<------------> -- Native bridging SIP/10-081d0338 and SIP/20-081caad8 set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.76.69, port 20952 Audio is at 10.0.76.248 port 16076 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.76.69:20952: INVITE sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c SIP/2.0 Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK75d3ceb0;rport From: "10" ;tag=as2c89c807 To: ;tag=6b72c313 Contact: Call-ID: 2384eb1945bf6b1f77c07de73b35a260@10.0.76.248 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 232 v=0 o=root 3725 3726 IN IP4 10.0.0.62 s=session c=IN IP4 10.0.0.62 t=0 0 m=audio 11248 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- astserv1*CLI> <--- SIP read from 10.0.76.69:20952 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK75d3ceb0;rport=5060 Contact: To: ;tag=6b72c313 From: "10";tag=as2c89c807 Call-ID: 2384eb1945bf6b1f77c07de73b35a260@10.0.76.248 CSeq: 103 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 178 v=0 o=- 0 2 IN IP4 10.0.76.69 s=CounterPath X-Lite 3.0 c=IN IP4 10.0.76.69 t=0 0 m=audio 6788 RTP/AVP 8 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv <-------------> --- (11 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.0.76.69:6788 Found description format telephone-event for ID 101 Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.0.76.69:6788 set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.76.69, port 20952 Transmitting (no NAT) to 10.0.76.69:20952: ACK sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c SIP/2.0 Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK604fdb65;rport From: "10" ;tag=as2c89c807 To: ;tag=6b72c313 Contact: Call-ID: 2384eb1945bf6b1f77c07de73b35a260@10.0.76.248 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- astserv1*CLI> <--- SIP read from 10.0.0.62:42964 ---> ACK sip:20@10.0.76.248 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.62:42964;branch=z9hG4bK-d87543-f4771065a06ace7d-1--d87543-;rport Max-Forwards: 70 Contact: To: "20 (Softphone)";tag=as5c96f43a From: ;tag=c3042d00 Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg. CSeq: 2 ACK Proxy-Authorization: Digest username="10",realm="asterisk",nonce="3424cabd",uri="sip:20@10.0.76.248",response="934fece6dd50b62ce9d66b3ec73034a1",algorithm=MD5 User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.62, port 42964 Audio is at 10.0.76.248 port 11760 Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.0.0.62:42964: INVITE sip:10@10.0.0.62:42964 SIP/2.0 Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK7c87e802;rport From: "20 (Softphone)";tag=as5c96f43a To: ;tag=c3042d00 Contact: Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg. CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 197 v=0 o=root 3725 3727 IN IP4 10.0.76.69 s=session c=IN IP4 10.0.76.69 t=0 0 m=audio 6788 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- astserv1*CLI> <--- SIP read from 10.0.0.62:42964 ---> SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK7c87e802;rport=5060 To: ;tag=c3042d00 From: "20 (Softphone)";tag=as5c96f43a Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg. CSeq: 102 INVITE User-Agent: X-Lite release 1011s stamp 41150 Warning: 305 devnull "SDP: Incompatible media format: no common codec." Content-Length: 0 <-------------> --- (9 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.62, port 42964 Transmitting (no NAT) to 10.0.0.62:42964: ACK sip:10@10.0.0.62:42964 SIP/2.0 Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK7c87e802;rport From: "20 (Softphone)";tag=as5c96f43a To: ;tag=c3042d00 Contact: Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg. CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '2384eb1945bf6b1f77c07de73b35a260@10.0.76.248' in 6528 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.76.69, port 20952 Reliably Transmitting (no NAT) to 10.0.76.69:20952: BYE sip:20@10.0.76.69:20952;rinstance=99e1c7336e6e471c SIP/2.0 Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK3502836a;rport From: "10" ;tag=as2c89c807 To: ;tag=6b72c313 Call-ID: 2384eb1945bf6b1f77c07de73b35a260@10.0.76.248 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- == Spawn extension (htl3r, 20, 1) exited non-zero on 'SIP/10-081d0338' Scheduling destruction of SIP dialog 'YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg.' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 10.0.0.62, port 42964 Reliably Transmitting (no NAT) to 10.0.0.62:42964: BYE sip:10@10.0.0.62:42964 SIP/2.0 Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK4b379144;rport From: "20 (Softphone)";tag=as5c96f43a To: ;tag=c3042d00 Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg. CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- astserv1*CLI> <--- SIP read from 10.0.76.69:20952 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK3502836a;rport=5060 Contact: To: ;tag=6b72c313 From: "10";tag=as2c89c807 Call-ID: 2384eb1945bf6b1f77c07de73b35a260@10.0.76.248 CSeq: 104 BYE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '2384eb1945bf6b1f77c07de73b35a260@10.0.76.248' Method: INVITE astserv1*CLI> <--- SIP read from 10.0.0.62:42964 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.76.248:5060;branch=z9hG4bK4b379144;rport=5060 Contact: To: ;tag=c3042d00 From: "20 (Softphone)";tag=as5c96f43a Call-ID: YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg. CSeq: 103 BYE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog 'YWQwMDY3Y2JlYWJiM2EzNmRkMDJmYzIwODIzMDRjOTg.' Method: ACK