Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- 12 headers, 0 lines Reliably Transmitting (NAT) to 220.238.243.11:5060: OPTIONS sip:6720@10.1.1.3:5060;line=gv8x1x75 SIP/2.0 Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK777cf286;rport From: "Unknown" ;tag=as6f7ad0c6 To: Contact: Call-ID: 2ea4724857b0e9ee0af63f64546275d6@209.212.101.56 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 10 Jun 2007 04:16:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- pabx*CLI> <-- SIP read from 220.238.243.11:5060: ACK sip:0860121218@209.212.101.56 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-piqzjmmtfmte;rport From: "6720" ;tag=eix4lx9av6 To: ;tag=as7e67fa4c Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- Retransmitting #2 (NAT) to 220.238.243.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nqtgfff2ftkz;received=220.238.243.11;rport=5060 From: "6720" ;tag=eix4lx9av6 To: ;tag=as7e67fa4c Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 22348 22348 IN IP4 209.212.101.56 s=session c=IN IP4 209.212.101.56 t=0 0 m=audio 10612 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pabx*CLI> <-- SIP read from 220.238.243.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK777cf286;rport=5060 From: "Unknown" ;tag=as6f7ad0c6 To: Call-ID: 2ea4724857b0e9ee0af63f64546275d6@209.212.101.56 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom320/6.5.8 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Length: 0 --- (14 headers 0 lines) --- Destroying call '2ea4724857b0e9ee0af63f64546275d6@209.212.101.56' pabx*CLI> <-- SIP read from 220.238.243.11:5060: ACK sip:0860121218@209.212.101.56 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-piqzjmmtfmte;rport From: "6720" ;tag=eix4lx9av6 To: ;tag=as7e67fa4c Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- Destroying call '683888ab6adac4b97a5dacb36e73bc0d@127.0.0.1' Retransmitting #3 (NAT) to 220.238.243.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nqtgfff2ftkz;received=220.238.243.11;rport=5060 From: "6720" ;tag=eix4lx9av6 To: ;tag=as7e67fa4c Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 22348 22348 IN IP4 209.212.101.56 s=session c=IN IP4 209.212.101.56 t=0 0 m=audio 10612 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pabx*CLI> <-- SIP read from 220.238.243.11:5060: ACK sip:0860121218@209.212.101.56 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-piqzjmmtfmte;rport From: "6720" ;tag=eix4lx9av6 To: ;tag=as7e67fa4c Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- -- Playing 'auth-incorrect' (language 'en') Retransmitting #4 (NAT) to 220.238.243.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nqtgfff2ftkz;received=220.238.243.11;rport=5060 From: "6720" ;tag=eix4lx9av6 To: ;tag=as7e67fa4c Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 22348 22348 IN IP4 209.212.101.56 s=session c=IN IP4 209.212.101.56 t=0 0 m=audio 10612 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pabx*CLI> <-- SIP read from 220.238.243.11:5060: ACK sip:0860121218@209.212.101.56 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-piqzjmmtfmte;rport From: "6720" ;tag=eix4lx9av6 To: ;tag=as7e67fa4c Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- 12 headers, 0 lines Reliably Transmitting (NAT) to 41.243.80.35:5060: OPTIONS sip:5021@192.168.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK610df032;rport From: "Unknown" ;tag=as35b091a1 To: Contact: Call-ID: 297eee0836f9e908313ab1f471a99316@209.212.101.56 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 10 Jun 2007 04:16:16 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- pabx*CLI> <-- SIP read from 41.243.80.35:5060: SIP/2.0 200 OK To: ;tag=a9378c97a509c43fi0 From: "Unknown" ;tag=as35b091a1 Call-ID: 297eee0836f9e908313ab1f471a99316@209.212.101.56 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK610df032 Server: Linksys/SPA3102-3.2.6(GWa) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura --- (10 headers 0 lines) --- Destroying call '297eee0836f9e908313ab1f471a99316@209.212.101.56' -- Playing 'auth-thankyou' (language 'en') Retransmitting #5 (NAT) to 220.238.243.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nqtgfff2ftkz;received=220.238.243.11;rport=5060 From: "6720" ;tag=eix4lx9av6 To: ;tag=as7e67fa4c Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 22348 22348 IN IP4 209.212.101.56 s=session c=IN IP4 209.212.101.56 t=0 0 m=audio 10612 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- 12 headers, 0 lines Reliably Transmitting (NAT) to 76.172.54.233:2059: OPTIONS sip:5030@192.168.1.101:2059;line=t0c378t2 SIP/2.0 Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK398bfb24;rport From: "Unknown" ;tag=as0003ecd1 To: Contact: Call-ID: 353729526c40e4103a1e564068b08ade@209.212.101.56 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 10 Jun 2007 04:16:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- -- Executing Macro("SIP/6720-0892aa10", "dialout-trunk|2|0860121218||") in new stack -- Executing Set("SIP/6720-0892aa10", "DIAL_TRUNK=2") in new stack -- Executing Set("SIP/6720-0892aa10", "_NODEST=") in new stack -- Executing Set("SIP/6720-0892aa10", "DIAL_NUMBER=0860121218") in new stack -- Executing Set("SIP/6720-0892aa10", "ROUTE_PASSWD=") in new stack -- Executing Set("SIP/6720-0892aa10", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "1?noauth") in new stack -- Goto (macro-dialout-trunk,s,8) -- Executing Set("SIP/6720-0892aa10", "GROUP()=OUT_2") in new stack -- Executing Macro("SIP/6720-0892aa10", "user-callerid|SKIPTTL") in new stack -- Executing NoOp("SIP/6720-0892aa10", "user-callerid: device 6720") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "0?report") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "0?start") in new stack -- Executing Set("SIP/6720-0892aa10", "REALCALLERIDNUM=6720") in new stack -- Executing NoOp("SIP/6720-0892aa10", "REALCALLERIDNUM is 6720") in new stack -- Executing Set("SIP/6720-0892aa10", "AMPUSER=6720") in new stack -- Executing Set("SIP/6720-0892aa10", "AMPUSERCIDNAME=Mark Slatem") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "0?report") in new stack -- Executing Set("SIP/6720-0892aa10", "CALLERID(all)=Mark Slatem <6720>") in new stack -- Executing Set("SIP/6720-0892aa10", "REALCALLERIDNUM=6720") in new stack -- Executing NoOp("SIP/6720-0892aa10", "TTL: ARG1: SKIPTTL") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "1?continue") in new stack -- Goto (macro-user-callerid,s,21) -- Executing NoOp("SIP/6720-0892aa10", "Using CallerID "Mark Slatem" <6720>") in new stack -- Executing Macro("SIP/6720-0892aa10", "record-enable|6720|OUT") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing DeadAGI("SIP/6720-0892aa10", "recordingcheck|20070610-061618|1181448966.4004") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20070610-061618|1181448966.4004: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/6720-0892aa10", "No recording needed") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "0?skipoutcid") in new stack -- Executing Set("SIP/6720-0892aa10", "DIAL_TRUNK_OPTIONS=") in new stack -- Executing Macro("SIP/6720-0892aa10", "outbound-callerid|2") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "1?start") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp("SIP/6720-0892aa10", "REALCALLERIDNUM is 6720") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,9) -- Executing Set("SIP/6720-0892aa10", "USEROUTCID=0113406720") in new stack -- Executing Set("SIP/6720-0892aa10", "EMERGENCYCID=") in new stack -- Executing Set("SIP/6720-0892aa10", "TRUNKOUTCID=<0113406700>") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,16) -- Executing GotoIf("SIP/6720-0892aa10", "0?usercid") in new stack -- Executing Set("SIP/6720-0892aa10", "CALLERID(all)=<0113406700>") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "0?report") in new stack -- Executing Set("SIP/6720-0892aa10", "CALLERID(all)=0113406720") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "1?report:hidecid") in new stack -- Goto (macro-outbound-callerid,s,22) -- Executing NoOp("SIP/6720-0892aa10", "CallerID set to "" <0113406720>") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "0?nomax") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "0?chanfull") in new stack -- Executing DeadAGI("SIP/6720-0892aa10", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix pabx*CLI> <-- SIP read from 220.238.243.11:5060: ACK sip:0860121218@209.212.101.56 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-piqzjmmtfmte;rport From: "6720" ;tag=eix4lx9av6 To: ;tag=as7e67fa4c Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- pabx*CLI> <-- SIP read from 76.172.54.233:2059: SIP/2.0 200 OK Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK398bfb24;rport=5060 From: "Unknown" ;tag=as0003ecd1 To: Call-ID: 353729526c40e4103a1e564068b08ade@209.212.101.56 CSeq: 102 OPTIONS Contact: ;flow-id=1 User-Agent: snom300/6.5.8 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Length: 0 --- (14 headers 0 lines) --- Destroying call '353729526c40e4103a1e564068b08ade@209.212.101.56' -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/6720-0892aa10", "OUTNUM=0860121218") in new stack -- Executing Set("SIP/6720-0892aa10", "custom=AMP") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "1?customtrunk") in new stack -- Goto (macro-dialout-trunk,s,22) -- Executing Set("SIP/6720-0892aa10", "pre_num=AMP:mISDN/g:out/") in new stack -- Executing Set("SIP/6720-0892aa10", "the_num=OUTNUM") in new stack -- Executing Set("SIP/6720-0892aa10", "post_num=") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "1?outnum:skipoutnum") in new stack -- Goto (macro-dialout-trunk,s,26) -- Executing Set("SIP/6720-0892aa10", "the_num=0860121218") in new stack -- Executing Dial("SIP/6720-0892aa10", "mISDN/g:out/0860121218|300|") in new stack -- Called g:out/0860121218 -- mISDN/1-u1210 is proceeding passing it to SIP/6720-0892aa10 -- mISDN/1-u1210 is ringing -- mISDN/1-u1210 answered SIP/6720-0892aa10 Retransmitting #6 (NAT) to 220.238.243.11:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nqtgfff2ftkz;received=220.238.243.11;rport=5060 From: "6720" ;tag=eix4lx9av6 To: ;tag=as7e67fa4c Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 243 v=0 o=root 22348 22348 IN IP4 209.212.101.56 s=session c=IN IP4 209.212.101.56 t=0 0 m=audio 10612 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- pabx*CLI> <-- SIP read from 220.238.243.11:5060: ACK sip:0860121218@209.212.101.56 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-piqzjmmtfmte;rport From: "6720" ;tag=eix4lx9av6 To: ;tag=as7e67fa4c Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- == Spawn extension (macro-dialout-trunk, s, 27) exited non-zero on 'SIP/6720-0892aa10' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 27) exited non-zero on 'SIP/6720-0892aa10' -- Executing Macro("SIP/6720-0892aa10", "hangupcall") in new stack -- Executing ResetCDR("SIP/6720-0892aa10", "w") in new stack -- Executing NoCDR("SIP/6720-0892aa10", "") in new stack -- Executing GotoIf("SIP/6720-0892aa10", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf("SIP/6720-0892aa10", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing Wait("SIP/6720-0892aa10", "5") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/6720-0892aa10' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/6720-0892aa10' 12 headers, 0 lines Reliably Transmitting (no NAT) to 213.137.73.140:5060: OPTIONS sip:natrelay.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK317c7428;rport From: "Unknown" ;tag=as440b7b7b To: Contact: Call-ID: 2e1860ef28e581f50eae63472924df48@209.212.101.56 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 10 Jun 2007 04:16:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- Destroying call '3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316' pabx*CLI> <-- SIP read from 213.137.73.140:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK317c7428;rport To: From: "Unknown" ;tag=as440b7b7b Call-ID: 2e1860ef28e581f50eae63472924df48@209.212.101.56 CSeq: 102 OPTIONS Content-Length: 0 --- (7 headers 0 lines) --- <-- SIP read from 213.137.73.140:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK317c7428;rport To: From: "Unknown" ;tag=as440b7b7b Call-ID: 2e1860ef28e581f50eae63472924df48@209.212.101.56 CSeq: 102 OPTIONS Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '2e1860ef28e581f50eae63472924df48@209.212.101.56' 12 headers, 0 lines Reliably Transmitting (no NAT) to 213.137.73.140:5060: OPTIONS sip:natrelay.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK5bbde63e;rport From: "Unknown" ;tag=as3dde12f1 To: Contact: Call-ID: 76ee36d01c0a95652c27d4612c67e418@209.212.101.56 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 10 Jun 2007 04:16:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- pabx*CLI> <-- SIP read from 213.137.73.140:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK5bbde63e;rport To: From: "Unknown" ;tag=as3dde12f1 Call-ID: 76ee36d01c0a95652c27d4612c67e418@209.212.101.56 CSeq: 102 OPTIONS Content-Length: 0 --- (7 headers 0 lines) --- <-- SIP read from 213.137.73.140:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK5bbde63e;rport To: From: "Unknown" ;tag=as3dde12f1 Call-ID: 76ee36d01c0a95652c27d4612c67e418@209.212.101.56 CSeq: 102 OPTIONS Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '76ee36d01c0a95652c27d4612c67e418@209.212.101.56'