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- Verbosity is at least 10
- == Using SIP RTP TOS bits 184
- == Using SIP RTP CoS mark 5
- == Using SIP VRTP TOS bits 136
- == Using SIP VRTP CoS mark 6
- -- Executing [*19101@from-internal:1] Goto("SIP/4656-0000071c", "from-internal-original,*19101,1") in new stack
- -- Goto (from-internal-original,*19101,1)
- -- Executing [*19101@from-internal-original:1] Macro("SIP/4656-0000071c", "user-callerid,SKIPTTL,") in new stack
- -- Executing [s@macro-user-callerid:1] Set("SIP/4656-0000071c", "AMPUSER=4656") in new stack
- -- Executing [s@macro-user-callerid:2] GotoIf("SIP/4656-0000071c", "0?report") in new stack
- -- Executing [s@macro-user-callerid:3] ExecIf("SIP/4656-0000071c", "1?Set(REALCALLERIDNUM=4656)") in new stack
- -- Executing [s@macro-user-callerid:4] Set("SIP/4656-0000071c", "AMPUSER=4656") in new stack
- -- Executing [s@macro-user-callerid:5] Set("SIP/4656-0000071c", "AMPUSERCIDNAME=Andy Ortlieb") in new stack
- -- Executing [s@macro-user-callerid:6] GotoIf("SIP/4656-0000071c", "0?report") in new stack
- -- Executing [s@macro-user-callerid:7] Set("SIP/4656-0000071c", "AMPUSERCID=4656") in new stack
- -- Executing [s@macro-user-callerid:8] Set("SIP/4656-0000071c", "CALLERID(all)="Andy Ortlieb" <4656>") in new stack
- -- Executing [s@macro-user-callerid:9] GotoIf("SIP/4656-0000071c", "1?continue") in new stack
- -- Goto (macro-user-callerid,s,18)
- -- Executing [s@macro-user-callerid:18] NoOp("SIP/4656-0000071c", "Using CallerID "Andy Ortlieb" <4656>") in new stack
- -- Executing [*19101@from-internal-original:2] NoOp("SIP/4656-0000071c", "Calling Out Route: dcptest") in new stack
- -- Executing [*19101@from-internal-original:3] Set("SIP/4656-0000071c", "MOHCLASS=default") in new stack
- -- Executing [*19101@from-internal-original:4] Set("SIP/4656-0000071c", "_NODEST=") in new stack
- -- Executing [*19101@from-internal-original:5] Macro("SIP/4656-0000071c", "record-enable,4656,OUT,") in new stack
- -- Executing [s@macro-record-enable:1] GotoIf("SIP/4656-0000071c", "1?check") in new stack
- -- Goto (macro-record-enable,s,4)
- -- Executing [s@macro-record-enable:4] ExecIf("SIP/4656-0000071c", "0?MacroExit()") in new stack
- -- Executing [s@macro-record-enable:5] GotoIf("SIP/4656-0000071c", "0?Group:OUT") in new stack
- -- Goto (macro-record-enable,s,15)
- -- Executing [s@macro-record-enable:15] GotoIf("SIP/4656-0000071c", "0?IN") in new stack
- -- Executing [s@macro-record-enable:16] ExecIf("SIP/4656-0000071c", "1?MacroExit()") in new stack
- -- Executing [*19101@from-internal-original:6] Macro("SIP/4656-0000071c", "dialout-trunk,8,101,") in new stack
- -- Executing [s@macro-dialout-trunk:1] Set("SIP/4656-0000071c", "DIAL_TRUNK=8") in new stack
- -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/4656-0000071c", "0?sub-pincheck,s,1") in new stack
- -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/4656-0000071c", "0?disabletrunk,1") in new stack
- -- Executing [s@macro-dialout-trunk:4] Set("SIP/4656-0000071c", "DIAL_NUMBER=101") in new stack
- -- Executing [s@macro-dialout-trunk:5] Set("SIP/4656-0000071c", "DIAL_TRUNK_OPTIONS=trT") in new stack
- -- Executing [s@macro-dialout-trunk:6] Set("SIP/4656-0000071c", "OUTBOUND_GROUP=OUT_8") in new stack
- -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/4656-0000071c", "1?nomax") in new stack
- -- Goto (macro-dialout-trunk,s,9)
- -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/4656-0000071c", "0?skipoutcid") in new stack
- -- Executing [s@macro-dialout-trunk:10] Set("SIP/4656-0000071c", "DIAL_TRUNK_OPTIONS=") in new stack
- -- Executing [s@macro-dialout-trunk:11] Macro("SIP/4656-0000071c", "outbound-callerid,8") in new stack
- -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/4656-0000071c", "0?Set(CALLERPRES()=)") in new stack
- -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/4656-0000071c", "0?Set(REALCALLERIDNUM=4656)") in new stack
- -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/4656-0000071c", "1?normcid") in new stack
- -- Goto (macro-outbound-callerid,s,6)
- -- Executing [s@macro-outbound-callerid:6] Set("SIP/4656-0000071c", "USEROUTCID=2242384656") in new stack
- -- Executing [s@macro-outbound-callerid:7] Set("SIP/4656-0000071c", "EMERGENCYCID=4146511878") in new stack
- -- Executing [s@macro-outbound-callerid:8] Set("SIP/4656-0000071c", "TRUNKOUTCID=") in new stack
- -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/4656-0000071c", "1?trunkcid") in new stack
- -- Goto (macro-outbound-callerid,s,12)
- -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/4656-0000071c", "0?Set(CALLERID(all)=)") in new stack
- -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/4656-0000071c", "1?Set(CALLERID(all)=2242384656)") in new stack
- -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/4656-0000071c", "0?Set(CALLERID(all)=)") in new stack
- -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/4656-0000071c", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
- -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/4656-0000071c", "0?sub-flp-8,s,1") in new stack
- -- Executing [s@macro-dialout-trunk:13] Set("SIP/4656-0000071c", "OUTNUM=101") in new stack
- -- Executing [s@macro-dialout-trunk:14] Set("SIP/4656-0000071c", "custom=SIP/DCP-test") in new stack
- -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/4656-0000071c", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
- -- Executing [s@macro-dialout-trunk:16] Macro("SIP/4656-0000071c", "dialout-trunk-predial-hook,") in new stack
- -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/4656-0000071c", "") in new stack
- -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/4656-0000071c", "0?bypass,1") in new stack
- -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/4656-0000071c", "0?customtrunk") in new stack
- -- Executing [s@macro-dialout-trunk:19] Dial("SIP/4656-0000071c", "SIP/DCP-test/101,300,") in new stack
- == Using SIP RTP TOS bits 184
- == Using SIP RTP CoS mark 5
- == Using SIP VRTP TOS bits 136
- == Using SIP VRTP CoS mark 6
- Audio is at 74.222.60.237 port 10064
- Video is at 74.222.60.237 port 10008
- Adding codec 0x4 (ulaw) to SDP
- Adding codec 0x2 (gsm) to SDP
- Adding codec 0x8 (alaw) to SDP
- Adding codec 0x10 (g726aal2) to SDP
- Adding codec 0x20 (adpcm) to SDP
- Adding codec 0x40 (slin) to SDP
- Adding codec 0x80 (lpc10) to SDP
- Adding codec 0x800 (g726) to SDP
- Adding codec 0x1000 (g722) to SDP
- Adding codec 0x8000 (slin16) to SDP
- Adding video codec 0x10000 (jpeg) to SDP
- Adding video codec 0x20000 (png) to SDP
- Adding video codec 0x40000 (h261) to SDP
- Adding video codec 0x80000 (h263) to SDP
- Adding video codec 0x100000 (h263p) to SDP
- Adding video codec 0x200000 (h264) to SDP
- Adding video codec 0x400000 (mpeg4) to SDP
- Adding non-codec 0x1 (telephone-event) to SDP
- Reliably Transmitting (NAT) to 64.105.229.19:5060:
- INVITE sip:101@64.105.229.19 SIP/2.0
- Via: SIP/2.0/UDP 74.222.60.237:5060;branch=z9hG4bK67dd29d0;rport
- Max-Forwards: 70
- From: "2242384656" <sip:2242384656@74.222.60.237>;tag=as78f5024d
- To: <sip:101@64.105.229.19>
- Contact: <sip:2242384656@74.222.60.237>
- Call-ID: 74f14dea4deaae6b2370d7b134384281@74.222.60.237
- CSeq: 102 INVITE
- User-Agent: Asterisk PBX 1.6.2.18
- Date: Wed, 05 Oct 2011 14:34:13 GMT
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces
- Content-Type: application/sdp
- Content-Length: 667
- v=0
- o=root 1381083937 1381083937 IN IP4 74.222.60.237
- s=Asterisk PBX 1.6.2.18
- c=IN IP4 74.222.60.237
- b=CT:1024
- t=0 0
- m=audio 10064 RTP/AVP 0 3 8 112 5 10 7 111 9 101
- a=rtpmap:0 PCMU/8000
- a=rtpmap:3 GSM/8000
- a=rtpmap:8 PCMA/8000
- a=rtpmap:112 AAL2-G726-32/8000
- a=rtpmap:5 DVI4/8000
- a=rtpmap:10 L16/8000
- a=rtpmap:7 LPC/8000
- a=rtpmap:111 G726-32/8000
- a=rtpmap:9 G722/8000
- a=rtpmap:101 telephone-event/8000
- a=fmtp:101 0-16
- a=ptime:20
- a=sendrecv
- m=video 10008 RTP/AVP 26 31 34 98 99 104
- a=rtpmap:26 JPEG/90000
- a=rtpmap:31 H261/90000
- a=rtpmap:34 H263/90000
- a=rtpmap:98 h263-1998/90000
- a=rtpmap:99 H264/90000
- a=rtpmap:104 MP4V-ES/90000
- a=sendrecv
- ---
- -- Called DCP-test/101
- -- SIP/DCP-test-0000071d is circuit-busy
- == Everyone is busy/congested at this time (1:0/1/0)
- <--- SIP read from UDP:64.105.229.19:5060 --->
- SIP/2.0 100 Trying
- Via: SIP/2.0/UDP 74.222.60.237:5060;branch=z9hG4bK67dd29d0;received=74.222.60.237;rport=5060
- From: "2242384656" <sip:2242384656@74.222.60.237>;tag=as78f5024d
- To: <sip:101@64.105.229.19>
- Call-ID: 74f14dea4deaae6b2370d7b134384281@74.222.60.237
- CSeq: 102 INVITE
- Server: Asterisk PBX 1.6.2.18
- Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
- Supported: replaces, timer
- Contact: <sip:101@64.105.229.19>
- Content-Length: 0
- <------------->
- --- (11 headers 0 lines) ---
- Really destroying SIP dialog '74f14dea4deaae6b2370d7b134384281@74.222.60.237' Method: INVITE
- -- Executing [s@macro-dialout-trunk:20] NoOp("SIP/4656-0000071c", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 1") in new stack
- -- Executing [s@macro-dialout-trunk:21] Goto("SIP/4656-0000071c", "s-CONGESTION,1") in new stack
- -- Goto (macro-dialout-trunk,s-CONGESTION,1)
- -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/4656-0000071c", "RC=1") in new stack
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