All pastes #554443 Raw Edit

trentster

public text v1 · immutable
·published 2007-06-10 04:18 UTC
rendered paste body
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
Content-Length: 0


--- (9 headers 0 lines) ---
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=z9hG4bK777cf286;rport
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as6f7ad0c6
To: <sip:6720@10.1.1.3:5060;line=gv8x1x75>
Contact: <sip:Unknown@209.212.101.56>
Call-ID: 2ea4724857b0e9ee0af63f64546275d6@209.212.101.56
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 10 Jun 2007 04:16:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
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-piqzjmmtfmte;rport
From: "6720" <sip:6720@pabx.itws.co.za>;tag=eix4lx9av6
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as7e67fa4c
Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
Content-Length: 0


--- (9 headers 0 lines) ---
Retransmitting #2 (NAT) to 220.238.243.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nqtgfff2ftkz;received=220.238.243.11;rport=5060
From: "6720" <sip:6720@pabx.itws.co.za>;tag=eix4lx9av6
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as7e67fa4c
Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0860121218@209.212.101.56>
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 10612 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: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK777cf286;rport=5060
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as6f7ad0c6
To: <sip:6720@10.1.1.3:5060;line=gv8x1x75>
Call-ID: 2ea4724857b0e9ee0af63f64546275d6@209.212.101.56
CSeq: 102 OPTIONS
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;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 '2ea4724857b0e9ee0af63f64546275d6@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-piqzjmmtfmte;rport
From: "6720" <sip:6720@pabx.itws.co.za>;tag=eix4lx9av6
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as7e67fa4c
Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
Content-Length: 0


--- (9 headers 0 lines) ---
Destroying call '683888ab6adac4b97a5dacb36e73bc0d@127.0.0.1'
Retransmitting #3 (NAT) to 220.238.243.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nqtgfff2ftkz;received=220.238.243.11;rport=5060
From: "6720" <sip:6720@pabx.itws.co.za>;tag=eix4lx9av6
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as7e67fa4c
Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0860121218@209.212.101.56>
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 10612 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-piqzjmmtfmte;rport
From: "6720" <sip:6720@pabx.itws.co.za>;tag=eix4lx9av6
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as7e67fa4c
Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
Content-Length: 0


--- (9 headers 0 lines) ---
    -- Playing 'auth-incorrect' (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-nqtgfff2ftkz;received=220.238.243.11;rport=5060
From: "6720" <sip:6720@pabx.itws.co.za>;tag=eix4lx9av6
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as7e67fa4c
Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0860121218@209.212.101.56>
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 10612 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-piqzjmmtfmte;rport
From: "6720" <sip:6720@pabx.itws.co.za>;tag=eix4lx9av6
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as7e67fa4c
Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
Content-Length: 0


--- (9 headers 0 lines) ---
12 headers, 0 lines
Reliably Transmitting (NAT) to 41.243.80.35:5060:
OPTIONS sip:5021@192.168.0.2:5060 SIP/2.0
Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK610df032;rport
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as35b091a1
To: <sip:5021@192.168.0.2:5060>
Contact: <sip:Unknown@209.212.101.56>
Call-ID: 297eee0836f9e908313ab1f471a99316@209.212.101.56
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 10 Jun 2007 04:16:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
pabx*CLI> 
<-- SIP read from 41.243.80.35:5060: 
SIP/2.0 200 OK
To: <sip:5021@192.168.0.2:5060>;tag=a9378c97a509c43fi0
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as35b091a1
Call-ID: 297eee0836f9e908313ab1f471a99316@209.212.101.56
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 209.212.101.56:5060;branch=z9hG4bK610df032
Server: Linksys/SPA3102-3.2.6(GWa)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura


--- (10 headers 0 lines) ---
Destroying call '297eee0836f9e908313ab1f471a99316@209.212.101.56'
    -- Playing 'auth-thankyou' (language 'en')
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-nqtgfff2ftkz;received=220.238.243.11;rport=5060
From: "6720" <sip:6720@pabx.itws.co.za>;tag=eix4lx9av6
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as7e67fa4c
Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0860121218@209.212.101.56>
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 10612 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 - - - -

---
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=z9hG4bK398bfb24;rport
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as0003ecd1
To: <sip:5030@192.168.1.101:2059;line=t0c378t2>
Contact: <sip:Unknown@209.212.101.56>
Call-ID: 353729526c40e4103a1e564068b08ade@209.212.101.56
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 10 Jun 2007 04:16:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
    -- 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|20070610-061618|1181448966.4004") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20070610-061618|1181448966.4004: 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
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-piqzjmmtfmte;rport
From: "6720" <sip:6720@pabx.itws.co.za>;tag=eix4lx9av6
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as7e67fa4c
Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
Content-Length: 0


--- (9 headers 0 lines) ---
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=z9hG4bK398bfb24;rport=5060
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as0003ecd1
To: <sip:5030@192.168.1.101:2059;line=t0c378t2>
Call-ID: 353729526c40e4103a1e564068b08ade@209.212.101.56
CSeq: 102 OPTIONS
Contact: <sip:5030@192.168.1.101:2059;line=t0c378t2>;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 '353729526c40e4103a1e564068b08ade@209.212.101.56'
    -- 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
    -- mISDN/1-u1210 is proceeding passing it to SIP/6720-0892aa10
    -- mISDN/1-u1210 is ringing
    -- mISDN/1-u1210 answered SIP/6720-0892aa10
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-nqtgfff2ftkz;received=220.238.243.11;rport=5060
From: "6720" <sip:6720@pabx.itws.co.za>;tag=eix4lx9av6
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as7e67fa4c
Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0860121218@209.212.101.56>
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 10612 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-piqzjmmtfmte;rport
From: "6720" <sip:6720@pabx.itws.co.za>;tag=eix4lx9av6
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as7e67fa4c
Call-ID: 3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316
CSeq: 2 ACK
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
Content-Length: 0


--- (9 headers 0 lines) ---
  == 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'
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=z9hG4bK317c7428;rport
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as440b7b7b
To: <sip:natrelay.deltathree.com>
Contact: <sip:Unknown@209.212.101.56>
Call-ID: 2e1860ef28e581f50eae63472924df48@209.212.101.56
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 10 Jun 2007 04:16:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '3c2a5b77ea60-k8g5xx9bw5oi@snom320-000413242316'
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=z9hG4bK317c7428;rport
To: <sip:natrelay.deltathree.com>
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as440b7b7b
Call-ID: 2e1860ef28e581f50eae63472924df48@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=z9hG4bK317c7428;rport
To: <sip:natrelay.deltathree.com>
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as440b7b7b
Call-ID: 2e1860ef28e581f50eae63472924df48@209.212.101.56
CSeq: 102 OPTIONS
Content-Length: 0


--- (7 headers 0 lines) ---
Destroying call '2e1860ef28e581f50eae63472924df48@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=z9hG4bK5bbde63e;rport
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as3dde12f1
To: <sip:natrelay.deltathree.com>
Contact: <sip:Unknown@209.212.101.56>
Call-ID: 76ee36d01c0a95652c27d4612c67e418@209.212.101.56
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 10 Jun 2007 04:16:27 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=z9hG4bK5bbde63e;rport
To: <sip:natrelay.deltathree.com>
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as3dde12f1
Call-ID: 76ee36d01c0a95652c27d4612c67e418@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=z9hG4bK5bbde63e;rport
To: <sip:natrelay.deltathree.com>
From: "Unknown" <sip:Unknown@209.212.101.56>;tag=as3dde12f1
Call-ID: 76ee36d01c0a95652c27d4612c67e418@209.212.101.56
CSeq: 102 OPTIONS
Content-Length: 0


--- (7 headers 0 lines) ---
Destroying call '76ee36d01c0a95652c27d4612c67e418@209.212.101.56'