ast2*CLI> sip debug SIP Debugging enabled Audio is at 70.85.7.246 port 12292 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 4.79.212.236:5060: INVITE sip:+17575125254@4.79.212.236 SIP/2.0 Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK4e51d182;rport From: "0008110000" ;tag=as45e71aaa To: Contact: Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 102 INVITE User-Agent: Asterisk Max-Forwards: 70 Date: Tue, 06 Jun 2006 22:28:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 226 v=0 o=root 26224 26224 IN IP4 70.85.7.246 s=session c=IN IP4 70.85.7.246 t=0 0 m=audio 12292 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- ast2*CLI> <-- SIP read from 4.79.212.236:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK4e51d182;rport=5060 From: "0008110000" ;tag=as45e71aaa To: Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 102 INVITE Server: Bandwidth.com SDE (v1.0) Content-Length: 0 Warning: 392 4.79.212.234:5060 "Noisy feedback tells: pid=15272 req_src_ip=4.79.212.236 req_src_port=5060 in_uri=sip:+17575125254@4.79.212.236 out_uri=sip:+17575125254@4.68.250.148:5060 via_cnt==2" --- (9 headers 0 lines)--- ast2*CLI> <-- SIP read from 4.79.212.236:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK4e51d182;rport=5060 Record-Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 184 v=0 o=- 1149632903 1149632904 IN IP4 209.244.42.237 s=- c=IN IP4 209.244.42.237 t=0 0 m=audio 60494 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (10 headers 9 lines)--- Found RTP audio format 0 Found RTP audio format 101 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264)/video=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.244.42.237:60494 ast2*CLI> <-- SIP read from 4.79.212.236:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK4e51d182;rport=5060 Record-Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 184 v=0 o=- 1149632903 1149632904 IN IP4 209.244.42.237 s=- c=IN IP4 209.244.42.237 t=0 0 m=audio 60494 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (10 headers 9 lines)--- Found RTP audio format 0 Found RTP audio format 101 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264)/video=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.244.42.237:60494 list_route: hop: list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 4.79.212.236, port 5060 Transmitting (no NAT) to 4.79.212.236:5060: ACK sip:+17575125254@4.68.250.148:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK40e0165a;rport Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Contact: Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 102 ACK User-Agent: Asterisk Max-Forwards: 70 Content-Length: 0 --- > Channel SIP/level3-e5bc was answered. Audio is at 70.85.7.246 port 19944 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 4.79.212.236:5060: INVITE sip:+17572181730@4.79.212.236 SIP/2.0 Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK50c629d2;rport From: "0008110000" ;tag=as2acc804c To: Contact: Call-ID: 1a4bb33c64bb8e6912b179a10bc5ccc1@70.85.7.246 CSeq: 102 INVITE User-Agent: Asterisk Max-Forwards: 70 Date: Tue, 06 Jun 2006 22:28:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 232 v=0 o=root 26224 26224 IN IP4 209.244.42.237 s=session c=IN IP4 209.244.42.237 t=0 0 m=audio 60494 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Called Level3/+17572181730 ast2*CLI> <-- SIP read from 4.79.212.236:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK50c629d2;rport=5060 From: "0008110000" ;tag=as2acc804c To: Call-ID: 1a4bb33c64bb8e6912b179a10bc5ccc1@70.85.7.246 CSeq: 102 INVITE Server: Bandwidth.com SDE (v1.0) Content-Length: 0 Warning: 392 4.79.212.234:5060 "Noisy feedback tells: pid=15263 req_src_ip=4.79.212.236 req_src_port=5060 in_uri=sip:+17572181730@4.79.212.236 out_uri=sip:+17572181730@4.68.250.148:5060 via_cnt==2" --- (9 headers 0 lines)--- ast2*CLI> <-- SIP read from 4.79.212.236:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK50c629d2;rport=5060 Record-Route: ,, From: "0008110000" ;tag=as2acc804c To: ;tag=VPST506071629460 Call-ID: 1a4bb33c64bb8e6912b179a10bc5ccc1@70.85.7.246 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 182 v=0 o=- 1149632918 1149632919 IN IP4 209.247.5.148 s=- c=IN IP4 209.247.5.148 t=0 0 m=audio 60290 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (10 headers 9 lines)--- Found RTP audio format 0 Found RTP audio format 101 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264)/video=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.247.5.148:60290 -- SIP/Level3-1117 is making progress passing it to SIP/level3-e5bc set_destination: Parsing for address/port to send to set_destination: set destination to 4.79.212.236, port 5060 Audio is at 70.85.7.246 port 12292 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 4.79.212.236:5060: INVITE sip:+17575125254@4.68.250.148:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK274d1f93;rport Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Contact: Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 103 INVITE User-Agent: Asterisk 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: 230 v=0 o=root 26224 26225 IN IP4 209.247.5.148 s=session c=IN IP4 209.247.5.148 t=0 0 m=audio 60290 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- ast2*CLI> <-- SIP read from 4.79.212.236:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK274d1f93;rport=5060 From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 103 INVITE Server: Bandwidth.com SDE (v1.0) Content-Length: 0 Warning: 392 4.79.212.234:5060 "Noisy feedback tells: pid=15267 req_src_ip=4.79.212.236 req_src_port=5060 in_uri=sip:+17575125254@4.68.250.148:5060;transport=udp out_uri=sip:+17575125254@4.68.250.148:5060;transport=udp via_cnt==2" --- (9 headers 0 lines)--- ast2*CLI> <-- SIP read from 4.79.212.236:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK274d1f93;rport=5060 Record-Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 103 INVITE Contact: Content-Type: application/sdp Content-Length: 184 v=0 o=- 1149632903 1149632905 IN IP4 209.244.42.237 s=- c=IN IP4 209.244.42.237 t=0 0 m=audio 60494 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (10 headers 9 lines)--- Found RTP audio format 0 Found RTP audio format 101 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264)/video=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.244.42.237:60494 set_destination: Parsing for address/port to send to set_destination: set destination to 4.79.212.236, port 5060 Transmitting (no NAT) to 4.79.212.236:5060: ACK sip:+17575125254@4.68.250.148:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK4d8dbc39;rport Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Contact: Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 103 ACK User-Agent: Asterisk Max-Forwards: 70 Content-Length: 0 --- -------------- Starting another call to transfer to ---------------------- <-- SIP read from 4.79.212.236:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK50c629d2;rport=5060 Record-Route: ,, From: "0008110000" ;tag=as2acc804c To: ;tag=VPST506071629460 Call-ID: 1a4bb33c64bb8e6912b179a10bc5ccc1@70.85.7.246 CSeq: 102 INVITE Contact: Content-Type: application/sdp Content-Length: 182 v=0 o=- 1149632918 1149632919 IN IP4 209.247.5.148 s=- c=IN IP4 209.247.5.148 t=0 0 m=audio 60290 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (10 headers 9 lines)--- Found RTP audio format 0 Found RTP audio format 101 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264)/video=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.247.5.148:60290 list_route: hop: list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 4.79.212.236, port 5060 Transmitting (no NAT) to 4.79.212.236:5060: ACK sip:+17572181730@4.68.250.148:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK3d06a81b;rport Route: ,, From: "0008110000" ;tag=as2acc804c To: ;tag=VPST506071629460 Contact: Call-ID: 1a4bb33c64bb8e6912b179a10bc5ccc1@70.85.7.246 CSeq: 102 ACK User-Agent: Asterisk Max-Forwards: 70 Content-Length: 0 --- -- SIP/Level3-1117 answered SIP/level3-e5bc -- Native bridging SIP/level3-e5bc and SIP/Level3-1117 ast2*CLI> <-- SIP read from 4.79.212.236:5060: BYE sip:0008110000@70.85.7.246 SIP/2.0 Record-Route: Record-Route: Record-Route: Via: SIP/2.0/UDP 4.79.212.236;branch=0 Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bK4643.e87bad3.0 Via: SIP/2.0/UDP 4.79.212.236;branch=0 Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1149573416481 From: ;tag=VPST506071629460 To: "0008110000" ;tag=as2acc804c Call-ID: 1a4bb33c64bb8e6912b179a10bc5ccc1@70.85.7.246 CSeq: 1 BYE Contact: Max-Forwards: 66 Content-Length: 0 --- (15 headers 0 lines)--- Sending to 4.79.212.236 : 5060 (no NAT) Transmitting (no NAT) to 4.79.212.236:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 4.79.212.236;branch=0;received=4.79.212.236 Via: SIP/2.0/UDP 4.79.212.234;branch=z9hG4bK4643.e87bad3.0 Via: SIP/2.0/UDP 4.79.212.236;branch=0 Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1149573416481 Record-Route: Record-Route: Record-Route: From: ;tag=VPST506071629460 To: "0008110000" ;tag=as2acc804c Call-ID: 1a4bb33c64bb8e6912b179a10bc5ccc1@70.85.7.246 CSeq: 1 BYE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 4.79.212.236, port 5060 Audio is at 70.85.7.246 port 12292 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 4.79.212.236:5060: INVITE sip:+17575125254@4.68.250.148:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK21eb0ae3;rport Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Contact: Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 104 INVITE User-Agent: Asterisk 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: 226 v=0 o=root 26224 26226 IN IP4 70.85.7.246 s=session c=IN IP4 70.85.7.246 t=0 0 m=audio 12292 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- == Spawn extension (custom-notify, 0, 1) exited non-zero on 'SIP/level3-e5bc' ast2*CLI> <-- SIP read from 4.79.212.236:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK21eb0ae3;rport=5060 From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 104 INVITE Server: Bandwidth.com SDE (v1.0) Content-Length: 0 Warning: 392 4.79.212.234:5060 "Noisy feedback tells: pid=15272 req_src_ip=4.79.212.236 req_src_port=5060 in_uri=sip:+17575125254@4.68.250.148:5060;transport=udp out_uri=sip:+17575125254@4.68.250.148:5060;transport=udp via_cnt==2" --- (9 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to 4.79.212.236, port 5060 Reliably Transmitting (no NAT) to 4.79.212.236:5060: BYE sip:+17575125254@4.68.250.148:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK22ac14b2;rport Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Contact: Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 105 BYE User-Agent: Asterisk Max-Forwards: 70 Content-Length: 0 --- Really destroying SIP dialog '1a4bb33c64bb8e6912b179a10bc5ccc1@70.85.7.246' Method: BYE ast2*CLI> <-- SIP read from 4.79.212.236:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK21eb0ae3;rport=5060 Record-Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 104 INVITE Contact: Content-Type: application/sdp Content-Length: 184 v=0 o=- 1149632903 1149632906 IN IP4 209.244.42.237 s=- c=IN IP4 209.244.42.237 t=0 0 m=audio 60494 RTP/AVP 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --- (10 headers 9 lines)--- Found RTP audio format 0 Found RTP audio format 101 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264)/video=0x3d07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|h261|h263|h263p|h264), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.244.42.237:60494 set_destination: Parsing for address/port to send to set_destination: set destination to 4.79.212.236, port 5060 Transmitting (no NAT) to 4.79.212.236:5060: ACK sip:+17575125254@4.68.250.148:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK06aff61f;rport Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Contact: Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 104 ACK User-Agent: Asterisk Max-Forwards: 70 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 4.79.212.236, port 5060 Reliably Transmitting (no NAT) to 4.79.212.236:5060: BYE sip:+17575125254@4.68.250.148:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK02eb0f87;rport Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Contact: Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 106 BYE User-Agent: Asterisk Max-Forwards: 70 Content-Length: 0 --- ast2*CLI> <-- SIP read from 4.79.212.236:5060: SIP/2.0 481 Call Leg Does Not Exist Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK22ac14b2;rport=5060 Record-Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 105 BYE Contact: Content-Length: 0 --- (9 headers 0 lines)--- Jun 6 18:28:52 WARNING[26248]: chan_sip.c:10948 handle_response: Remote host can't match request BYE to call '33bba34f2798bb494efa52a6155ae938@70.85.7.246'. Giving up. ast2*CLI> <-- SIP read from 4.79.212.236:5060: SIP/2.0 481 Call Leg Does Not Exist Via: SIP/2.0/UDP 70.85.7.246:5060;branch=z9hG4bK02eb0f87;rport=5060 Record-Route: ,, From: "0008110000" ;tag=as45e71aaa To: ;tag=VPST506071629460 Call-ID: 33bba34f2798bb494efa52a6155ae938@70.85.7.246 CSeq: 106 BYE Contact: Content-Length: 0 --- (9 headers 0 lines)--- Jun 6 18:28:52 WARNING[26248]: chan_sip.c:10948 handle_response: Remote host can't match request BYE to call '33bba34f2798bb494efa52a6155ae938@70.85.7.246'. Giving up. Really destroying SIP dialog '33bba34f2798bb494efa52a6155ae938@70.85.7.246' Method: INVITE ast2*CLI>