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- -- Executing [8017478048@from-trunk:5] ExecIf("SIP/vitel-inbound-00000031", "0 ?Set(CALLERID(name)=8014100203)") in new stack
- -- Executing [8017478048@from-trunk:6] Set("SIP/vitel-inbound-00000031", "CHANNEL(musicclass)=moh") in new stack
- -- Executing [8017478048@from-trunk:7] Set("SIP/vitel-inbound-00000031", "__MOHCLASS=moh") in new stack
- -- Executing [8017478048@from-trunk:8] Ringing("SIP/vitel-inbound-00000031", "") in new stack
- <--- Transmitting (NAT) to 66.241.96.221:5060 --->
- SIP/2.0 180 Ringing
- Via: SIP/2.0/UDP 66.241.96.221:5060;branch=z9hG4bK242209e1;received=66.241.96.221;rport=5060
- From: "8014100203" <sip:8014100203@66.241.96.221>;tag=as6056c176
- To: <sip:8017478048@24.2.66.2:5060>;tag=as34b2bc52
- Call-ID: 7c1454a27f4333035455d68e0f484ae0@66.241.96.221
- CSeq: 102 INVITE
- Server: FPBX-2.10.0(1.8.7.1)
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:8017478048@24.2.66.2:5060>
- Content-Length: 0
- <------------>
- -- Executing [8017478048@from-trunk:9] Wait("SIP/vitel-inbound-00000031", "2") in new stack
- -- Executing [8017478048@from-trunk:10] Set("SIP/vitel-inbound-00000031", "__CALLINGPRES_SV=allowed_not_screened") in new stack
- -- Executing [8017478048@from-trunk:11] Set("SIP/vitel-inbound-00000031", "CALLERPRES()=allowed_not_screened") in new stack
- -- Executing [8017478048@from-trunk:12] Goto("SIP/vitel-inbound-00000031", "ivr-2,s,1") in new stack
- -- Goto (ivr-2,s,1)
- -- Executing [s@ivr-2:1] Set("SIP/vitel-inbound-00000031", "TIMEOUT_LOOPCOUNT=0") in new stack
- -- Executing [s@ivr-2:2] Set("SIP/vitel-inbound-00000031", "INVALID_LOOPCOUNT=0") in new stack
- -- Executing [s@ivr-2:3] Set("SIP/vitel-inbound-00000031", "_IVR_CONTEXT_ivr-2=") in new stack
- -- Executing [s@ivr-2:4] Set("SIP/vitel-inbound-00000031", "_IVR_CONTEXT=ivr-2") in new stack
- -- Executing [s@ivr-2:5] Set("SIP/vitel-inbound-00000031", "__IVR_RETVM=RETURN") in new stack
- -- Executing [s@ivr-2:6] GotoIf("SIP/vitel-inbound-00000031", "0?skip") in new stack
- -- Executing [s@ivr-2:7] Answer("SIP/vitel-inbound-00000031", "") in new stack
- Audio is at 5060
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x8 (alaw) to SDP
- Adding codec 0x2 (gsm) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- <--- Reliably Transmitting (NAT) to 66.241.96.221:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 66.241.96.221:5060;branch=z9hG4bK242209e1;received=66.241.96.221;rport=5060
- From: "8014100203" <sip:8014100203@66.241.96.221>;tag=as6056c176
- To: <sip:8017478048@24.2.66.2:5060>;tag=as34b2bc52
- Call-ID: 7c1454a27f4333035455d68e0f484ae0@66.241.96.221
- CSeq: 102 INVITE
- Server: FPBX-2.10.0(1.8.7.1)
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Contact: <sip:8017478048@24.2.66.2:5060>
- Content-Type: application/sdp
- Content-Length: 277
- v=0
- o=root 1917445873 1917445873 IN IP4 24.2.66.2
- s=Asterisk PBX 1.8.7.1
- c=IN IP4 24.2.66.2
- t=0 0
- m=audio 16808 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- <------------>
- <--- SIP read from UDP:66.241.96.221:5060 --->
- ACK sip:8017478048@24.2.66.2:5060 SIP/2.0
- Via: SIP/2.0/UDP 66.241.96.221:5060;branch=z9hG4bK444485e3;rport
- From: "8014100203" <sip:8014100203@66.241.96.221>;tag=as6056c176
- To: <sip:8017478048@24.2.66.2:5060>;tag=as34b2bc52
- Contact: <sip:8014100203@66.241.96.221>
- Call-ID: 7c1454a27f4333035455d68e0f484ae0@66.241.96.221
- CSeq: 102 ACK
- User-Agent: packetrino
- Max-Forwards: 70
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- -- Executing [s@ivr-2:8] Wait("SIP/vitel-inbound-00000031", "1") in new stack
- -- Executing [s@ivr-2:9] Set("SIP/vitel-inbound-00000031", "IVR_MSG=custom/ccwtechivr") in new stack
- -- Executing [s@ivr-2:10] Set("SIP/vitel-inbound-00000031", "TIMEOUT(digit)=3") in new stack
- -- Digit timeout set to 3.000
- -- Executing [s@ivr-2:11] ExecIf("SIP/vitel-inbound-00000031", "1?Background(custom/ccwtechivr)") in new stack
- -- <SIP/vitel-inbound-00000031> Playing 'custom/ccwtechivr.slin' (language 'en')
- == CDR updated on SIP/vitel-inbound-00000031
- -- Executing [1@ivr-2:1] Goto("SIP/vitel-inbound-00000031", "timeconditions,1,1") in new stack
- -- Goto (timeconditions,1,1)
- -- Executing [1@timeconditions:1] GotoIfTime("SIP/vitel-inbound-00000031", "08:00-17:00,mon-thu,*,*?truestate") in new stack
- -- Executing [1@timeconditions:2] GotoIf("SIP/vitel-inbound-00000031", "0?truegoto") in new stack
- -- Executing [1@timeconditions:3] ExecIf("SIP/vitel-inbound-00000031", "0?Set(DB(TC/1)=)") in new stack
- -- Executing [1@timeconditions:4] GotoIf("SIP/vitel-inbound-00000031", "1?ext-group,602,1") in new stack
- -- Goto (ext-group,602,1)
- -- Executing [602@ext-group:1] Macro("SIP/vitel-inbound-00000031", "user-callerid,") in new stack
- -- Executing [s@macro-user-callerid:1] Set("SIP/vitel-inbound-00000031", "AMPUSER=8014100203") in new stack
- -- Executing [s@macro-user-callerid:2] GotoIf("SIP/vitel-inbound-00000031", "0?report") in new stack
- -- Executing [s@macro-user-callerid:3] ExecIf("SIP/vitel-inbound-00000031", "1?Set(REALCALLERIDNUM=8014100203)") in new stack
- -- Executing [s@macro-user-callerid:4] Set("SIP/vitel-inbound-00000031", "AMPUSER=") in new stack
- -- Executing [s@macro-user-callerid:5] Set("SIP/vitel-inbound-00000031", "AMPUSERCIDNAME=") in new stack
- -- Executing [s@macro-user-callerid:6] GotoIf("SIP/vitel-inbound-00000031", "1?report") in new stack
- -- Goto (macro-user-callerid,s,13)
- -- Executing [s@macro-user-callerid:13] GotoIf("SIP/vitel-inbound-00000031", "0?continue") in new stack
- -- Executing [s@macro-user-callerid:14] Set("SIP/vitel-inbound-00000031", "__TTL=64") in new stack
- -- Executing [s@macro-user-callerid:15] GotoIf("SIP/vitel-inbound-00000031", "1?continue") in new stack
- -- Goto (macro-user-callerid,s,26)
- -- Executing [s@macro-user-callerid:26] Set("SIP/vitel-inbound-00000031", "CALLERID(number)=8014100203") in new stack
- -- Executing [s@macro-user-callerid:27] Set("SIP/vitel-inbound-00000031", "CALLERID(name)=SALT LAKE, UT") in new stack
- -- Executing [s@macro-user-callerid:28] Set("SIP/vitel-inbound-00000031", "CHANNEL(language)=en") in new stack
- -- Executing [602@ext-group:2] Macro("SIP/vitel-inbound-00000031", "blkvm-setifempty,") in new stack
- -- Executing [s@macro-blkvm-setifempty:1] GotoIf("SIP/vitel-inbound-00000031", "1?init") in new stack
- -- Goto (macro-blkvm-setifempty,s,4)
- -- Executing [s@macro-blkvm-setifempty:4] Set("SIP/vitel-inbound-00000031", "__BLKVM_CHANNEL=SIP/vitel-inbound-00000031") in new stack
- -- Executing [s@macro-blkvm-setifempty:5] Set("SIP/vitel-inbound-00000031", "SHARED(BLKVM,SIP/vitel-inbound-00000031)=TRUE") in new stack
- -- Executing [s@macro-blkvm-setifempty:6] Set("SIP/vitel-inbound-00000031", "GOSUB_RETVAL=TRUE") in new stack
- -- Executing [s@macro-blkvm-setifempty:7] MacroExit("SIP/vitel-inbound-00000031", "") in new stack
- -- Executing [602@ext-group:3] GotoIf("SIP/vitel-inbound-00000031", "1?skipov") in new stack
- -- Goto (ext-group,602,6)
- -- Executing [602@ext-group:6] Set("SIP/vitel-inbound-00000031", "RRNODEST=") in new stack
- -- Executing [602@ext-group:7] Set("SIP/vitel-inbound-00000031", "__NODEST=602") in new stack
- -- Executing [602@ext-group:8] GosubIf("SIP/vitel-inbound-00000031", "0?sub-rgsetcid,s,1()") in new stack
- -- Executing [602@ext-group:9] Gosub("SIP/vitel-inbound-00000031", "sub-record-check,s,1(rg,602,dontcare)") in new stack
- -- Executing [s@sub-record-check:1] GotoIf("SIP/vitel-inbound-00000031", "1?check") in new stack
- -- Goto (sub-record-check,s,6)
- -- Executing [s@sub-record-check:6] Set("SIP/vitel-inbound-00000031", "__MON_FMT=wav") in new stack
- -- Executing [s@sub-record-check:7] GotoIf("SIP/vitel-inbound-00000031", "1?next") in new stack
- -- Goto (sub-record-check,s,10)
- -- Executing [s@sub-record-check:10] ExecIf("SIP/vitel-inbound-00000031", "0?Return()") in new stack
- -- Executing [s@sub-record-check:11] GotoIf("SIP/vitel-inbound-00000031", "0?rg,1") in new stack
- -- Executing [s@sub-record-check:12] Set("SIP/vitel-inbound-00000031", "__REC_STATUS=INITIALIZED") in new stack
- -- Executing [s@sub-record-check:13] ExecIf("SIP/vitel-inbound-00000031", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
- -- Executing [s@sub-record-check:14] Set("SIP/vitel-inbound-00000031", "NOW=1337856316") in new stack
- -- Executing [s@sub-record-check:15] Set("SIP/vitel-inbound-00000031", "__DAY=24") in new stack
- -- Executing [s@sub-record-check:16] Set("SIP/vitel-inbound-00000031", "__MONTH=05") in new stack
- -- Executing [s@sub-record-check:17] Set("SIP/vitel-inbound-00000031", "__YEAR=2012") in new stack
- -- Executing [s@sub-record-check:18] Set("SIP/vitel-inbound-00000031", "__TIMESTR=20120524-044516") in new stack
- -- Executing [s@sub-record-check:19] Set("SIP/vitel-inbound-00000031", "__FROMEXTEN=8014100203") in new stack
- -- Executing [s@sub-record-check:20] Set("SIP/vitel-inbound-00000031", "__CALLFILENAME=rg-602-8014100203-20120524-044516-1337856306.69") in new stack
- -- Executing [s@sub-record-check:21] Goto("SIP/vitel-inbound-00000031", "rg,1") in new stack
- -- Goto (sub-record-check,rg,1)
- -- Executing [rg@sub-record-check:1] GosubIf("SIP/vitel-inbound-00000031", "0?record,1(rg,dontcare,8014100203)") in new stack
- -- Executing [rg@sub-record-check:2] Return("SIP/vitel-inbound-00000031", "") in new stack
- -- Executing [602@ext-group:10] Set("SIP/vitel-inbound-00000031", "RingGroupMethod=ringall") in new stack
- -- Executing [602@ext-group:11] GotoIf("SIP/vitel-inbound-00000031", "0?DIALGRP") in new stack
- -- Executing [602@ext-group:12] Answer("SIP/vitel-inbound-00000031", "") in new stack
- -- Executing [602@ext-group:13] Wait("SIP/vitel-inbound-00000031", "1") in new stack
- -- Executing [602@ext-group:14] Playback("SIP/vitel-inbound-00000031", "custom/plzwait") in new stack
- -- <SIP/vitel-inbound-00000031> Playing 'custom/plzwait.slin' (language 'en')
- -- Executing [602@ext-group:15] Macro("SIP/vitel-inbound-00000031", "dial,20,m(moh)t,101-8012096684#") in new stack
- -- Executing [s@macro-dial:1] GotoIf("SIP/vitel-inbound-00000031", "0?dial") in new stack
- -- Executing [s@macro-dial:2] SetMusicOnHold("SIP/vitel-inbound-00000031", "moh") in new stack
- -- Executing [s@macro-dial:3] AGI("SIP/vitel-inbound-00000031", "dialparties.agi") in new stack
- -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
- dialparties.agi: Starting New Dialparties.agi
- dialparties.agi: Caller ID name is 'SALT LAKE, UT' number is '8014100203'
- > dialparties.agi: USE_CONFIRMATION: 'FALSE'
- > dialparties.agi: RINGGROUP_INDEX: ''
- dialparties.agi: Methodology of ring is 'ringall'
- -- dialparties.agi: Added extension 101 to extension map
- -- dialparties.agi: Added extension 8012096684# to extension map
- -- dialparties.agi: Extension 101 cf is disabled
- -- dialparties.agi: Extension 8012096684# cf is disabled
- -- dialparties.agi: Extension 101 do not disturb is disabled
- > dialparties.agi: extnum 101 has: cw: 1; hascfb: 0 [] hascfu: 0 []
- -- dialparties.agi: dbset CALLTRACE/101 to 8014100203
- > dialparties.agi: extnum 8012096684# has: cw: 0; hascfb: 0 [] hascfu: 0 []
- > dialparties.agi: Built External dialstring component for 8012096684: Local/8012096684@from-internal/n
- -- dialparties.agi: Filtered ARG3: 101-8012096684
- > dialparties.agi: NODEST: 602 adding M(auto-blkvm) to dialopts: m(moh)tM(auto-blkvm)
- > dialparties.agi: NODEST: 602 blkvm enabled macro already in dialopts: m(moh)tM(auto-blkvm)
- -- <SIP/vitel-inbound-00000031>AGI Script dialparties.agi completed, returning 0
- -- Executing [s@macro-dial:7] Dial("SIP/vitel-inbound-00000031", "SIP/101&Local/8012096684@from-internal/n,20,m(moh)tM(auto-blkvm)") in new stack
- == Using SIP RTP TOS bits 184
- == Using SIP RTP CoS mark 5
- Audio is at 5060
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x8 (alaw) to SDP
- Adding codec 0x2 (gsm) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (no NAT) to 10.0.0.21:5060:
- INVITE sip:101@10.0.0.21:5060 SIP/2.0
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK36e91778
- Max-Forwards: 70
- From: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
- To: <sip:101@10.0.0.21:5060>
- Contact: <sip:8014100203@24.2.66.2:5060>
- Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
- CSeq: 102 INVITE
- User-Agent: FPBX-2.10.0(1.8.7.1)
- Date: Thu, 24 May 2012 10:45:20 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Type: application/sdp
- Content-Length: 277
- v=0
- o=root 1090802124 1090802124 IN IP4 24.2.66.2
- s=Asterisk PBX 1.8.7.1
- c=IN IP4 24.2.66.2
- t=0 0
- m=audio 18128 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- -- Called SIP/101
- -- Called Local/8012096684@from-internal/n
- -- Started music on hold, class 'default', on SIP/vitel-inbound-00000031
- -- SIP/101-00000032 connected line has changed. Saving it until answer for SIP/vitel-inbound-00000031
- -- Executing [8012096684@from-internal:1] Macro("Local/8012096684@from-internal-6479;2", "user-callerid,LIMIT,") in new stack
- -- Executing [s@macro-user-callerid:1] Set("Local/8012096684@from-internal-6479;2", "AMPUSER=8014100203") in new stack
- -- Executing [s@macro-user-callerid:2] GotoIf("Local/8012096684@from-internal-6479;2", "0?report") in new stack
- -- Executing [s@macro-user-callerid:3] ExecIf("Local/8012096684@from-internal-6479;2", "1?Set(REALCALLERIDNUM=8014100203)") in new stack
- -- Executing [s@macro-user-callerid:4] Set("Local/8012096684@from-internal-6479;2", "AMPUSER=") in new stack
- -- Executing [s@macro-user-callerid:5] Set("Local/8012096684@from-internal-6479;2", "AMPUSERCIDNAME=") in new stack
- -- Executing [s@macro-user-callerid:6] GotoIf("Local/8012096684@from-internal-6479;2", "1?report") in new stack
- -- Goto (macro-user-callerid,s,13)
- -- Executing [s@macro-user-callerid:13] GotoIf("Local/8012096684@from-internal-6479;2", "1?continue") in new stack
- -- Goto (macro-user-callerid,s,26)
- -- Executing [s@macro-user-callerid:26] Set("Local/8012096684@from-internal-6479;2", "CALLERID(number)=8014100203") in new stack
- -- Executing [s@macro-user-callerid:27] Set("Local/8012096684@from-internal-6479;2", "CALLERID(name)=SALT LAKE, UT") in new stack
- -- Executing [s@macro-user-callerid:28] Set("Local/8012096684@from-internal-6479;2", "CHANNEL(language)=en") in new stack
- -- Executing [8012096684@from-internal:2] Set("Local/8012096684@from-internal-6479;2", "MOHCLASS=moh") in new stack
- -- Executing [8012096684@from-internal:3] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(TRUNKCIDOVERRIDE=<8014100203>)") in new stack
- -- Executing [8012096684@from-internal:4] Set("Local/8012096684@from-internal-6479;2", "_NODEST=") in new stack
- -- Executing [8012096684@from-internal:5] Gosub("Local/8012096684@from-internal-6479;2", "sub-record-check,s,1(out,8012096684,)") in new stack
- -- Executing [s@sub-record-check:1] GotoIf("Local/8012096684@from-internal-6479;2", "1?check") in new stack
- -- Goto (sub-record-check,s,6)
- -- Executing [s@sub-record-check:6] Set("Local/8012096684@from-internal-6479;2", "__MON_FMT=wav") in new stack
- -- Executing [s@sub-record-check:7] GotoIf("Local/8012096684@from-internal-6479;2", "1?next") in new stack
- -- Goto (sub-record-check,s,10)
- -- Executing [s@sub-record-check:10] ExecIf("Local/8012096684@from-internal-6479;2", "0?Return()") in new stack
- -- Executing [s@sub-record-check:11] GotoIf("Local/8012096684@from-internal-6479;2", "1?out,1") in new stack
- -- Goto (sub-record-check,out,1)
- -- Executing [out@sub-record-check:1] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(__REC_POLICY_MODE=)") in new stack
- -- Executing [out@sub-record-check:2] GosubIf("Local/8012096684@from-internal-6479;2", "0?record,1(exten,8012096684,8014100203)") in new stack
- -- Executing [out@sub-record-check:3] Return("Local/8012096684@from-internal-6479;2", "") in new stack
- -- Executing [8012096684@from-internal:6] Macro("Local/8012096684@from-internal-6479;2", "dialout-trunk,2,18012096684,") in new stack
- -- Executing [s@macro-dialout-trunk:1] Set("Local/8012096684@from-internal-6479;2", "DIAL_TRUNK=2") in new stack
- -- Executing [s@macro-dialout-trunk:2] GosubIf("Local/8012096684@from-internal-6479;2", "0?sub-pincheck,s,1()") in new stack
- -- Executing [s@macro-dialout-trunk:3] GotoIf("Local/8012096684@from-internal-6479;2", "0?disabletrunk,1") in new stack
- -- Executing [s@macro-dialout-trunk:4] Set("Local/8012096684@from-internal-6479;2", "DIAL_NUMBER=18012096684") in new stack
- -- Executing [s@macro-dialout-trunk:5] Set("Local/8012096684@from-internal-6479;2", "DIAL_TRUNK_OPTIONS=tr") in new stack
- <--- SIP read from UDP:10.0.0.21:5060 --->
- SIP/2.0 100 Trying
- To: <sip:101@10.0.0.21:5060>
- From: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
- Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
- CSeq: 102 INVITE
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK36e91778
- Server: Linksys/PAP2T-5.1.6(LS)
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- -- Executing [s@macro-dialout-trunk:6] Set("Local/8012096684@from-internal-6479;2", "OUTBOUND_GROUP=OUT_2") in new stack
- -- Executing [s@macro-dialout-trunk:7] GotoIf("Local/8012096684@from-internal-6479;2", "0?nomax") in new stack
- -- Executing [s@macro-dialout-trunk:8] GotoIf("Local/8012096684@from-internal-6479;2", "0?chanfull") in new stack
- -- Executing [s@macro-dialout-trunk:9] GotoIf("Local/8012096684@from-internal-6479;2", "0?skipoutcid") in new stack
- -- Executing [s@macro-dialout-trunk:10] Set("Local/8012096684@from-internal-6479;2", "DIAL_TRUNK_OPTIONS=") in new stack
- -- Executing [s@macro-dialout-trunk:11] Macro("Local/8012096684@from-internal-6479;2", "outbound-callerid,2") in new stack
- -- Executing [s@macro-outbound-callerid:1] ExecIf("Local/8012096684@from-internal-6479;2", "1?Set(CALLERPRES()=allowed_not_screened)") in new stack
- -- Executing [s@macro-outbound-callerid:2] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(REALCALLERIDNUM=8014100203)") in new stack
- -- Executing [s@macro-outbound-callerid:3] GotoIf("Local/8012096684@from-internal-6479;2", "0?normcid") in new stack
- -- Executing [s@macro-outbound-callerid:4] Set("Local/8012096684@from-internal-6479;2", "USEROUTCID=8014100203") in new stack
- -- Executing [s@macro-outbound-callerid:5] GotoIf("Local/8012096684@from-internal-6479;2", "1?bypass") in new stack
- -- Goto (macro-outbound-callerid,s,7)
- -- Executing [s@macro-outbound-callerid:7] Set("Local/8012096684@from-internal-6479;2", "EMERGENCYCID=") in new stack
- -- Executing [s@macro-outbound-callerid:8] Set("Local/8012096684@from-internal-6479;2", "TRUNKOUTCID=<8017478048>") in new stack
- -- Executing [s@macro-outbound-callerid:9] GotoIf("Local/8012096684@from-internal-6479;2", "1?trunkcid") in new stack
- -- Goto (macro-outbound-callerid,s,12)
- -- Executing [s@macro-outbound-callerid:12] ExecIf("Local/8012096684@from-internal-6479;2", "1?Set(CALLERID(all)=<8017478048>)") in new stack
- -- Executing [s@macro-outbound-callerid:13] ExecIf("Local/8012096684@from-internal-6479;2", "1?Set(CALLERID(all)=8014100203)") in new stack
- -- Executing [s@macro-outbound-callerid:14] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(CALLERID(all)=)") in new stack
- -- Executing [s@macro-outbound-callerid:15] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
- -- Executing [s@macro-dialout-trunk:12] GosubIf("Local/8012096684@from-internal-6479;2", "0?sub-flp-2,s,1()") in new stack
- -- Executing [s@macro-dialout-trunk:13] Set("Local/8012096684@from-internal-6479;2", "OUTNUM=18012096684") in new stack
- -- Executing [s@macro-dialout-trunk:14] Set("Local/8012096684@from-internal-6479;2", "custom=SIP/vitel-outbound") in new stack
- -- Executing [s@macro-dialout-trunk:15] ExecIf("Local/8012096684@from-internal-6479;2", "1?Set(DIAL_TRUNK_OPTIONS=M(setmusic^moh))") in new stack
- -- Executing [s@macro-dialout-trunk:16] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^moh)M(confirm))") in new stack
- -- Executing [s@macro-dialout-trunk:17] Macro("Local/8012096684@from-internal-6479;2", "dialout-trunk-predial-hook,") in new stack
- -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("Local/8012096684@from-internal-6479;2", "") in new stack
- -- Executing [s@macro-dialout-trunk:18] GotoIf("Local/8012096684@from-internal-6479;2", "0?bypass,1") in new stack
- -- Executing [s@macro-dialout-trunk:19] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(CONNECTEDLINE(num,i)=18012096684)") in new stack
- -- Executing [s@macro-dialout-trunk:20] ExecIf("Local/8012096684@from-internal-6479;2", "0?Set(CONNECTEDLINE(name,i)=CID:8014100203)") in new stack
- -- Executing [s@macro-dialout-trunk:21] GotoIf("Local/8012096684@from-internal-6479;2", "0?customtrunk") in new stack
- -- Executing [s@macro-dialout-trunk:22] Dial("Local/8012096684@from-internal-6479;2", "SIP/vitel-outbound/18012096684,300,M(setmusic^moh)") in new stack
- == Using SIP RTP TOS bits 184
- == Using SIP RTP CoS mark 5
- <--- SIP read from UDP:10.0.0.21:5060 --->
- SIP/2.0 180 Ringing
- To: <sip:101@10.0.0.21:5060>;tag=c95afd287417c8d0i0
- From: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
- Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
- CSeq: 102 INVITE
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK36e91778
- Server: Linksys/PAP2T-5.1.6(LS)
- Content-Length: 0
- <------------->
- --- (8 headers 0 lines) ---
- -- SIP/101-00000032 is ringing
- Audio is at 5060
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x8 (alaw) to SDP
- Adding codec 0x2 (gsm) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 64.2.142.17:5060:
- INVITE sip:18012096684@outbound.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0aaf31c1;rport
- Max-Forwards: 70
- From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
- To: <sip:18012096684@outbound.vitelity.net>
- Contact: <sip:ccwt_pbx@24.2.66.2:5060>
- Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
- CSeq: 102 INVITE
- User-Agent: FPBX-2.10.0(1.8.7.1)
- Date: Thu, 24 May 2012 10:45:20 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Remote-Party-ID: "8014100203" <sip:8014100203@24.2.66.2>;party=calling;privacy=off;screen=no
- Content-Type: application/sdp
- Content-Length: 275
- v=0
- o=root 740845620 740845620 IN IP4 24.2.66.2
- s=Asterisk PBX 1.8.7.1
- c=IN IP4 24.2.66.2
- t=0 0
- m=audio 19396 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- -- Called SIP/vitel-outbound/18012096684
- <--- SIP read from UDP:64.2.142.17:5060 --->
- SIP/2.0 407 Proxy Authentication Required
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0aaf31c1;received=24.2.66.2;rport=5060
- From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
- To: <sip:18012096684@outbound.vitelity.net>;tag=as0c51706e
- Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4b0c3da9"
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Transmitting (NAT) to 64.2.142.17:5060:
- ACK sip:18012096684@outbound.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0aaf31c1;rport
- Max-Forwards: 70
- From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
- To: <sip:18012096684@outbound.vitelity.net>;tag=as0c51706e
- Contact: <sip:ccwt_pbx@24.2.66.2:5060>
- Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
- CSeq: 102 ACK
- User-Agent: FPBX-2.10.0(1.8.7.1)
- Content-Length: 0
- ---
- Audio is at 5060
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x8 (alaw) to SDP
- Adding codec 0x2 (gsm) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 64.2.142.17:5060:
- INVITE sip:18012096684@outbound.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;rport
- Max-Forwards: 70
- From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
- To: <sip:18012096684@outbound.vitelity.net>
- Contact: <sip:ccwt_pbx@24.2.66.2:5060>
- Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
- CSeq: 103 INVITE
- User-Agent: FPBX-2.10.0(1.8.7.1)
- Proxy-Authorization: Digest username="ccwt_pbx", realm="asterisk", algorithm=MD5, uri="sip:18012096684@outbound.vitelity.net", nonce="4b0c3da9", response="2f75855df13197b2a86df0821110ed2a"
- Date: Thu, 24 May 2012 10:45:20 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Remote-Party-ID: "8014100203" <sip:8014100203@24.2.66.2>;party=calling;privacy=off;screen=no
- Content-Type: application/sdp
- Content-Length: 275
- v=0
- o=root 740845620 740845621 IN IP4 24.2.66.2
- s=Asterisk PBX 1.8.7.1
- c=IN IP4 24.2.66.2
- t=0 0
- m=audio 19396 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- ---
- <--- SIP read from UDP:64.2.142.17:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;received=24.2.66.2;rport=5060
- From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
- To: <sip:18012096684@outbound.vitelity.net>
- Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
- CSeq: 103 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:18012096684@64.2.142.17>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- <--- SIP read from UDP:64.2.142.17:5060 --->
- SIP/2.0 183 Session Progress
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;received=24.2.66.2;rport=5060
- From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
- To: <sip:18012096684@outbound.vitelity.net>;tag=as43c26d7d
- Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
- CSeq: 103 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Contact: <sip:18012096684@64.2.142.17>
- ontent-Type: application/sdp
- Content-Length: 285
- v=0
- o=root 15443 15443 IN IP4 64.2.142.17
- s=session
- c=IN IP4 64.2.142.17
- t=0 0
- m=audio 15014 RTP/AVP 0 8 3 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=silenceSupp:off - - - -
- a=ptime:20
- a=sendrecv
- <------------->
- --- (12 headers 14 lines) ---
- Found RTP audio format 0
- Found RTP audio format 8
- Found RTP audio format 3
- Found RTP audio format 101
- Found audio description format PCMU for ID 0
- Found audio description format PCMA for ID 8
- Found audio description format GSM for ID 3
- Found audio description format telephone-event for ID 101
- Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
- Peer audio RTP is at port 64.2.142.17:15014
- -- SIP/vitel-outbound-00000033 is making progress passing it to Local/8012096684@from-internal-6479;2
- -- Local/8012096684@from-internal-6479;1 is making progress passing it to SIP/vitel-inbound-00000031
- <--- SIP read from UDP:10.0.0.21:5060 --->
- SIP/2.0 200 OK
- To: <sip:101@10.0.0.21:5060>;tag=c95afd287417c8d0i0
- From: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
- Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
- CSeq: 102 INVITE
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK36e91778
- Contact: Allen <sip:101@10.0.0.21:5060>
- Server: Linksys/PAP2T-5.1.6(LS)
- Content-Length: 251
- Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
- Supported: x-sipura, replaces
- Content-Type: application/sdp
- v=0
- o=- 36286809 36286809 IN IP4 10.0.0.21
- s=-
- c=IN IP4 10.0.0.21
- t=0 0
- m=audio 16408 RTP/AVP 0 100 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:100 NSE/8000
- a=fmtp:100 192-193
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-15
- a=ptime:10
- a=sendrecv
- <------------->
- --- (12 headers 13 lines) ---
- Found RTP audio format 0
- Found RTP audio format 100
- Found RTP audio format 101
- Found audio description format PCMU for ID 0
- Found unknown media description format NSE for ID 100
- Found audio description format telephone-event for ID 101
- Capabilities: us - 0xe (gsm|ulaw|alaw), 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 10.0.0.21:16408
- list_route: hop: <sip:101@10.0.0.21:5060>
- set_destination: Parsing <sip:101@10.0.0.21:5060> for address/port to send to
- set_destination: set destination to 10.0.0.21:5060
- Transmitting (no NAT) to 10.0.0.21:5060:
- ACK sip:101@10.0.0.21:5060 SIP/2.0
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK1312b086
- Max-Forwards: 70
- From: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
- To: <sip:101@10.0.0.21:5060>;tag=c95afd287417c8d0i0
- Contact: <sip:8014100203@24.2.66.2:5060>
- Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
- CSeq: 102 ACK
- User-Agent: FPBX-2.10.0(1.8.7.1)
- Content-Length: 0
- ---
- -- SIP/101-00000032 connected line has changed. Saving it until answer for SIP/vitel-inbound-00000031
- -- SIP/101-00000032 answered SIP/vitel-inbound-00000031
- Scheduling destruction of SIP dialog '73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060' in 32000 ms (Method: INVITE)
- Reliably Transmitting (NAT) to 64.2.142.17:5060:
- CANCEL sip:18012096684@outbound.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;rport
- Max-Forwards: 70
- From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
- To: <sip:18012096684@outbound.vitelity.net>
- Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
- CSeq: 103 CANCEL
- User-Agent: FPBX-2.10.0(1.8.7.1)
- Content-Length: 0
- ---
- Scheduling destruction of SIP dialog '73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060' in 32000 ms (Method: INVITE)
- == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'Local/8012096684@from-internal-6479;2' in macro 'dialout-trunk'
- == Spawn extension (from-internal, 8012096684, 6) exited non-zero on 'Local/8012096684@from-internal-6479;2'
- -- Executing [s@macro-auto-blkvm:1] Set("SIP/101-00000032", "__MACRO_RESULT=") in new stack
- -- Executing [s@macro-auto-blkvm:2] Macro("SIP/101-00000032", "blkvm-clr,") in new stack
- -- Executing [h@from-internal:1] Hangup("Local/8012096684@from-internal-6479;2", "") in new stack
- == Spawn extension (from-internal, h, 1) exited non-zero on 'Local/8012096684@from-internal-6479;2'
- -- Executing [s@macro-blkvm-clr:1] Set("SIP/101-00000032", "SHARED(BLKVM,SIP/vitel-inbound-00000031)=") in new stack
- -- Executing [s@macro-blkvm-clr:2] Set("SIP/101-00000032", "GOSUB_RETVAL=") in new stack
- -- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/101-00000032", "") in new stack
- -- Executing [s@macro-auto-blkvm:3] ExecIf("SIP/101-00000032", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=101)") in new stack
- -- Executing [s@macro-auto-blkvm:4] ExecIf("SIP/101-00000032", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Allen)") in new stack
- -- Stopped music on hold on SIP/vitel-inbound-00000031
- <--- SIP read from UDP:64.2.142.17:5060 --->
- SIP/2.0 487 Request Terminated
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;received=24.2.66.2;rport=5060
- From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
- To: <sip:18012096684@outbound.vitelity.net>;tag=as43c26d7d
- Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
- CSeq: 103 INVITE
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- Transmitting (NAT) to 64.2.142.17:5060:
- ACK sip:18012096684@outbound.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;rport
- Max-Forwards: 70
- From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
- To: <sip:18012096684@outbound.vitelity.net>;tag=as43c26d7d
- Contact: <sip:ccwt_pbx@24.2.66.2:5060>
- Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
- CSeq: 103 ACK
- User-Agent: FPBX-2.10.0(1.8.7.1)
- Content-Length: 0
- ---
- <--- SIP read from UDP:64.2.142.17:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK2f0d3c1b;received=24.2.66.2;rport=5060
- From: "8014100203" <sip:ccwt_pbx@24.2.66.2>;tag=as6011faed
- To: <sip:18012096684@outbound.vitelity.net>;tag=as43c26d7d
- Call-ID: 73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060
- CSeq: 103 CANCEL
- User-Agent: Asterisk PBX
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- Really destroying SIP dialog '73ddf3252cd2392c42bac30a0cbf6cf8@24.2.66.2:5060' Method: INVITE
- Really destroying SIP dialog '3691bc792f8f478d2df520f95421c67c@127.0.0.1' Method: REGISTER
- Reliably Transmitting (no NAT) to 10.0.0.21:5060:
- OPTIONS sip:101@10.0.0.21:5060 SIP/2.0
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK5533299c
- Max-Forwards: 70
- From: "Unknown" <sip:Unknown@24.2.66.2>;tag=as7c2b8c12
- To: <sip:101@10.0.0.21:5060>
- Contact: <sip:Unknown@24.2.66.2:5060>
- Call-ID: 19616e8a5d7cc8e208172ecf5eb618ea@24.2.66.2:5060
- CSeq: 102 OPTIONS
- User-Agent: FPBX-2.10.0(1.8.7.1)
- Date: Thu, 24 May 2012 10:45:42 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- ---
- <--- SIP read from UDP:10.0.0.21:5060 --->
- SIP/2.0 200 OK
- To: <sip:101@10.0.0.21:5060>;tag=59981087efc9843ei0
- From: "Unknown" <sip:Unknown@24.2.66.2>;tag=as7c2b8c12
- Call-ID: 19616e8a5d7cc8e208172ecf5eb618ea@24.2.66.2:5060
- CSeq: 102 OPTIONS
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK5533299c
- Server: Linksys/PAP2T-5.1.6(LS)
- Content-Length: 0
- Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
- Supported: x-sipura, replaces
- <------------->
- --- (10 headers 0 lines) ---
- Really destroying SIP dialog '19616e8a5d7cc8e208172ecf5eb618ea@24.2.66.2:5060' Method: OPTIONS
- <--- SIP read from UDP:10.0.0.21:5060 --->
- BYE sip:8014100203@24.2.66.2:5060 SIP/2.0
- Via: SIP/2.0/UDP 10.0.0.21:5060;branch=z9hG4bK-f1c9fda4
- From: <sip:101@10.0.0.21:5060>;tag=c95afd287417c8d0i0
- To: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
- Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
- CSeq: 101 BYE
- Max-Forwards: 70
- User-Agent: Linksys/PAP2T-5.1.6(LS)
- Content-Length: 0
- <------------->
- --- (9 headers 0 lines) ---
- Sending to 10.0.0.21:5060 (no NAT)
- Scheduling destruction of SIP dialog '434acf061c5563f74228d2da0d655276@24.2.66.2:5060' in 6400 ms (Method: BYE)
- <--- Transmitting (no NAT) to 10.0.0.21:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 10.0.0.21:5060;branch=z9hG4bK-f1c9fda4;received=10.0.0.21
- From: <sip:101@10.0.0.21:5060>;tag=c95afd287417c8d0i0
- To: "SALT LAKE, UT" <sip:8014100203@24.2.66.2>;tag=as5d7e5abc
- Call-ID: 434acf061c5563f74228d2da0d655276@24.2.66.2:5060
- CSeq: 101 BYE
- Server: FPBX-2.10.0(1.8.7.1)
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
- Supported: replaces, timer
- Content-Length: 0
- <------------>
- -- Executing [h@macro-dial:1] Macro("SIP/vitel-inbound-00000031", "hangupcall") in new stack
- -- Executing [s@macro-hangupcall:1] GotoIf("SIP/vitel-inbound-00000031", "1?theend") in new stack
- -- Goto (macro-hangupcall,s,3)
- -- Executing [s@macro-hangupcall:3] Hangup("SIP/vitel-inbound-00000031", "") in new stack
- == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/vitel-inbound-00000031' in macro 'hangupcall'
- == Spawn extension (macro-dial, h, 1) exited non-zero on 'SIP/vitel-inbound-00000031'
- == Spawn extension (macro-dial, s, 7) exited non-zero on 'SIP/vitel-inbound-00000031' in macro 'dial'
- == Spawn extension (ext-group, 602, 15) exited non-zero on 'SIP/vitel-inbound-00000031'
- Scheduling destruction of SIP dialog '7c1454a27f4333035455d68e0f484ae0@66.241.96.221' in 32000 ms (Method: ACK)
- set_destination: Parsing <sip:8014100203@66.241.96.221> for address/port to send to
- set_destination: set destination to 66.241.96.221:5060
- Reliably Transmitting (NAT) to 66.241.96.221:5060:
- BYE sip:8014100203@66.241.96.221 SIP/2.0
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK77471b8a;rport
- Max-Forwards: 70
- From: <sip:8017478048@24.2.66.2:5060>;tag=as34b2bc52
- To: "8014100203" <sip:8014100203@66.241.96.221>;tag=as6056c176
- Call-ID: 7c1454a27f4333035455d68e0f484ae0@66.241.96.221
- CSeq: 102 BYE
- User-Agent: FPBX-2.10.0(1.8.7.1)
- X-Asterisk-HangupCause: Normal Clearing
- X-Asterisk-HangupCauseCode: 16
- Content-Length: 0
- ---
- <--- SIP read from UDP:66.241.96.221:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK77471b8a;received=24.2.66.2;rport=5060
- From: <sip:8017478048@24.2.66.2:5060>;tag=as34b2bc52
- To: "8014100203" <sip:8014100203@66.241.96.221>;tag=as6056c176
- Call-ID: 7c1454a27f4333035455d68e0f484ae0@66.241.96.221
- CSeq: 102 BYE
- User-Agent: packetrino
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- SIP Response message for INCOMING dialog BYE arrived
- Really destroying SIP dialog '7c1454a27f4333035455d68e0f484ae0@66.241.96.221' Method: ACK
- > doing dnsmgr_lookup for 'inbound27.vitelity.net'
- REGISTER 11 headers, 0 lines
- Reliably Transmitting (NAT) to 66.241.96.221:5060:
- REGISTER sip:inbound27.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK69da1905;rport
- Max-Forwards: 70
- From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as43d1760d
- To: <sip:ccwt_pbx@inbound27.vitelity.net>
- Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
- CSeq: 1743 REGISTER
- User-Agent: FPBX-2.10.0(1.8.7.1)
- Authorization: Digest username="ccwt_pbx", realm="asterisk", algorithm=MD5, uri="sip:inbound27.vitelity.net", nonce="2148295e", response="b0a92dc82d45d575f4744e2979c20b85"
- Expires: 120
- Contact: <sip:s@24.2.66.2:5060>
- Content-Length: 0
- ---
- <--- SIP read from UDP:66.241.96.221:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK69da1905;received=24.2.66.2;rport=5060
- From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as43d1760d
- To: <sip:ccwt_pbx@inbound27.vitelity.net>
- Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
- CSeq: 1743 REGISTER
- User-Agent: packetrino
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- <--- SIP read from UDP:66.241.96.221:5060 --->
- SIP/2.0 401 Unauthorized
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK69da1905;received=24.2.66.2;rport=5060
- From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as43d1760d
- To: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as1ac189f0
- Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
- CSeq: 1743 REGISTER
- User-Agent: packetrino
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="66e9c3b7"
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Responding to challenge, registration to domain/host name inbound27.vitelity.net
- > doing dnsmgr_lookup for 'inbound27.vitelity.net'
- REGISTER 11 headers, 0 lines
- Reliably Transmitting (NAT) to 66.241.96.221:5060:
- REGISTER sip:inbound27.vitelity.net SIP/2.0
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0d572584;rport
- Max-Forwards: 70
- From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as2d999541
- To: <sip:ccwt_pbx@inbound27.vitelity.net>
- Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
- CSeq: 1744 REGISTER
- User-Agent: FPBX-2.10.0(1.8.7.1)
- Authorization: Digest username="ccwt_pbx", realm="asterisk", algorithm=MD5, uri="sip:inbound27.vitelity.net", nonce="66e9c3b7", response="bc17716d8d6afa2621f68b7ede110660"
- Expires: 120
- Contact: <sip:s@24.2.66.2:5060>
- Content-Length: 0
- ---
- <--- SIP read from UDP:66.241.96.221:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0d572584;received=24.2.66.2;rport=5060
- From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as2d999541
- To: <sip:ccwt_pbx@inbound27.vitelity.net>
- Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
- CSeq: 1744 REGISTER
- User-Agent: packetrino
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Length: 0
- <------------->
- --- (10 headers 0 lines) ---
- <--- SIP read from UDP:66.241.96.221:5060 --->
- SIP/2.0 200 OK
- Via: SIP/2.0/UDP 24.2.66.2:5060;branch=z9hG4bK0d572584;received=24.2.66.2;rport=5060
- From: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as2d999541
- To: <sip:ccwt_pbx@inbound27.vitelity.net>;tag=as1ac189f0
- Call-ID: 3691bc792f8f478d2df520f95421c67c@127.0.0.1
- CSeq: 1744 REGISTER
- User-Agent: packetrino
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Expires: 60
- Contact: <sip:s@24.2.66.2:5060>;expires=60
- Date: Thu, 24 May 2012 10:45:46 GMT
- Content-Length: 0
- <------------->
- --- (13 headers 0 lines) ---
- Scheduling destruction of SIP dialog '3691bc792f8f478d2df520f95421c67c@127.0.0.1' in 32000 ms (Method: REGISTER)
- Really destroying SIP dialog '434acf061c5563f74228d2da0d655276@24.2.66.2:5060' Method: BYE
- pbx*CLI>
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