rendered paste bodyasterisk1*CLI> sip debug
SIP Debugging enabled
asterisk1*CLI>
<-- SIP read from 192.168.0.104:47681:
INVITE sip:7000@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-9b318f2b053afa50-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:2200@192.168.0.104:47681>
To: "7000"<sip:7000@192.168.0.103>
From: "Doogies Extension"<sip:2200@192.168.0.103>;tag=36272437
Call-ID: cf6b000fc22b4c24@dm95YWdlcg..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: eventlist
User-Agent: eyeBeamLicensed release 9999a stamp 25770
Content-Length: 436
v=0
o=- 211292324 2 IN IP4 192.168.0.104
s=<CounterPath eyeBeam 1.2>
c=IN IP4 192.168.0.104
t=0 0
m=audio 47682 RTP/AVP 107 100 106 6 0 105 8 3 4 5 101
a=alt:1 1 : sojasovc syysekfa 192.168.0.104 47682
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:04DEEEC8A2D7463BB9BADE665BF795BB
--- (13 headers 15 lines) ---
Using INVITE request as basis request - cf6b000fc22b4c24@dm95YWdlcg..
Sending to 192.168.0.104 : 47681 (NAT)
Reliably Transmitting (NAT) to 192.168.0.104:47681:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-9b318f2b053afa50-1--d87543-;received=192.168.0.104;rport=47681
From: "Doogies Extension"<sip:2200@192.168.0.103>;tag=36272437
To: "7000"<sip:7000@192.168.0.103>;tag=as6ee7be5a
Call-ID: cf6b000fc22b4c24@dm95YWdlcg..
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78e03a35"
Content-Length: 0
---
Scheduling destruction of call 'cf6b000fc22b4c24@dm95YWdlcg..' in 15000 ms
Found user '2200'
<-- SIP read from 192.168.0.104:47681:
ACK sip:7000@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-9b318f2b053afa50-1--d87543-;rport
To: "7000"<sip:7000@192.168.0.103>;tag=as6ee7be5a
From: "Doogies Extension"<sip:2200@192.168.0.103>;tag=36272437
Call-ID: cf6b000fc22b4c24@dm95YWdlcg..
CSeq: 1 ACK
Content-Length: 0
--- (7 headers 0 lines) ---
asterisk1*CLI>
<-- SIP read from 192.168.0.104:47681:
INVITE sip:7000@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-b16a0724d1177555-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:2200@192.168.0.104:47681>
To: "7000"<sip:7000@192.168.0.103>
From: "Doogies Extension"<sip:2200@192.168.0.103>;tag=36272437
Call-ID: cf6b000fc22b4c24@dm95YWdlcg..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="2200",realm="asterisk",nonce="78e03a35",uri="sip:7000@192.168.0.103",response="d127683acae9ddee834512ed118d7b15",algorithm=MD5
Supported: eventlist
User-Agent: eyeBeamLicensed release 9999a stamp 25770
Content-Length: 436
v=0
o=- 211292324 2 IN IP4 192.168.0.104
s=<CounterPath eyeBeam 1.2>
c=IN IP4 192.168.0.104
t=0 0
m=audio 47682 RTP/AVP 107 100 106 6 0 105 8 3 4 5 101
a=alt:1 1 : sojasovc syysekfa 192.168.0.104 47682
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:04DEEEC8A2D7463BB9BADE665BF795BB
--- (14 headers 15 lines) ---
Using INVITE request as basis request - cf6b000fc22b4c24@dm95YWdlcg..
Sending to 192.168.0.104 : 47681 (NAT)
Found user '2200'
Found RTP audio format 107
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 4
Found RTP audio format 5
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.104:47682
Found description format BV32
Found description format SPEEX
Found description format SPEEX-FEC
Found description format SPEEX-FEC
Found description format telephone-event
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x22f (g723|gsm|ulaw|alaw|adpcm|speex)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 7000 in from-internal (domain 192.168.0.103)
list_route: hop: <sip:2200@192.168.0.104:47681>
Transmitting (NAT) to 192.168.0.104:47681:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-b16a0724d1177555-1--d87543-;received=192.168.0.104;rport=47681
From: "Doogies Extension"<sip:2200@192.168.0.103>;tag=36272437
To: "7000"<sip:7000@192.168.0.103>
Call-ID: cf6b000fc22b4c24@dm95YWdlcg..
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7000@192.168.0.103>
Content-Length: 0
---
-- Executing Macro("SIP/2200-08757218", "exten-vm|7000|7000") in new stack
-- Executing Macro("SIP/2200-08757218", "user-callerid") in new stack
-- Executing NoOp("SIP/2200-08757218", "user-callerid: device 2200") in new stack
-- Executing GotoIf("SIP/2200-08757218", "0?report") in new stack
-- Executing GotoIf("SIP/2200-08757218", "0?start") in new stack
-- Executing Set("SIP/2200-08757218", "REALCALLERIDNUM=2200") in new stack
-- Executing NoOp("SIP/2200-08757218", "REALCALLERIDNUM is 2200") in new stack
-- Executing Set("SIP/2200-08757218", "AMPUSER=2200") in new stack
-- Executing Set("SIP/2200-08757218", "AMPUSERCIDNAME=Doogie PC") in new stack
-- Executing GotoIf("SIP/2200-08757218", "0?report") in new stack
-- Executing Set("SIP/2200-08757218", "CALLERID(all)=Doogie PC <2200>") in new stack
-- Executing Set("SIP/2200-08757218", "REALCALLERIDNUM=2200") in new stack
-- Executing NoOp("SIP/2200-08757218", "TTL: ARG1: 7000") in new stack
-- Executing GotoIf("SIP/2200-08757218", "0?continue") in new stack
-- Executing Set("SIP/2200-08757218", "_TTL=64") in new stack
-- Executing GotoIf("SIP/2200-08757218", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/2200-08757218", "Using CallerID "Doogie PC" <2200>") in new stack
-- Executing Set("SIP/2200-08757218", "FROMCONTEXT=exten-vm") in new stack
-- Executing Set("SIP/2200-08757218", "VMBOX=7000") in new stack
-- Executing Set("SIP/2200-08757218", "EXTTOCALL=7000") in new stack
-- Executing Set("SIP/2200-08757218", "CFUEXT=") in new stack
-- Executing Set("SIP/2200-08757218", "RT=17") in new stack
-- Executing Macro("SIP/2200-08757218", "record-enable|7000|IN") in new stack
-- Executing GotoIf("SIP/2200-08757218", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/2200-08757218", "recordingcheck|20070406-092448|1175865888.732") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20070406-092448|1175865888.732: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/2200-08757218", "No recording needed") in new stack
-- Executing Macro("SIP/2200-08757218", "dial|17|tr|7000") in new stack
-- Executing DeadAGI("SIP/2200-08757218", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: priority is 1
dialparties.agi: Caller ID name is 'Doogie PC' number is '2200'
dialparties.agi: Methodology of ring is 'none'
> dialparties.agi: USE_CONFIRMATION: 'FALSE'
> dialparties.agi: RINGGROUP_INDEX: ''
-- dialparties.agi: Added extension 7000 to extension map
-- dialparties.agi: Extension 7000 cf is disabled
-- dialparties.agi: Extension 7000 do not disturb is disabled
> dialparties.agi: extnum: 7000
> dialparties.agi: exthascw: 1
> dialparties.agi: exthascfb: 0
> dialparties.agi: extcfb:
> dialparties.agi: exthascfu: 0
> dialparties.agi: extcfu:
-- dialparties.agi: dbset CALLTRACE/7000 to 2200
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/2200-08757218", "SIP/7000|17|tr") in new stack
Destroying call '3e9d0e9a1944918352c8fffc15fab152@127.0.0.1'
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing GosubIf("SIP/2200-08757218", "0?docfu|1") in new stack
-- Executing NoOp("SIP/2200-08757218", "Voicemail is 7000") in new stack
-- Executing GotoIf("SIP/2200-08757218", "0?s-CHANUNAVAIL|1") in new stack
-- Executing NoOp("SIP/2200-08757218", "Sending to Voicemail box 7000") in new stack
-- Executing Macro("SIP/2200-08757218", "vm|7000|CHANUNAVAIL") in new stack
-- Executing Macro("SIP/2200-08757218", "user-callerid|SKIPTTL") in new stack
-- Executing NoOp("SIP/2200-08757218", "user-callerid: Doogie PC 2200") in new stack
-- Executing GotoIf("SIP/2200-08757218", "0?report") in new stack
-- Executing GotoIf("SIP/2200-08757218", "1?start") in new stack
-- Goto (macro-user-callerid,s,5)
-- Executing NoOp("SIP/2200-08757218", "REALCALLERIDNUM is 2200") in new stack
-- Executing Set("SIP/2200-08757218", "AMPUSER=2200") in new stack
-- Executing Set("SIP/2200-08757218", "AMPUSERCIDNAME=Doogie PC") in new stack
-- Executing GotoIf("SIP/2200-08757218", "0?report") in new stack
-- Executing Set("SIP/2200-08757218", "CALLERID(all)=Doogie PC <2200>") in new stack
-- Executing Set("SIP/2200-08757218", "REALCALLERIDNUM=2200") in new stack
-- Executing NoOp("SIP/2200-08757218", "TTL: 64 ARG1: SKIPTTL") in new stack
-- Executing GotoIf("SIP/2200-08757218", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/2200-08757218", "Using CallerID "Doogie PC" <2200>") in new stack
-- Executing Set("SIP/2200-08757218", "VMGAIN=") in new stack
-- Executing GotoIf("SIP/2200-08757218", "1?s-CHANUNAVAIL|1") in new stack
-- Goto (macro-vm,s-CHANUNAVAIL,1)
-- Executing Macro("SIP/2200-08757218", "get-vmcontext|7000") in new stack
-- Executing Set("SIP/2200-08757218", "VMCONTEXT=default") in new stack
-- Executing GotoIf("SIP/2200-08757218", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing NoOp("SIP/2200-08757218", "") in new stack
-- Executing VoiceMail("SIP/2200-08757218", "7000@default|u") in new stack
We're at 192.168.0.103 port 10286
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 192.168.0.104:47681:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-b16a0724d1177555-1--d87543-;received=192.168.0.104;rport=47681
From: "Doogies Extension"<sip:2200@192.168.0.103>;tag=36272437
To: "7000"<sip:7000@192.168.0.103>;tag=as5171fcf2
Call-ID: cf6b000fc22b4c24@dm95YWdlcg..
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7000@192.168.0.103>
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 3113 3113 IN IP4 192.168.0.103
s=session
c=IN IP4 192.168.0.103
t=0 0
m=audio 10286 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 - - - -
---
-- Playing 'vm-theperson' (language 'en')
asterisk1*CLI>
<-- SIP read from 192.168.0.104:47681:
ACK sip:7000@192.168.0.103 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.104:47681;branch=z9hG4bK-d87543-c55d581b561ce05c-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:2200@192.168.0.104:47681>
To: "7000"<sip:7000@192.168.0.103>;tag=as5171fcf2
From: "Doogies Extension"<sip:2200@192.168.0.103>;tag=36272437
Call-ID: cf6b000fc22b4c24@dm95YWdlcg..
CSeq: 2 ACK
Proxy-Authorization: Digest username="2200",realm="asterisk",nonce="78e03a35",uri="sip:7000@192.168.0.103",response="d127683acae9ddee834512ed118d7b15",algorithm=MD5
User-Agent: eyeBeamLicensed release 9999a stamp 25770
Content-Length: 0
--- (11 headers 0 lines) ---
12 headers, 0 lines
Reliably Transmitting (NAT) to 86.108.53.127:60987:
OPTIONS sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754 SIP/2.0
Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK6af2052d;rport
From: "Unknown" <sip:Unknown@64.231.189.60>;tag=as784a1432
To: <sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754>
Contact: <sip:Unknown@64.231.189.60>
Call-ID: 0f03d8a7506ac0e5721e56725265888f@64.231.189.60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 06 Apr 2007 13:24:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/0' (language 'en')
Retransmitting #1 (NAT) to 86.108.53.127:60987:
OPTIONS sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754 SIP/2.0
Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK6af2052d;rport
From: "Unknown" <sip:Unknown@64.231.189.60>;tag=as784a1432
To: <sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754>
Contact: <sip:Unknown@64.231.189.60>
Call-ID: 0f03d8a7506ac0e5721e56725265888f@64.231.189.60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 06 Apr 2007 13:24:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
-- Playing 'digits/0' (language 'en')
Retransmitting #2 (NAT) to 86.108.53.127:60987:
OPTIONS sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754 SIP/2.0
Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK6af2052d;rport
From: "Unknown" <sip:Unknown@64.231.189.60>;tag=as784a1432
To: <sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754>
Contact: <sip:Unknown@64.231.189.60>
Call-ID: 0f03d8a7506ac0e5721e56725265888f@64.231.189.60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 06 Apr 2007 13:24:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.112:5060:
OPTIONS sip:2000@192.168.0.112:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK6c2cb148;rport
From: "Unknown" <sip:Unknown@192.168.0.103>;tag=as5de0f7a0
To: <sip:2000@192.168.0.112:5060>
Contact: <sip:Unknown@192.168.0.103>
Call-ID: 41ceba507a4736af32c1f31351f5f7c9@192.168.0.103
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 06 Apr 2007 13:24:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.112:5061:
OPTIONS sip:2100@192.168.0.112:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1063e71a;rport
From: "Unknown" <sip:Unknown@192.168.0.103>;tag=as24dca3d4
To: <sip:2100@192.168.0.112:5061>
Contact: <sip:Unknown@192.168.0.103>
Call-ID: 5d8317305dfe9de12837ba260240206d@192.168.0.103
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 06 Apr 2007 13:24:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.0.112:5060:
SIP/2.0 200 OK
To: <sip:2000@192.168.0.112:5060>;tag=e2d5239726b10c7di0
From: "Unknown" <sip:Unknown@192.168.0.103>;tag=as5de0f7a0
Call-ID: 41ceba507a4736af32c1f31351f5f7c9@192.168.0.103
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK6c2cb148
Server: Sipura/SPA2000-3.1.5
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Accept-Language: en_us
--- (11 headers 0 lines) ---
Destroying call '41ceba507a4736af32c1f31351f5f7c9@192.168.0.103'
asterisk1*CLI>
<-- SIP read from 192.168.0.112:5061:
SIP/2.0 200 OK
To: <sip:2100@192.168.0.112:5061>;tag=75105ac74f83bf0di1
From: "Unknown" <sip:Unknown@192.168.0.103>;tag=as24dca3d4
Call-ID: 5d8317305dfe9de12837ba260240206d@192.168.0.103
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK1063e71a
Server: Sipura/SPA2000-3.1.5
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Accept-Language: en_us
--- (11 headers 0 lines) ---
Destroying call '5d8317305dfe9de12837ba260240206d@192.168.0.103'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 69.158.200.30:5060:
OPTIONS sip:mcstoronto.ca SIP/2.0
Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK44637480;rport
From: "Unknown" <sip:Unknown@64.231.189.60>;tag=as12c84306
To: <sip:mcstoronto.ca>
Contact: <sip:Unknown@64.231.189.60>
Call-ID: 4fb960cb34344d322244d05b2306eb9c@64.231.189.60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 06 Apr 2007 13:24:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 69.158.200.30:5060:
SIP/2.0 404 Not Found
From: "Unknown"<sip:Unknown@64.231.189.60>;tag=as12c84306
To: <sip:mcstoronto.ca>;tag=485685578
Call-ID: 4fb960cb34344d322244d05b2306eb9c@64.231.189.60
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 64.231.189.60:5060;rport=5060;branch=z9hG4bK44637480
contact: <sip:mcstoronto.ca:5060;maddr=69.158.200.30>
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,com.nortelnetworks.im.encryption
Content-Length: 0
--- (9 headers 0 lines) ---
Destroying call '4fb960cb34344d322244d05b2306eb9c@64.231.189.60'
-- Playing 'digits/0' (language 'en')
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.104:47681:
OPTIONS sip:2200@192.168.0.104:47681;rinstance=799a10aa2b7e2adc SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK3e24c4b8;rport
From: "Unknown" <sip:Unknown@192.168.0.103>;tag=as1af7228e
To: <sip:2200@192.168.0.104:47681;rinstance=799a10aa2b7e2adc>
Contact: <sip:Unknown@192.168.0.103>
Call-ID: 35f83474080a7e6400eab899525cb58b@192.168.0.103
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 06 Apr 2007 13:24:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
asterisk1*CLI>
<-- SIP read from 192.168.0.104:47681:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK3e24c4b8;rport=5060
Contact: <sip:192.168.0.104:47681>
To: <sip:2200@192.168.0.104:47681;rinstance=799a10aa2b7e2adc>;tag=8602e969
From: "Unknown"<sip:Unknown@192.168.0.103>;tag=as1af7228e
Call-ID: 35f83474080a7e6400eab899525cb58b@192.168.0.103
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: eventlist
User-Agent: eyeBeamLicensed release 9999a stamp 25770
Content-Length: 0
--- (13 headers 0 lines) ---
Destroying call '35f83474080a7e6400eab899525cb58b@192.168.0.103'
Retransmitting #3 (NAT) to 86.108.53.127:60987:
OPTIONS sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754 SIP/2.0
Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK6af2052d;rport
From: "Unknown" <sip:Unknown@64.231.189.60>;tag=as784a1432
To: <sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754>
Contact: <sip:Unknown@64.231.189.60>
Call-ID: 0f03d8a7506ac0e5721e56725265888f@64.231.189.60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 06 Apr 2007 13:24:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
-- Playing 'vm-isunavail' (language 'en')
Retransmitting #4 (NAT) to 86.108.53.127:60987:
OPTIONS sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754 SIP/2.0
Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK6af2052d;rport
From: "Unknown" <sip:Unknown@64.231.189.60>;tag=as784a1432
To: <sip:7000@86.108.53.127:60810;rinstance=6b7e627658434754>
Contact: <sip:Unknown@64.231.189.60>
Call-ID: 0f03d8a7506ac0e5721e56725265888f@64.231.189.60
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 06 Apr 2007 13:24:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '0f03d8a7506ac0e5721e56725265888f@64.231.189.60'
asterisk1*CLI>
<-- SIP read from 192.168.0.104:47681:
--- (0 headers 1 lines) ---
-- Playing 'vm-intro' (language 'en')
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 69.158.200.30:5060:
REGISTER sip:mcstoronto.ca SIP/2.0
Via: SIP/2.0/UDP 64.231.189.60:5060;branch=z9hG4bK70a65362;rport
From: <sip:dv008446@mcstoronto.ca>;tag=as17802292
To: <sip:dv008446@mcstoronto.ca>
Call-ID: 7bee30ab72d02a0b0a15efa933a4b701@127.0.0.1
CSeq: 109 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="dv008446", realm="Realm", algorithm=MD5, uri="sip:mcstoronto.ca", nonce="MTE3NTg2NTY4MTk1NjJjNGZlNmNhZjJkN2QxZjdmY2FlZTY2YTQ3ZmExNGU2", response="70d0611882545e7009f46bb06ad1663a", opaque="", qop=auth, cnonce="1bad84ac", nc=00000003
Expires: 120
Contact: <sip:4162258875@64.231.189.60>
Event: registration
Content-Length: 0
---
asterisk1*CLI> sip debug
<-- SIP read from 69.158.200.30:5060:
SIP/2.0 100 Trying
From: <sip:dv008446@mcstoronto.ca>;tag=as17802292
To: <sip:dv008446@mcstoronto.ca>
Call-ID: 7bee30ab72d02a0b0a15efa933a4b701@127.0.0.1
CSeq: 109 REGISTER
Via: SIP/2.0/UDP 64.231.189.60:5060;rport=5060;branch=z9hG4bK70a65362
Content-Length: 0
--- (7 headers 0 lines) ---
asterisk1*CLI> sip debug
<-- SIP read from 69.158.200.30:5060:
SIP/2.0 200 Registration Successful
From: "DOUGHAN TURK"<sip:dv008446@mcstoronto.ca>;tag=as17802292
To: <sip:dv008446@mcstoronto.ca>;tag=1231123177
Call-ID: 7bee30ab72d02a0b0a15efa933a4b701@127.0.0.1
CSeq: 109 REGISTER
Via: SIP/2.0/UDP 64.231.189.60:5060;rport=5060;branch=z9hG4bK70a65362
contact: <sip:4162258875@64.231.189.60>;expires=109
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,com.nortelnetworks.im.encryption
Content-Length: 0
--- (9 headers 0 lines) ---
Scheduling destruction of call '7bee30ab72d02a0b0a15efa933a4b701@127.0.0.1' in 32000 ms
asterisk1*CLI> sip no debug
SIP Debugging Disabled
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/default/7000/tmp/z5RN8c format: wav49, 0x86ca178
-- x=1, open writing: /var/spool/asterisk/voicemail/default/7000/tmp/z5RN8c format: wav, 0x86d30d8
-- User hung up
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/2200-08757218' in macro 'vm'
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/2200-08757218' in macro 'exten-vm'
== Spawn extension (macro-vm, s-CHANUNAVAIL, 2) exited non-zero on 'SIP/2200-08757218'