rendered paste body<-- SIP read from 24.7.133.237:32767:
OPTIONS sip:38.103.144.10:5060 SIP/2.0
Via: SIP/2.0/UDP 24.7.133.237:32767;rport;branch=z9hG4bKc0a8006b0000001045c82f6300001e9200000a7f
Content-Length: 0
Call-ID: 7AA441D2-A834-44C7-B7F3-D4FB119514B6@192.168.0.107
CSeq: 856 OPTIONS
From: <sip:2134@38.103.144.10:5060>;tag=30468348116720
Max-Forwards: 70
To: <sip:38.103.144.10:5060>
--- (8 headers 0 lines) ---
Looking for s in from-sip-external (domain 38.103.144.10)
Transmitting (no NAT) to 24.7.133.237:32767:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 24.7.133.237:32767;branch=z9hG4bKc0a8006b0000001045c82f6300001e9200000a7f;received=24.7.133.237;rport=32767
From: <sip:2134@38.103.144.10:5060>;tag=30468348116720
To: <sip:38.103.144.10:5060>;tag=as46d58896
Call-ID: 7AA441D2-A834-44C7-B7F3-D4FB119514B6@192.168.0.107
CSeq: 856 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:38.103.144.10>
Accept: application/sdp
Content-Length: 0
---
Destroying call '7AA441D2-A834-44C7-B7F3-D4FB119514B6@192.168.0.107'
PP1*CLI>
<-- SIP read from 24.7.133.237:63464:
OPTIONS sip:38.103.144.10:5060 SIP/2.0
Via: SIP/2.0/UDP 24.7.133.237:63464;rport;branch=z9hG4bKc0a8006a0000001045c82f5100004e86000002e2
Content-Length: 0
Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106
CSeq: 206 OPTIONS
From: <sip:1234@38.103.144.10:5060>;tag=86210098413065
Max-Forwards: 70
To: <sip:38.103.144.10:5060>
--- (8 headers 0 lines) ---
Looking for s in from-sip-external (domain 38.103.144.10)
Transmitting (no NAT) to 24.7.133.237:63464:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 24.7.133.237:63464;branch=z9hG4bKc0a8006a0000001045c82f5100004e86000002e2;received=24.7.133.237;rport=63464
From: <sip:1234@38.103.144.10:5060>;tag=86210098413065
To: <sip:38.103.144.10:5060>;tag=as2bbc15f7
Call-ID: C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106
CSeq: 206 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:38.103.144.10>
Accept: application/sdp
Content-Length: 0
---
Destroying call 'C5106F3E-FA9B-4275-8628-37A9BB55E694@192.168.0.106'
PP1*CLI>
<-- SIP read from 204.147.183.18:5060:
INVITE sip:s@204.147.183.18:5060 SIP/2.0
Record-Route: <sip:204.147.183.18;ftag=a069d4400018ad2643bb06040e08b79f;lr=on>
Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bK4c8c.5947a85.0
Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK1301794d29e37a77c285feaae3246095;rport=1024
Max-Forwards: 16
From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=a069d4400018ad2643bb06040e08b79f
To: <sip:08918917@sip-ny1.stanaphone.com>
Call-ID: C07353DE-19FD11D4-A72D8CDF-8489C4E@66.114.240.12
CSeq: 200 INVITE
Contact: Anonymous <sip:204.147.183.55:1024>
Expires: 300
User-Agent: Sippy
cisco-GUID: 3228263589-436015572-2804583647-138976334
h323-conf-id: 3228263589-436015572-2804583647-138976334
Content-Length: 357
Content-Type: application/sdp
v=0
o=CiscoSystemsSIP-GW-UserAgent 4114 1455 IN IP4 204.147.183.18
s=SIP Call
i=oldc:c=IN IP4 66.114.240.12
t=0 0
m=audio 58924 RTP/AVP 0 18 3 101
c=IN IP4 204.147.183.18
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=direction:passive
a=nortpproxy:yes
--- (16 headers 15 lines) ---
Using INVITE request as basis request - C07353DE-19FD11D4-A72D8CDF-8489C4E@66.114.240.12
Sending to 204.147.183.18 : 5060 (non-NAT)
Found peer 'stanap'
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 204.147.183.18:58924
Found description format PCMU
Found description format G729
Found description format GSM
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x106 (gsm|ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for s in from-sip-external (domain 204.147.183.18)
list_route: hop: <sip:204.147.183.18;ftag=a069d4400018ad2643bb06040e08b79f;lr=on>
Transmitting (no NAT) to 204.147.183.18:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bK4c8c.5947a85.0;received=204.147.183.18
Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK1301794d29e37a77c285feaae3246095;rport=1024
From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=a069d4400018ad2643bb06040e08b79f
To: <sip:08918917@sip-ny1.stanaphone.com>
Call-ID: C07353DE-19FD11D4-A72D8CDF-8489C4E@66.114.240.12
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@38.103.144.10>
Content-Length: 0
---
-- Executing Ringing("SIP/08918917-b640e6b0", "") in new stack
Transmitting (no NAT) to 204.147.183.18:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bK4c8c.5947a85.0;received=204.147.183.18
Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK1301794d29e37a77c285feaae3246095;rport=1024
From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=a069d4400018ad2643bb06040e08b79f
To: <sip:08918917@sip-ny1.stanaphone.com>;tag=as650b83d7
Call-ID: C07353DE-19FD11D4-A72D8CDF-8489C4E@66.114.240.12
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@38.103.144.10>
Content-Length: 0
---
-- Executing GotoIf("SIP/08918917-b640e6b0", "1?from-trunk||1") in new stack
-- Goto (from-trunk,s,1)
-- Executing Set("SIP/08918917-b640e6b0", "FROM_DID=s") in new stack
-- Executing Goto("SIP/08918917-b640e6b0", "ext-local|1234|1") in new stack
-- Goto (ext-local,1234,1)
-- Executing Macro("SIP/08918917-b640e6b0", "exten-vm|1234|1234") in new stack
-- Executing Macro("SIP/08918917-b640e6b0", "user-callerid") in new stack
-- Executing NoOp("SIP/08918917-b640e6b0", "user-callerid: 9166067933") in new stack
-- Executing GotoIf("SIP/08918917-b640e6b0", "0?report") in new stack
-- Executing GotoIf("SIP/08918917-b640e6b0", "0?start") in new stack
-- Executing Set("SIP/08918917-b640e6b0", "REALCALLERIDNUM=9166067933") in new stack
-- Executing NoOp("SIP/08918917-b640e6b0", "REALCALLERIDNUM is 9166067933") in new stack
-- Executing Set("SIP/08918917-b640e6b0", "AMPUSER=") in new stack
-- Executing Set("SIP/08918917-b640e6b0", "AMPUSERCIDNAME=") in new stack
-- Executing GotoIf("SIP/08918917-b640e6b0", "1?report") in new stack
-- Goto (macro-user-callerid,s,11)
-- Executing NoOp("SIP/08918917-b640e6b0", "TTL: ARG1: 1234") in new stack
-- Executing GotoIf("SIP/08918917-b640e6b0", "0?continue") in new stack
-- Executing Set("SIP/08918917-b640e6b0", "_TTL=64") in new stack
-- Executing GotoIf("SIP/08918917-b640e6b0", "1?continue") in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp("SIP/08918917-b640e6b0", "Using CallerID "" <9166067933>") in new stack
-- Executing Set("SIP/08918917-b640e6b0", "FROMCONTEXT=exten-vm") in new stack
-- Executing Set("SIP/08918917-b640e6b0", "VMBOX=1234") in new stack
-- Executing Set("SIP/08918917-b640e6b0", "EXTTOCALL=1234") in new stack
-- Executing Set("SIP/08918917-b640e6b0", "CFUEXT=") in new stack
-- Executing Set("SIP/08918917-b640e6b0", "RT=55") in new stack
-- Executing Macro("SIP/08918917-b640e6b0", "record-enable|1234|IN") in new stack
-- Executing GotoIf("SIP/08918917-b640e6b0", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI("SIP/08918917-b640e6b0", "recordingcheck|20070206-103316|1170775996.40") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20070206-103316|1170775996.40: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/08918917-b640e6b0", "No recording needed") in new stack
-- Executing Macro("SIP/08918917-b640e6b0", "dial|55|trw|1234") in new stack
-- Executing DeadAGI("SIP/08918917-b640e6b0", "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 'unknown' number is '9166067933'
dialparties.agi: Methodology of ring is 'none'
> dialparties.agi: USE_CONFIRMATION: 'FALSE'
> dialparties.agi: RINGGROUP_INDEX: ''
-- dialparties.agi: Added extension 1234 to extension map
-- dialparties.agi: Extension 1234 cf is disabled
-- dialparties.agi: Extension 1234 do not disturb is disabled
> dialparties.agi: extnum: 1234
> dialparties.agi: exthascw: 1
> dialparties.agi: exthascfb: 0
> dialparties.agi: extcfb:
> dialparties.agi: exthascfu: 0
> dialparties.agi: extcfu:
-- dialparties.agi: dbset CALLTRACE/1234 to 9166067933
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/08918917-b640e6b0", "SIP/1234|55|trw") in new stack
We're at 38.103.144.10 port 17142
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (NAT) to 24.7.133.237:63464:
INVITE sip:1234@24.7.133.237:63464 SIP/2.0
Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK3dd64247;rport
From: "9166067933" <sip:9166067933@38.103.144.10>;tag=as3255c8d8
To: <sip:1234@24.7.133.237:63464>
Contact: <sip:9166067933@38.103.144.10>
Call-ID: 332ddd2f3ae3571b68d6e3ab7921a7ef@38.103.144.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 06 Feb 2007 15:33:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 240
v=0
o=root 4945 4945 IN IP4 38.103.144.10
s=session
c=IN IP4 38.103.144.10
t=0 0
m=audio 17142 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 1234
Transmitting (no NAT) to 204.147.183.18:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bK4c8c.5947a85.0;received=204.147.183.18
Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK1301794d29e37a77c285feaae3246095;rport=1024
From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=a069d4400018ad2643bb06040e08b79f
To: <sip:08918917@sip-ny1.stanaphone.com>;tag=as650b83d7
Call-ID: C07353DE-19FD11D4-A72D8CDF-8489C4E@66.114.240.12
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@38.103.144.10>
Content-Length: 0
---
PP1*CLI>
<-- SIP read from 24.7.133.237:63464:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 38.103.144.10:5060;rport=5060;received=38.103.144.10;branch=z9hG4bK3dd64247
Content-Length: 0
Call-ID: 332ddd2f3ae3571b68d6e3ab7921a7ef@38.103.144.10
CSeq: 102 INVITE
From: "9166067933"<sip:9166067933@38.103.144.10>;tag=as3255c8d8
Server: SJphone/1.60.289a (SJ Labs)
To: "unknown"<sip:1234@24.7.133.237:63464>;tag=8621074843830
--- (8 headers 0 lines) ---
PP1*CLI>
<-- SIP read from 24.7.133.237:63464:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 38.103.144.10:5060;rport=5060;received=38.103.144.10;branch=z9hG4bK3dd64247
Content-Length: 0
Contact: <sip:1234@24.7.133.237:63464>
Call-ID: 332ddd2f3ae3571b68d6e3ab7921a7ef@38.103.144.10
CSeq: 102 INVITE
From: "9166067933"<sip:9166067933@38.103.144.10>;tag=as3255c8d8
Server: SJphone/1.60.289a (SJ Labs)
To: "unknown"<sip:1234@24.7.133.237:63464>;tag=8621074843830
--- (9 headers 0 lines) ---
-- SIP/1234-081e8110 is ringing
PP1*CLI>
<-- SIP read from 24.7.133.237:63464:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 38.103.144.10:5060;rport=5060;received=38.103.144.10;branch=z9hG4bK3dd64247
Content-Length: 219
Contact: <sip:1234@24.7.133.237:63464>
Call-ID: 332ddd2f3ae3571b68d6e3ab7921a7ef@38.103.144.10
Content-Type: application/sdp
CSeq: 102 INVITE
From: "9166067933"<sip:9166067933@38.103.144.10>;tag=as3255c8d8
Server: SJphone/1.60.289a (SJ Labs)
To: "unknown"<sip:1234@24.7.133.237:63464>;tag=8621074843830
v=0
o=- 3379736023 3379736023 IN IP4 24.7.133.237
s=SJphone
c=IN IP4 24.7.133.237
t=0 0
a=direction:active
m=audio 49182 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
--- (10 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 24.7.133.237:49182
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:1234@24.7.133.237:63464>
set_destination: Parsing <sip:1234@24.7.133.237:63464> for address/port to send to
set_destination: set destination to 24.7.133.237, port 63464
Transmitting (NAT) to 24.7.133.237:63464:
ACK sip:1234@24.7.133.237:63464 SIP/2.0
Via: SIP/2.0/UDP 38.103.144.10:5060;branch=z9hG4bK64893096;rport
From: "9166067933" <sip:9166067933@38.103.144.10>;tag=as3255c8d8
To: <sip:1234@24.7.133.237:63464>;tag=8621074843830
Contact: <sip:9166067933@38.103.144.10>
Call-ID: 332ddd2f3ae3571b68d6e3ab7921a7ef@38.103.144.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/1234-081e8110 answered SIP/08918917-b640e6b0
We're at 38.103.144.10 port 18144
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 204.147.183.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.147.183.18;branch=z9hG4bK4c8c.5947a85.0;received=204.147.183.18
Via: SIP/2.0/UDP 204.147.183.55:1024;branch=z9hG4bK1301794d29e37a77c285feaae3246095;rport=1024
Record-Route: <sip:204.147.183.18;ftag=a069d4400018ad2643bb06040e08b79f;lr=on>
From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=a069d4400018ad2643bb06040e08b79f
To: <sip:08918917@sip-ny1.stanaphone.com>;tag=as650b83d7
Call-ID: C07353DE-19FD11D4-A72D8CDF-8489C4E@66.114.240.12
CSeq: 200 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@38.103.144.10>
Content-Type: application/sdp
Content-Length: 216
v=0
o=root 4945 4945 IN IP4 38.103.144.10
s=session
c=IN IP4 38.103.144.10
t=0 0
m=audio 18144 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
PP1*CLI>
<-- SIP read from 204.147.183.18:5060:
ACK sip:s@204.147.183.18:5060 SIP/2.0
Via: SIP/2.0/UDP 204.147.183.18;branch=0
Via: SIP/2.0/UDP 204.147.183.55:1024;rport=1024;branch=z9hG4bK2cc3e5643859418549c0530ea87ab971
Max-Forwards: 16
From: <sip:9166067933@sip-ny1.stanaphone.com>;tag=a069d4400018ad2643bb06040e08b79f
To: <sip:08918917@sip-ny1.stanaphone.com>;tag=as650b83d7
Call-ID: C07353DE-19FD11D4-A72D8CDF-8489C4E@66.114.240.12
CSeq: 200 ACK
Expires: 300
User-Agent: Sippy
--- (10 headers 0 lines) ---
PP1*CLI>
<-- SIP read from 24.7.133.237:32767:
REGISTER sip:38.103.144.10:5060 SIP/2.0
Via: SIP/2.0/UDP 24.7.133.237:32767;rport;branch=z9hG4bKc0a8006b0000001045c82f6f00004cee00000a81
Content-Length: 0
Contact: <sip:2134@24.7.133.237:32767>
Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107
CSeq: 291 REGISTER
From: <sip:2134@38.103.144.10:5060>;tag=3046947271721
Max-Forwards: 70
To: <sip:2134@38.103.144.10:5060>
User-Agent: SJphone/1.60.289a (SJ Labs)
--- (10 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 24.7.133.237 : 32767 (NAT)
Transmitting (NAT) to 24.7.133.237:32767:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 24.7.133.237:32767;branch=z9hG4bKc0a8006b0000001045c82f6f00004cee00000a81;received=24.7.133.237;rport=32767
From: <sip:2134@38.103.144.10:5060>;tag=3046947271721
To: <sip:2134@38.103.144.10:5060>
Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107
CSeq: 291 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:2134@38.103.144.10>
Content-Length: 0
--1*CLI>
Transmitting (NAT) to 24.7.133.237:32767:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 24.7.133.237:32767;branch=z9hG4bKc0a8006b0000001045c82f6f00004cee00000a81;received=24.7.133.237;rport=32767
From: <sip:2134@38.103.144.10:5060>;tag=3046947271721
To: <sip:2134@38.103.144.10:5060>;tag=as7cd9134c
Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107
CSeq: 291 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a7e46be"
Content-Length: 0
---
Scheduling destruction of call '854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107' in 15000 ms
PP1*CLI>
<-- SIP read from 24.7.133.237:32767:
REGISTER sip:38.103.144.10:5060 SIP/2.0
Via: SIP/2.0/UDP 24.7.133.237:32767;rport;branch=z9hG4bKc0a8006b0000001045c82f6f0000789800000a84
Content-Length: 0
Contact: <sip:2134@24.7.133.237:32767>
Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107
CSeq: 292 REGISTER
From: <sip:2134@38.103.144.10:5060>;tag=30469481717155
Max-Forwards: 70
To: <sip:2134@38.103.144.10:5060>
User-Agent: SJphone/1.60.289a (SJ Labs)
Authorization: Digest username="2134",realm="asterisk",nonce="7a7e46be",uri="sip:38.103.144.10:5060",response="038a338e25e22cc5d7fa58e95b4881d6",algorithm="MD5"
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 24.7.133.237 : 32767 (NAT)
Transmitting (NAT) to 24.7.133.237:32767:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 24.7.133.237:32767;branch=z9hG4bKc0a8006b0000001045c82f6f0000789800000a84;received=24.7.133.237;rport=32767
From: <sip:2134@38.103.144.10:5060>;tag=30469481717155
To: <sip:2134@38.103.144.10:5060>
Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107
CSeq: 292 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:2134@38.103.144.10>
Content-Length: 0
---*CLI>
Transmitting (NAT) to 24.7.133.237:32767:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 24.7.133.237:32767;branch=z9hG4bKc0a8006b0000001045c82f6f0000789800000a84;received=24.7.133.237;rport=32767
From: <sip:2134@38.103.144.10:5060>;tag=30469481717155
To: <sip:2134@38.103.144.10:5060>;tag=as7cd9134c
Call-ID: 854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107
CSeq: 292 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 120
Contact: <sip:2134@24.7.133.237:32767>;expires=120
Date: Tue, 06 Feb 2007 15:33:21 GMT
Content-Length: 0
---
Scheduling destruction of call '854F1D85-7F38-466B-B9CC-4109D7E4F808@192.168.0.107' in 15000 ms