag=as7a95fade To: <sip:+4970712976872@sip.voipstunt.com> Destroying call '5a497edb4781951240e675014577faa1@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@192.168.10.2>;tag=as439face3 To: <sip:sip.voipstunt.com> Contact: <sip:asterisk@192.168.10.2> Call-ID: 48a6cf253c3390fc290eeeaf0c360c4f@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@sip.voipstunt.com> Contact: sip:+4970712976872@80.239.235.200:5060 Call-ID: 5a497edb4781951240e675014577faa1@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@192.168.10.2>;tag=as439face3 To: <sip:sip.voipstunt.com> Contact: sip:80.239.235.200:5060 Call-ID: 48a6cf253c3390fc290eeeaf0c360c4f@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: