<------------> Scheduling destruction of SIP dialog '3d1ec90657e91ed92cba7f1a6d9805d0@192.168.11.5' in 32000 ms (Method: OPTIONS) csgtacsip1*CLI> Really destroying SIP dialog 'deb2b9aa1ee926c3' Method: REGISTER csgtacsip1*CLI> <--- SIP read from UDP://216.82.224.202:5060 ---> SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 70.0.0.0:5060;received=70.0.0.0;branch=z9hG4bK749ebea7;rport=5060Record-Route: From: "asterisk" ;tag=as1f2b1b2aTo: ;tag=VPST50603522629853Call-ID: 7ff93f6b53d7215938dca9a555419ee5@70.0.0.0CSeq: 102 INVITEContact: Content-Type: application/sdpContent-Length: 182v=0o=- 1259534667 1259534668 IN IP4 209.247.23.75s=-c=IN IP4 209.247.23.75t=0 0m=audio 60792 RTP/AVP 0 101a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20 <-------------> --- (10 headers 9 lines) --- csgtacsip1*CLI> Found RTP audio format 0 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), 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 209.247.23.75:60792 csgtacsip1*CLI> -- SIP/bandwidth-primary-00000005 is making progress passing it to SIP/1594-00000004 csgtacsip1*CLI> Audio is at 70.0.0.0 port 10566 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP csgtacsip1*CLI> <--- Transmitting (NAT) to 71.231.179.213:5060 ---> SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 192.168.19.7:5060;branch=z9hG4bK476a03165;received=71.231.179.213From: SIP PHONE 1594 ;tag=bb3af7df42047acTo: 18002662278 ;tag=as1450dd76Call-ID: a509e1614e60ef2079955eca1d42ad5e@192.168.19.7CSeq: 1123593381 INVITEServer: Asterisk PBX SVN-branch-1.6.1-r231301MAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerRequire: timerSession-Expires: 1800;refresher=uasContact: Content-Type: application/sdpContent-Length: 279v=0o=root 145027767 145027767 IN IP4 70.0.0.0s=Asterisk PBX SVN-branch-1.6.1- r231301Mc=IN IP4 70.0.0.0t=0 0m=audio 10566 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - - a=ptime:20a=sendrecv <------------> Scheduling destruction of SIP dialog '3d1ec90657e91ed92cba7f1a6d9805d0@192.168.11.5' in 32000 ms (Method: OPTIONS) csgtacsip1*CLI> Really destroying SIP dialog '0b9a45ab488da597' Method: REGISTER csgtacsip1*CLI> Really destroying SIP dialog '241d7013d523dca6' Method: REGISTER csgtacsip1*CLI> Really destroying SIP dialog '4994fc0e27ec5196' Method: REGISTER csgtacsip1*CLI> <--- SIP read from UDP://216.82.224.202:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 70.0.0.0:5060;received=70.0.0.0;branch=z9hG4bK749ebea7;rport=5060Record-Route: From: "asterisk" ;tag=as1f2b1b2aTo: ;tag=VPST50603522629853Call-ID: 7ff93f6b53d7215938dca9a555419ee5@70.0.0.0CSeq: 102 INVITEContact: Content-Type: application/sdpContent-Length: 182v=0o=- 1259534667 1259534668 IN IP4 209.247.23.75s=-c=IN IP4 209.247.23.75t=0 0m=audio 60792 RTP/AVP 0 101a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20 <-------------> --- (10 headers 9 lines) --- csgtacsip1*CLI> list_route: hop: csgtacsip1*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 216.82.224.202, port 5060 Transmitting (no NAT) to 216.82.224.202:5060: ACK sip:+18002662278@209.247.16.221:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 70.0.0.0:5060;branch=z9hG4bK7771d70e;rportRoute: Max-Forwards: 70From: "asterisk" ;tag=as1f2b1b2aTo: ;tag=VPST50603522629853Contact: Call-ID: 7ff93f6b53d7215938dca9a555419ee5@70.0.0.0CSeq: 102 ACKUser- Agent: Asterisk PBX SVN-branch-1.6.1-r231301MContent-Length: 0 --- csgtacsip1*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 216.82.224.202, port 5060 csgtacsip1*CLI> Reliably Transmitting (no NAT) to 216.82.224.202:5060: BYE sip:+18002662278@209.247.16.221:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 70.0.0.0:5060;branch=z9hG4bK5646936a;rportRoute: Max-Forwards: 70From: "asterisk" ;tag=as1f2b1b2aTo: ;tag=VPST50603522629853Call-ID: 7ff93f6b53d7215938dca9a555419ee5@70.0.0.0CSeq: 103 BYEUser-Agent: Asterisk PBX SVN-branch- 1.6.1-r231301MX-Asterisk-HangupCause: UnknownX-Asterisk-HangupCauseCode: 0Content-Length: 0 --- csgtacsip1*CLI> Scheduling destruction of SIP dialog '7ff93f6b53d7215938dca9a555419ee5@70.0.0.0' in 6400 ms (Method: INVITE) csgtacsip1*CLI> -- SIP/bandwidth-primary-00000005 answered SIP/1594-00000004 csgtacsip1*CLI> Audio is at 70.0.0.0 port 10566 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP csgtacsip1*CLI> <--- Reliably Transmitting (NAT) to 71.231.179.213:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 192.168.19.7:5060;branch=z9hG4bK476a03165;received=71.231.179.213From: SIP PHONE 1594 ;tag=bb3af7df42047acTo: 18002662278 ;tag=as1450dd76Call-ID: a509e1614e60ef2079955eca1d42ad5e@192.168.19.7CSeq: 1123593381 INVITEServer: Asterisk PBX SVN-branch-1.6.1-r231301MAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerRequire: timerSession-Expires: 1800;refresher=uasContact: Content-Type: application/sdpContent-Length: 279v=0o=root 145027767 145027768 IN IP4 70.0.0.0s=Asterisk PBX SVN-branch-1.6.1- r231301Mc=IN IP4 70.0.0.0t=0 0m=audio 10566 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - - a=ptime:20a=sendrecv <------------> csgtacsip1*CLI> -- Packet2Packet bridging SIP/1594-00000004 and SIP/bandwidth-primary-00000005 csgtacsip1*CLI> <--- SIP read from UDP://216.82.224.202:5060 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 70.0.0.0:5060;received=70.0.0.0;branch=z9hG4bK5646936a;rport=5060Record-Route: From: "asterisk" ;tag=as1f2b1b2aTo: ;tag=VPST50603522629853Call-ID: 7ff93f6b53d7215938dca9a555419ee5@70.0.0.0CSeq: 103 BYEContact: Content-Length: 0 <-------------> csgtacsip1*CLI> --- (9 headers 0 lines) --- csgtacsip1*CLI> <--- SIP read from UDP://71.231.179.213:5060 ---> ACK sip:18002662278@70.0.0.0 SIP/2.0Via: SIP/2.0/UDP 192.168.19.7:5060;branch=z9hG4bK7a505abfdMax-Forwards: 70Content-Length: 0To: 18002662278 ;tag=as1450dd76From: SIP PHONE 1594 ;tag=bb3af7df42047acCall-ID: a509e1614e60ef2079955eca1d42ad5e@192.168.19.7CSeq: 1123593381 ACKContact: SIP PHONE 1594 Authorization:Digest response="838771c6caef4571441f59b0bb310420",username="1594",realm="asterisk",nonce="2c5abd08",algorithm=MD5,uri="sip:18002662278@rtel.csgopenline.com:50 60"User-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (10 headers 0 lines) --- csgtacsip1*CLI> Really destroying SIP dialog '5b9c776928ee194a662776c47567e960@70.0.0.0' Method: OPTIONS csgtacsip1*CLI> Really destroying SIP dialog '74f869570e06361054cc09ba0072e4ae@192.168.11.5' Method: OPTIONS Really destroying SIP dialog '5bf884d652ac4a6b4e1e55047cb85041@192.168.11.5' Method: OPTIONS Really destroying SIP dialog '76d7d1731e81c25a06f272d26f1dc445@192.168.11.5' Method: OPTIONS <------------> Scheduling destruction of SIP dialog '36fed57e59e520d1' in 32000 ms (Method: REGISTER) csgtacsip1*CLI> <--- SIP read from UDP://67.148.102.2:47597 ---> SIP/2.0 200 OKVia: SIP/2.0/UDP 70.0.0.0:5060;branch=z9hG4bK7a9a2bc8;rport=5060;received=70.0.0.0From: "asterisk" ;tag=as4df6036eTo: ;tag=661083303Call-ID: 76a77c72417a33602c29628b11892ef1@70.0.0.0CSeq: 102 OPTIONSAllow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFOServer: Aastra 57i/2.5.2.30Supported: gruu, timer, 100rel, replaces, pathContent-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '76a77c72417a33602c29628b11892ef1@70.0.0.0' Method: OPTIONS csgtacsip1*CLI> Really destroying SIP dialog '7ca834523e737050' Method: REGISTER csgtacsip1*CLI> [Nov 29 14:47:34] WARNING[19315]: chan_sip.c:3588 __sip_autodestruct: Autodestruct on dialog '7ff93f6b53d7215938dca9a555419ee5@70.0.0.0' with owner in place (Method: INVITE) csgtacsip1*CLI> Scheduling destruction of SIP dialog '7ff93f6b53d7215938dca9a555419ee5@70.0.0.0' in 6400 ms (Method: INVITE) csgtacsip1*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 216.82.224.202, port 5060 Reliably Transmitting (no NAT) to 216.82.224.202:5060: BYE sip:+18002662278@209.247.16.221:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 70.0.0.0:5060;branch=z9hG4bK10daa319;rportRoute: Max-Forwards: 70From: "asterisk" ;tag=as1f2b1b2aTo: ;tag=VPST50603522629853Call-ID: 7ff93f6b53d7215938dca9a555419ee5@70.0.0.0CSeq: 104 BYEUser-Agent: Asterisk PBX SVN-branch- 1.6.1-r231301MX-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0 --- csgtacsip1*CLI> == Spawn extension (from-staff, 18002662278, 4) exited non-zero on 'SIP/1594-00000004' csgtacsip1*CLI> Scheduling destruction of SIP dialog 'a509e1614e60ef2079955eca1d42ad5e@192.168.19.7' in 7424 ms (Method: ACK) csgtacsip1*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.19.7, port 5060 Reliably Transmitting (NAT) to 71.231.179.213:5060: BYE sip:1594@192.168.19.7:5060;transport=udp SIP/2.0Via: SIP/2.0/UDP 70.0.0.0:5060;branch=z9hG4bK17d7cbf4;rportMax-Forwards: 70From: 18002662278 ;tag=as1450dd76To: SIP PHONE 1594 ;tag=bb3af7df42047acCall-ID: a509e1614e60ef2079955eca1d42ad5e@192.168.19.7CSeq: 102 BYEUser-Agent: Asterisk PBX SVN-branch-1.6.1-r231301MX-Asterisk-HangupCause: Normal ClearingX- Asterisk-HangupCauseCode: 16Content-Length: 0 --- csgtacsip1*CLI> <--- SIP read from UDP://71.231.179.213:5060 ---> SIP/2.0 200 OKCall-ID: a509e1614e60ef2079955eca1d42ad5e@192.168.19.7CSeq: 102 BYEFrom: 18002662278 ;tag=as1450dd76To: SIP PHONE 1594 ;tag=bb3af7df42047acVia: SIP/2.0/UDP 70.0.0.0:5060;branch=z9hG4bK17d7cbf4;rportContent-Length: 0Supported: replacesUser-Agent: Aastra 480i Cordless/1.4.3.23 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 <-------------> --- (9 headers 0 lines) --- csgtacsip1*CLI> SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'a509e1614e60ef2079955eca1d42ad5e@192.168.19.7' Method: ACK csgtacsip1*CLI> <--- SIP read from UDP://216.82.224.202:5060 ---> SIP/2.0 481 Call Leg Does Not ExistVia: SIP/2.0/UDP 70.0.0.0:5060;received=70.0.0.0;branch=z9hG4bK10daa319;rport=5060Record-Route: From: "asterisk" ;tag=as1f2b1b2aTo: ;tag=VPST50603522629853Call-ID: 7ff93f6b53d7215938dca9a555419ee5@70.0.0.0