UDP transport (answer call, then h323 endpoint hangs up, BYE is correctly send to sip endpoint): -- SIP/ipbx-gw-08281258 answered H323/ip$192.168.40.7:61762/694 bill*CLI> bill*CLI> bill*CLI> Scheduling destruction of SIP dialog '388cd2d40a700cb603cc5e42470c73be@192.168.40.4' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.38.20, port 5060 Reliably Transmitting (no NAT) to 192.168.38.20:5060: BYE sip:324@192.168.38.20 SIP/2.0 Via: SIP/2.0/UDP 192.168.40.4:5060;branch=z9hG4bK4ca7624f;rport Max-Forwards: 70 From: "Pavel Jezek" ;tag=as42fa06d2 To: ;tag=as58a3c80c Call-ID: 388cd2d40a700cb603cc5e42470c73be@192.168.40.4 CSeq: 104 BYE User-Agent: Asterisk PBX SVN-trunk-r104031M Authorization: *************** Content-Length: 0 --- == Spawn extension (from-ccm, *324, 2) exited non-zero on 'H323/ip$192.168.40.7:61762/694' bill*CLI> <--- SIP read from UDP://192.168.38.20:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.40.4:5060;branch=z9hG4bK4ca7624f;received=192.168.36.134;rport=60720 From: "Pavel Jezek" ;tag=as42fa06d2 To: ;tag=as58a3c80c Call-ID: 388cd2d40a700cb603cc5e42470c73be@192.168.40.4 CSeq: 104 BYE User-Agent: Asterisk PBX SVN-trunk-r105677M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '388cd2d40a700cb603cc5e42470c73be@192.168.40.4' Method: INVITE TLS transport (answer call, then h323 endpoint hangs up, BYE is NOT send to sip endpoint at all): -- SIP/ipbx-gw-08257d80 answered H323/ip$192.168.40.7:61947/696 bill*CLI> bill*CLI> Scheduling destruction of SIP dialog '128146fe24c3510d7e2708a86a69d9ae@192.168.40.4' in 6400 ms (Method: INVITE) == Spawn extension (from-ccm, *324, 2) exited non-zero on 'H323/ip$192.168.40.7:61947/696'