<--- SIP read from 192.168.3.199:5060 ---> INVITE sip:7572731028@192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.199;branch=z9hG4bK5d1f14d04 Max-Forwards: 70 Content-Length: 226 To: 7572731028 ;tag=as745befed From: "Chuck <22>" ;tag=f3f1a497846d1ed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 651528776 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: "Chuck <22>" Supported: replaces User-Agent: Aastra 480i Cordless/1.4.1.1077 Brcm Callctrl/1.5 MxSF/v3.2.6.26 v=0 o=MxSIP 0 1890023598 IN IP4 192.168.3.199 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 3000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> --- (15 headers 11 lines) --- Sending to 192.168.3.199 : 5060 (no NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:3000 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 0.0.0.0:3000 Audio is at 192.168.3.1 port 13406 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.3.199:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.199;branch=z9hG4bK5d1f14d04;received=192.168.3.199 From: "Chuck <22>" ;tag=f3f1a497846d1ed To: 7572731028 ;tag=as745befed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 651528776 INVITE User-Agent: Asterisk PBX SVN-trunk-r60763 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 234 v=0 o=root 718 720 IN IP4 192.168.3.1 s=session c=IN IP4 192.168.3.1 t=0 0 m=audio 13406 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 <------------> -- Started music on hold, class 'default', on SIP/teliax-sip-09aa8590 <--- SIP read from 192.168.3.199:5060 ---> ACK sip:7572731028@192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.199;branch=z9hG4bK7a48d409a Max-Forwards: 70 Content-Length: 0 To: 7572731028 ;tag=as745befed From: "Chuck <22>" ;tag=f3f1a497846d1ed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 651528776 ACK Contact: "Chuck <22>" User-Agent: Aastra 480i Cordless/1.4.1.1077 Brcm Callctrl/1.5 MxSF/v3.2.6.26 <-------------> --- (10 headers 0 lines) --- == Manager 'admin' logged off from 127.0.0.1 <--- SIP read from 216.89.79.2:5060 ---> OPTIONS sip:s@192.168.2.2 SIP/2.0 Via: SIP/2.0/UDP 216.89.79.2:5060;branch=z9hG4bK1bea5df9;rport From: "Unknown" ;tag=as7ebb8638 To: Contact: Call-ID: 3c69093813dc5a8467ba9a4b0be7e721@216.89.79.2 CSeq: 102 OPTIONS User-Agent: "TELIAX GEAR v2.1.0x" Max-Forwards: 70 Date: Sun, 08 Apr 2007 22:44:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Looking for s in from-sip-external (domain 192.168.2.2) <--- Transmitting (no NAT) to 216.89.79.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 216.89.79.2:5060;branch=z9hG4bK1bea5df9;received=216.89.79.2;rport=5060 From: "Unknown" ;tag=as7ebb8638 To: ;tag=as5825f0b1 Call-ID: 3c69093813dc5a8467ba9a4b0be7e721@216.89.79.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r60763 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '3c69093813dc5a8467ba9a4b0be7e721@216.89.79.2' in 32000 ms (Method: OPTIONS) <--- SIP read from 192.168.3.199:5060 ---> INVITE sip:7572731028@192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.199;branch=z9hG4bK817b199e0 Max-Forwards: 70 Content-Length: 232 To: 7572731028 ;tag=as745befed From: "Chuck <22>" ;tag=f3f1a497846d1ed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 651528777 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: "Chuck <22>" Supported: replaces User-Agent: Aastra 480i Cordless/1.4.1.1077 Brcm Callctrl/1.5 MxSF/v3.2.6.26 v=0 o=MxSIP 0 1890023599 IN IP4 192.168.3.199 s=SIP Call c=IN IP4 192.168.3.199 t=0 0 m=audio 3000 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - <-------------> --- (15 headers 11 lines) --- Sending to 192.168.3.199 : 5060 (no NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.3.199:3000 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 192.168.3.199:3000 Audio is at 192.168.3.1 port 13406 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.3.199:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.199;branch=z9hG4bK817b199e0;received=192.168.3.199 From: "Chuck <22>" ;tag=f3f1a497846d1ed To: 7572731028 ;tag=as745befed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 651528777 INVITE User-Agent: Asterisk PBX SVN-trunk-r60763 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 234 v=0 o=root 718 721 IN IP4 192.168.3.1 s=session c=IN IP4 192.168.3.1 t=0 0 m=audio 13406 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 <------------> <--- SIP read from 192.168.3.199:5060 ---> ACK sip:7572731028@192.168.3.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.199;branch=z9hG4bKa6d9ea2d3 Max-Forwards: 70 Content-Length: 0 To: 7572731028 ;tag=as745befed From: "Chuck <22>" ;tag=f3f1a497846d1ed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 651528777 ACK Contact: "Chuck <22>" User-Agent: Aastra 480i Cordless/1.4.1.1077 Brcm Callctrl/1.5 MxSF/v3.2.6.26 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '2b2539155228ffc57e759a0c110f7efe@127.0.0.1' Method: REGISTER Really destroying SIP dialog '3c69093813dc5a8467ba9a4b0be7e721@216.89.79.2' Method: OPTIONS == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 Reliably Transmitting (no NAT) to 192.168.3.199:5060: OPTIONS sip:22@192.168.3.199 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK5b9a211e Max-Forwards: 70 From: "Unknown" ;tag=as3e7d08f2 To: Contact: Call-ID: 5d0ff9bf314c9aa2087ef333028b58c8@192.168.3.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r60763 Date: Sun, 08 Apr 2007 22:42:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.3.199:5060 ---> SIP/2.0 200 OK Call-ID: 5d0ff9bf314c9aa2087ef333028b58c8@192.168.3.1 CSeq: 102 OPTIONS From: "Unknown" ;tag=as3e7d08f2 To: ;tag=77b236fd567cc2f Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK5b9a211e Content-Length: 0 Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Contact: Supported: replaces User-Agent: Aastra 480i Cordless/1.4.1.1077 Brcm Callctrl/1.5 MxSF/v3.2.6.26 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '1d98c0a44390308d7541611471c16afe@192.168.3.1' Method: OPTIONS == Manager 'admin' logged off from 127.0.0.1 <--- SIP read from 216.89.79.2:5060 ---> BYE sip:17178987911@192.168.2.2 SIP/2.0 Via: SIP/2.0/UDP 216.89.79.2:5060;branch=z9hG4bK0e868902;rport From: ;tag=as3f74d892 To: "CT Inc" ;tag=as22da0d03 Contact: Call-ID: 4509b38933cff80a0bb9c09e4aced962@192.168.2.2 CSeq: 102 BYE User-Agent: "TELIAX GEAR v2.1.0x" Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 216.89.79.2 : 5060 (NAT) <--- Transmitting (NAT) to 216.89.79.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 216.89.79.2:5060;branch=z9hG4bK0e868902;received=216.89.79.2;rport=5060 From: ;tag=as3f74d892 To: "CT Inc" ;tag=as22da0d03 Call-ID: 4509b38933cff80a0bb9c09e4aced962@192.168.2.2 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r60763 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Stopped music on hold on SIP/teliax-sip-09aa8590 == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/22-09aa4000' Scheduling destruction of SIP dialog 'a96aa79c4d369cd42c074669d6744f74@192.168.3.199' in 32000 ms (Method: ACK) set_destination: Parsing <22> for address/port to send to Reliably Transmitting (no NAT) to 192.168.3.199:5060: BYE 22 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK56a0aaf6 Max-Forwards: 70 From: 7572731028 ;tag=as745befed To: "Chuck <22>" ;tag=f3f1a497846d1ed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r60763 Content-Length: 0 --- Huh? Child handler, but nobody there? Really destroying SIP dialog '4509b38933cff80a0bb9c09e4aced962@192.168.2.2' Method: BYE Retransmitting #1 (no NAT) to 192.168.3.199:5060: BYE 22 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK56a0aaf6 Max-Forwards: 70 From: 7572731028 ;tag=as745befed To: "Chuck <22>" ;tag=f3f1a497846d1ed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r60763 Content-Length: 0 --- Retransmitting #2 (no NAT) to 192.168.3.199:5060: BYE 22 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK56a0aaf6 Max-Forwards: 70 From: 7572731028 ;tag=as745befed To: "Chuck <22>" ;tag=f3f1a497846d1ed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r60763 Content-Length: 0 --- *CLI> Retransmitting #3 (no NAT) to 192.168.3.199:5060: BYE 22 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK56a0aaf6 Max-Forwards: 70 From: 7572731028 ;tag=as745befed To: "Chuck <22>" ;tag=f3f1a497846d1ed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r60763 Content-Length: 0 --- <--- SIP read from 216.89.79.2:5060 ---> OPTIONS sip:s@192.168.2.2 SIP/2.0 Via: SIP/2.0/UDP 216.89.79.2:5060;branch=z9hG4bK4145724e;rport From: "Unknown" ;tag=as23c20848 To: Contact: Call-ID: 2548a481442824462765929f7d7d18ed@216.89.79.2 CSeq: 102 OPTIONS User-Agent: "TELIAX GEAR v2.1.0x" Max-Forwards: 70 Date: Sun, 08 Apr 2007 22:45:06 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Looking for s in from-sip-external (domain 192.168.2.2) <--- Transmitting (no NAT) to 216.89.79.2:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 216.89.79.2:5060;branch=z9hG4bK4145724e;received=216.89.79.2;rport=5060 From: "Unknown" ;tag=as23c20848 To: ;tag=as1b15a87c Call-ID: 2548a481442824462765929f7d7d18ed@216.89.79.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r60763 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Accept: application/sdp Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2548a481442824462765929f7d7d18ed@216.89.79.2' in 32000 ms (Method: OPTIONS) Retransmitting #4 (no NAT) to 192.168.3.199:5060: BYE 22 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK56a0aaf6 Max-Forwards: 70 From: 7572731028 ;tag=as745befed To: "Chuck <22>" ;tag=f3f1a497846d1ed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r60763 Content-Length: 0 --- Retransmitting #5 (no NAT) to 192.168.3.199:5060: BYE 22 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.1:5060;branch=z9hG4bK56a0aaf6 Max-Forwards: 70 From: 7572731028 ;tag=as745befed To: "Chuck <22>" ;tag=f3f1a497846d1ed Call-ID: a96aa79c4d369cd42c074669d6744f74@192.168.3.199 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-trunk-r60763 Content-Length: 0 --- Waiting for inactivity to perform halt Waiting for inactivity to perform halt... Executing last minute cleanups == Destroying musiconhold processes Asterisk cleanly ending (0).