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- <--- SIP read from UDP:74.222.60.237:5060 --->
- 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
- <------------->
- --- (14 headers 28 lines) ---
- == Using SIP RTP TOS bits 184
- == Using SIP RTP CoS mark 5
- Sending to 74.222.60.237 : 5060 (NAT)
- Using INVITE request as basis request - 74f14dea4deaae6b2370d7b134384281@74.222.60.237
- Found peer 'ds-all' for '2242384656' from 74.222.60.237:5060
- Found RTP audio format 0
- Found RTP audio format 3
- Found RTP audio format 8
- Found RTP audio format 112
- Found RTP audio format 5
- Found RTP audio format 10
- Found RTP audio format 7
- Found RTP audio format 111
- Found RTP audio format 9
- Found RTP audio format 101
- Found audio description format PCMU for ID 0
- Found audio description format GSM for ID 3
- Found audio description format PCMA for ID 8
- Found audio description format AAL2-G726-32 for ID 112
- Found audio description format DVI4 for ID 5
- Found audio description format L16 for ID 10
- Found audio description format LPC for ID 7
- Found audio description format G726-32 for ID 111
- Found audio description format G722 for ID 9
- Found audio description format telephone-event for ID 101
- Found RTP video format 26
- Found RTP video format 31
- Found RTP video format 34
- Found RTP video format 98
- Found RTP video format 99
- Found RTP video format 104
- Found video description format H261 for ID 31
- Found video description format H263 for ID 34
- Found video description format h263-1998 for ID 98
- Found video description format H264 for ID 99
- Found video description format MP4V-ES for ID 104
- Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0x18fe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g726aal2|g722)/video=0x7d0000 (jpeg|h261|h263|h263p|h264|mpeg4)/text=0x0 (nothing), combined - 0x7d18fe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g726aal2|g722|jpeg|h261|h263|h263p|h264|mpeg4)
- Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
- Peer audio RTP is at port 74.222.60.237:10064
- Looking for 101 in from-trunk (domain 64.105.229.19)
- list_route: hop: <sip:2242384656@74.222.60.237>
- <--- Transmitting (NAT) to 74.222.60.237:5060 --->
- 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
- <------------>
- -- Executing [101@from-trunk:1] Macro("SIP/ds-all-0000002f", "exten-vm,101,101") in new stack
- -- Executing [s@macro-exten-vm:1] Macro("SIP/ds-all-0000002f", "user-callerid,") in new stack
- -- Executing [s@macro-user-callerid:1] Set("SIP/ds-all-0000002f", "AMPUSER=2242384656") in new stack
- -- Executing [s@macro-user-callerid:2] GotoIf("SIP/ds-all-0000002f", "0?report") in new stack
- -- Executing [s@macro-user-callerid:3] ExecIf("SIP/ds-all-0000002f", "1?Set(REALCALLERIDNUM=2242384656)") in new stack
- -- Executing [s@macro-user-callerid:4] Set("SIP/ds-all-0000002f", "AMPUSER=") in new stack
- -- Executing [s@macro-user-callerid:5] Set("SIP/ds-all-0000002f", "AMPUSERCIDNAME=") in new stack
- -- Executing [s@macro-user-callerid:6] GotoIf("SIP/ds-all-0000002f", "1?report") in new stack
- -- Goto (macro-user-callerid,s,9)
- -- Executing [s@macro-user-callerid:9] GotoIf("SIP/ds-all-0000002f", "0?continue") in new stack
- -- Executing [s@macro-user-callerid:10] Set("SIP/ds-all-0000002f", "__TTL=64") in new stack
- -- Executing [s@macro-user-callerid:11] GotoIf("SIP/ds-all-0000002f", "1?continue") in new stack
- -- Goto (macro-user-callerid,s,18)
- -- Executing [s@macro-user-callerid:18] NoOp("SIP/ds-all-0000002f", "Using CallerID "2242384656" <2242384656>") in new stack
- -- Executing [s@macro-exten-vm:2] Set("SIP/ds-all-0000002f", "RingGroupMethod=none") in new stack
- -- Executing [s@macro-exten-vm:3] Set("SIP/ds-all-0000002f", "VMBOX=101") in new stack
- -- Executing [s@macro-exten-vm:4] Set("SIP/ds-all-0000002f", "__EXTTOCALL=101") in new stack
- -- Executing [s@macro-exten-vm:5] Set("SIP/ds-all-0000002f", "CFUEXT=") in new stack
- -- Executing [s@macro-exten-vm:6] Set("SIP/ds-all-0000002f", "CFBEXT=") in new stack
- -- Executing [s@macro-exten-vm:7] Set("SIP/ds-all-0000002f", "RT=15") in new stack
- -- Executing [s@macro-exten-vm:8] Macro("SIP/ds-all-0000002f", "record-enable,101,IN") in new stack
- -- Executing [s@macro-record-enable:1] GotoIf("SIP/ds-all-0000002f", "1?check") in new stack
- -- Goto (macro-record-enable,s,4)
- -- Executing [s@macro-record-enable:4] ExecIf("SIP/ds-all-0000002f", "0?MacroExit()") in new stack
- -- Executing [s@macro-record-enable:5] GotoIf("SIP/ds-all-0000002f", "0?Group:OUT") in new stack
- -- Goto (macro-record-enable,s,15)
- -- Executing [s@macro-record-enable:15] GotoIf("SIP/ds-all-0000002f", "1?IN") in new stack
- -- Goto (macro-record-enable,s,20)
- -- Executing [s@macro-record-enable:20] ExecIf("SIP/ds-all-0000002f", "1?MacroExit()") in new stack
- -- Executing [s@macro-exten-vm:9] Macro("SIP/ds-all-0000002f", "dial-one,15,trTwW,101") in new stack
- -- Executing [s@macro-dial-one:1] Set("SIP/ds-all-0000002f", "DEXTEN=101") in new stack
- -- Executing [s@macro-dial-one:2] Set("SIP/ds-all-0000002f", "DIALSTATUS_CW=") in new stack
- -- Executing [s@macro-dial-one:3] GosubIf("SIP/ds-all-0000002f", "0?screen,1") in new stack
- -- Executing [s@macro-dial-one:4] GosubIf("SIP/ds-all-0000002f", "0?cf,1") in new stack
- -- Executing [s@macro-dial-one:5] GotoIf("SIP/ds-all-0000002f", "1?skip1") in new stack
- -- Goto (macro-dial-one,s,8)
- -- Executing [s@macro-dial-one:8] GotoIf("SIP/ds-all-0000002f", "0?nodial") in new stack
- -- Executing [s@macro-dial-one:9] GotoIf("SIP/ds-all-0000002f", "0?continue") in new stack
- -- Executing [s@macro-dial-one:10] Set("SIP/ds-all-0000002f", "EXTHASCW=ENABLED") in new stack
- -- Executing [s@macro-dial-one:11] GotoIf("SIP/ds-all-0000002f", "0?next1:cwinusebusy") in new stack
- -- Goto (macro-dial-one,s,23)
- -- Executing [s@macro-dial-one:23] GotoIf("SIP/ds-all-0000002f", "1?next3:continue") in new stack
- -- Goto (macro-dial-one,s,24)
- -- Executing [s@macro-dial-one:24] ExecIf("SIP/ds-all-0000002f", "0?Set(DIALSTATUS_CW=BUSY)") in new stack
- -- Executing [s@macro-dial-one:25] GotoIf("SIP/ds-all-0000002f", "0?nodial") in new stack
- -- Executing [s@macro-dial-one:26] GosubIf("SIP/ds-all-0000002f", "1?dstring,1:dlocal,1") in new stack
- -- Executing [dstring@macro-dial-one:1] Set("SIP/ds-all-0000002f", "DSTRING=") in new stack
- -- Executing [dstring@macro-dial-one:2] Set("SIP/ds-all-0000002f", "DEVICES=101") in new stack
- -- Executing [dstring@macro-dial-one:3] ExecIf("SIP/ds-all-0000002f", "0?Return()") in new stack
- -- Executing [dstring@macro-dial-one:4] Set("SIP/ds-all-0000002f", "LOOPCNT=1") in new stack
- -- Executing [dstring@macro-dial-one:5] Set("SIP/ds-all-0000002f", "ITER=1") in new stack
- -- Executing [dstring@macro-dial-one:6] Set("SIP/ds-all-0000002f", "THISDIAL=SIP/101") in new stack
- -- Executing [dstring@macro-dial-one:7] GosubIf("SIP/ds-all-0000002f", "1?zap2dahdi,1") in new stack
- -- Executing [zap2dahdi@macro-dial-one:1] ExecIf("SIP/ds-all-0000002f", "0?Return()") in new stack
- -- Executing [zap2dahdi@macro-dial-one:2] Set("SIP/ds-all-0000002f", "NEWDIAL=") in new stack
- -- Executing [zap2dahdi@macro-dial-one:3] Set("SIP/ds-all-0000002f", "LOOPCNT2=1") in new stack
- -- Executing [zap2dahdi@macro-dial-one:4] Set("SIP/ds-all-0000002f", "ITER2=1") in new stack
- -- Executing [zap2dahdi@macro-dial-one:5] Set("SIP/ds-all-0000002f", "THISPART2=SIP/101") in new stack
- -- Executing [zap2dahdi@macro-dial-one:6] ExecIf("SIP/ds-all-0000002f", "0?Set(THISPART2=DAHDI/101)") in new stack
- -- Executing [zap2dahdi@macro-dial-one:7] Set("SIP/ds-all-0000002f", "NEWDIAL=SIP/101&") in new stack
- -- Executing [zap2dahdi@macro-dial-one:8] Set("SIP/ds-all-0000002f", "ITER2=2") in new stack
- -- Executing [zap2dahdi@macro-dial-one:9] GotoIf("SIP/ds-all-0000002f", "0?begin2") in new stack
- -- Executing [zap2dahdi@macro-dial-one:10] Set("SIP/ds-all-0000002f", "THISDIAL=SIP/101") in new stack
- -- Executing [zap2dahdi@macro-dial-one:11] Return("SIP/ds-all-0000002f", "") in new stack
- -- Executing [dstring@macro-dial-one:8] Set("SIP/ds-all-0000002f", "DSTRING=SIP/101&") in new stack
- -- Executing [dstring@macro-dial-one:9] Set("SIP/ds-all-0000002f", "ITER=2") in new stack
- -- Executing [dstring@macro-dial-one:10] GotoIf("SIP/ds-all-0000002f", "0?begin") in new stack
- -- Executing [dstring@macro-dial-one:11] Set("SIP/ds-all-0000002f", "DSTRING=SIP/101") in new stack
- -- Executing [dstring@macro-dial-one:12] Return("SIP/ds-all-0000002f", "") in new stack
- -- Executing [s@macro-dial-one:27] GotoIf("SIP/ds-all-0000002f", "0?nodial") in new stack
- -- Executing [s@macro-dial-one:28] GotoIf("SIP/ds-all-0000002f", "1?skiptrace") in new stack
- -- Goto (macro-dial-one,s,30)
- -- Executing [s@macro-dial-one:30] Set("SIP/ds-all-0000002f", "D_OPTIONS=trTwW") in new stack
- -- Executing [s@macro-dial-one:31] ExecIf("SIP/ds-all-0000002f", "0?SIPAddHeader(Alert-Info: )") in new stack
- -- Executing [s@macro-dial-one:32] ExecIf("SIP/ds-all-0000002f", "0?SIPAddHeader()") in new stack
- -- Executing [s@macro-dial-one:33] ExecIf("SIP/ds-all-0000002f", "0?SetMusicOnHold()") in new stack
- -- Executing [s@macro-dial-one:34] GosubIf("SIP/ds-all-0000002f", "0?qwait,1") in new stack
- -- Executing [s@macro-dial-one:35] Set("SIP/ds-all-0000002f", "__CWIGNORE=") in new stack
- -- Executing [s@macro-dial-one:36] Set("SIP/ds-all-0000002f", "__KEEPCID=TRUE") in new stack
- -- Executing [s@macro-dial-one:37] Dial("SIP/ds-all-0000002f", "SIP/101,15,trTwW") in new stack
- == Using SIP RTP TOS bits 184
- == Using SIP RTP CoS mark 5
- -- Called 101
- <--- Transmitting (NAT) to 74.222.60.237:5060 --->
- SIP/2.0 180 Ringing
- 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>;tag=as3b88c1da
- 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
- <------------>
- -- SIP/101-00000030 is ringing
- <--- Transmitting (NAT) to 74.222.60.237:5060 --->
- SIP/2.0 180 Ringing
- 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>;tag=as3b88c1da
- 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
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