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 14464 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-yjwbfxkffkcl;rport From: "6720" ;tag=2ps68pkrvk To: ;tag=as12f04922 Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- -- Playing 'auth-thankyou' (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-90ifcmbqip4r;received=220.238.243.11;rport=5060 From: "6720" ;tag=2ps68pkrvk To: ;tag=as12f04922 Call-ID: 3c2a60a8493e-7gnq2vbwskte@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 14464 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-yjwbfxkffkcl;rport From: "6720" ;tag=2ps68pkrvk To: ;tag=as12f04922 Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- -- 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|20070610-063823|1181450294.4006") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck 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=z9hG4bK168163b4;rport From: "Unknown" ;tag=as78862e94 To: Contact: Call-ID: 029e111b269291d24d7f14c06be57e81@209.212.101.56 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 10 Jun 2007 04:38:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- recordingcheck|20070610-063823|1181450294.4006: 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 -- 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 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=z9hG4bK168163b4;rport=5060 From: "Unknown" ;tag=as78862e94 To: Call-ID: 029e111b269291d24d7f14c06be57e81@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 '029e111b269291d24d7f14c06be57e81@209.212.101.56' -- mISDN/1-u1211 is proceeding passing it to SIP/6720-0892aa10 REGISTER 13 headers, 0 lines REGISTER attempt 1 to 16466880880@natrelay.deltathree.com Reliably Transmitting (no NAT) to 213.137.73.140:5060: REGISTER sip:natrelay.deltathree.com SIP/2.0 Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK7cacb561;rport From: ;tag=as3f32fe47 To: Call-ID: 683888ab6adac4b97a5dacb36e73bc0d@127.0.0.1 CSeq: 713 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="30315332", realm="deltathree.com", algorithm=MD5, uri="sip:natrelay.deltathree.com", nonce="b6159d0e60b894a4a677a920870d97c6", response="d9a6928c1f88bd15eb6a6a4806b0a243", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- -- mISDN/1-u1211 is ringing 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=z9hG4bK7cacb561;rport To: From: ;tag=as3f32fe47 Call-ID: 683888ab6adac4b97a5dacb36e73bc0d@127.0.0.1 CSeq: 713 REGISTER Content-Length: 0 --- (7 headers 0 lines) --- pabx*CLI> <-- SIP read from 213.137.73.140:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK7cacb561;rport From: ;tag=as3f32fe47 To: Call-ID: 683888ab6adac4b97a5dacb36e73bc0d@127.0.0.1 CSeq: 713 REGISTER Contact: expires: 120 Content-Length: 0 --- (9 headers 0 lines) --- Scheduling destruction of call '683888ab6adac4b97a5dacb36e73bc0d@127.0.0.1' in 32000 ms 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-90ifcmbqip4r;received=220.238.243.11;rport=5060 From: "6720" ;tag=2ps68pkrvk To: ;tag=as12f04922 Call-ID: 3c2a60a8493e-7gnq2vbwskte@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 14464 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 - - - - --- -- mISDN/1-u1211 answered SIP/6720-0892aa10 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=z9hG4bK7c22c8c4;rport From: "Unknown" ;tag=as018ff021 To: Contact: Call-ID: 5acdcfed62122a0512b0eeb22aa7f00d@209.212.101.56 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 10 Jun 2007 04:38:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 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=z9hG4bK7c22c8c4;rport=5060 From: "Unknown" ;tag=as018ff021 To: Call-ID: 5acdcfed62122a0512b0eeb22aa7f00d@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 '5acdcfed62122a0512b0eeb22aa7f00d@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-yjwbfxkffkcl;rport From: "6720" ;tag=2ps68pkrvk To: ;tag=as12f04922 Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- 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-90ifcmbqip4r;received=220.238.243.11;rport=5060 From: "6720" ;tag=2ps68pkrvk To: ;tag=as12f04922 Call-ID: 3c2a60a8493e-7gnq2vbwskte@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 14464 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-yjwbfxkffkcl;rport From: "6720" ;tag=2ps68pkrvk To: ;tag=as12f04922 Call-ID: 3c2a60a8493e-7gnq2vbwskte@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 (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=z9hG4bK2787c1e2;rport From: "Unknown" ;tag=as6b09822b To: Contact: Call-ID: 20efd9907cf459df0ba60fc12522eb99@209.212.101.56 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 10 Jun 2007 04:38:32 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=z9hG4bK2787c1e2;rport To: From: "Unknown" ;tag=as6b09822b Call-ID: 20efd9907cf459df0ba60fc12522eb99@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=z9hG4bK2787c1e2;rport To: From: "Unknown" ;tag=as6b09822b Call-ID: 20efd9907cf459df0ba60fc12522eb99@209.212.101.56 CSeq: 102 OPTIONS Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '20efd9907cf459df0ba60fc12522eb99@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=z9hG4bK3898dbd1;rport From: "Unknown" ;tag=as17d0b675 To: Contact: Call-ID: 6e23a290783d30cf2246dd2e0dd4164e@209.212.101.56 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 10 Jun 2007 04:38:33 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=z9hG4bK3898dbd1;rport To: From: "Unknown" ;tag=as17d0b675 Call-ID: 6e23a290783d30cf2246dd2e0dd4164e@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=z9hG4bK3898dbd1;rport To: From: "Unknown" ;tag=as17d0b675 Call-ID: 6e23a290783d30cf2246dd2e0dd4164e@209.212.101.56 CSeq: 102 OPTIONS Content-Length: 0 --- (7 headers 0 lines) --- Destroying call '6e23a290783d30cf2246dd2e0dd4164e@209.212.101.56' == 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' Destroying call '3c2a60a8493e-7gnq2vbwskte@snom320-000413242316' pabx*CLI> <-- SIP read from 220.238.243.11:5060: BYE sip:0860121218@209.212.101.56 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-vm4kqfoffb6n;rport From: "6720" ;tag=2ps68pkrvk To: ;tag=as12f04922 Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 CSeq: 3 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom320/6.5.8 RTP-RxStat: Total_Rx_Pkts=683,Rx_Pkts=683,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=1073,Tx_Pkts=1073,Remote_Tx_Pkts=0 Content-Length: 0 --- (12 headers 0 lines) --- Transmitting (no NAT) to 220.238.243.11:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-vm4kqfoffb6n;received=220.238.243.11;rport=5060 From: "6720" ;tag=2ps68pkrvk To: ;tag=as12f04922 Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- pabx*CLI> <-- SIP read from 220.238.243.11:5060: BYE sip:0860121218@209.212.101.56 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-vm4kqfoffb6n;rport From: "6720" ;tag=2ps68pkrvk To: ;tag=as12f04922 Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 CSeq: 3 BYE Max-Forwards: 70 Contact: ;flow-id=1 User-Agent: snom320/6.5.8 RTP-RxStat: Total_Rx_Pkts=683,Rx_Pkts=683,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=1073,Tx_Pkts=1073,Remote_Tx_Pkts=0 Content-Length: 0 --- (12 headers 0 lines) --- Transmitting (no NAT) to 220.238.243.11:5060: SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-vm4kqfoffb6n;received=220.238.243.11;rport=5060 From: "6720" ;tag=2ps68pkrvk To: ;tag=as12f04922 Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- pabx*CLI>