rendered paste body
--- (10 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.110 : 1000 (NAT)
Transmitting (NAT) to 192.168.1.110:1000:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.1.110:1000;branch=z9hG4bKc0a8016e0000001045518ec7000039 4e000000d5;received=192.168.1.110;rport=1000
From: <sip:aao_pwner@192.168.1.109:5060>;tag=1317104624552
To: <sip:aao_pwner@192.168.1.109:5060>;tag=as7936160c
Call-ID: 53A3BAF2-608A-4345-BC21-644D98D20857@192.168.1.110
CSeq: 28 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:aao_pwner@192.168.1.109>
Content-Length: 0
---
Scheduling destruction of call '53A3BAF2-608A-4345-BC21-644D98D20857@192.168.1.1 10' in 15000 ms
asterisk1*CLI> sip debug
SIP Debugging re-enabled
asterisk1*CLI>
<-- SIP read from 192.168.1.110:1000:
OPTIONS sip:192.168.1.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:1000;rport;branch=z9hG4bKc0a8016e0000001045518eca 00002adb000000d7
Content-Length: 0
Call-ID: 0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110
CSeq: 80 OPTIONS
From: <sip:aao_pwner@192.168.1.109:5060>;tag=1317439013277
Max-Forwards: 70
To: <sip:192.168.1.109:5060>
--- (8 headers 0 lines)---
Looking for s in from-sip-external (domain 192.168.1.109)
Transmitting (no NAT) to 192.168.1.110:1000:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:1000;rport;branch=z9hG4bKc0a8016e0000001045518eca 00002adb000000d7;received=192.168.1.110
From: <sip:aao_pwner@192.168.1.109:5060>;tag=1317439013277
To: <sip:192.168.1.109:5060>;tag=as64bc43e3
Call-ID: 0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110
CSeq: 80 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.1.109>
Accept: application/sdp
Content-Length: 0
---
Destroying call '0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110'
Destroying call '53A3BAF2-608A-4345-BC21-644D98D20857@192.168.1.110'
asterisk1*CLI>
<-- SIP read from 192.168.1.110:1000:
OPTIONS sip:192.168.1.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:1000;rport;branch=z9hG4bKc0a8016e0000001045518ede00003f46000000d9
Content-Length: 0
Call-ID: 0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110
CSeq: 81 OPTIONS
From: <sip:aao_pwner@192.168.1.109:5060>;tag=1319439024851
Max-Forwards: 70
To: <sip:192.168.1.109:5060>
--- (8 headers 0 lines)---
Looking for s in from-sip-external (domain 192.168.1.109)
Transmitting (no NAT) to 192.168.1.110:1000:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:1000;rport;branch=z9hG4bKc0a8016e0000001045518ede00003f46000000d9;received=192.168.1.110
From: <sip:aao_pwner@192.168.1.109:5060>;tag=1319439024851
To: <sip:192.168.1.109:5060>;tag=as3358b453
Call-ID: 0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110
CSeq: 81 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.1.109>
Accept: application/sdp
Content-Length: 0
---
Destroying call '0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110'
asterisk1*CLI>
<-- SIP read from 192.168.1.110:1000:
OPTIONS sip:192.168.1.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:1000;rport;branch=z9hG4bKc0a8016e0000001045518ef200007bc5000000db
Content-Length: 0
Call-ID: 0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110
CSeq: 82 OPTIONS
From: <sip:aao_pwner@192.168.1.109:5060>;tag=1321439027341
Max-Forwards: 70
To: <sip:192.168.1.109:5060>
--- (8 headers 0 lines)---
Looking for s in from-sip-external (domain 192.168.1.109)
Transmitting (no NAT) to 192.168.1.110:1000:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:1000;rport;branch=z9hG4bKc0a8016e0000001045518ef200007bc5000000db;received=192.168.1.110
From: <sip:aao_pwner@192.168.1.109:5060>;tag=1321439027341
To: <sip:192.168.1.109:5060>;tag=as6bd48a1c
Call-ID: 0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110
CSeq: 82 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.1.109>
Accept: application/sdp
Content-Length: 0
---
Destroying call '0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110'
asterisk1*CLI>
<-- SIP read from 192.168.1.110:1000:
REGISTER sip:192.168.1.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:1000;rport;branch=z9hG4bKc0a8016e0000001045518f0300005568000000dd
Content-Length: 0
Contact: <sip:aao_pwner@192.168.1.110:1000>
Call-ID: 53A3BAF2-608A-4345-BC21-644D98D20857@192.168.1.110
CSeq: 29 REGISTER
From: <sip:aao_pwner@192.168.1.109:5060>;tag=132310467266
Max-Forwards: 70
To: <sip:aao_pwner@192.168.1.109:5060>
User-Agent: SJphone/1.60.289a (SJ Labs)
--- (10 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.1.110 : 1000 (NAT)
Transmitting (NAT) to 192.168.1.110:1000:
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.1.110:1000;branch=z9hG4bKc0a8016e0000001045518f0300005568000000dd;received=192.168.1.110;rport=1000
From: <sip:aao_pwner@192.168.1.109:5060>;tag=132310467266
To: <sip:aao_pwner@192.168.1.109:5060>;tag=as293db6af
Call-ID: 53A3BAF2-608A-4345-BC21-644D98D20857@192.168.1.110
CSeq: 29 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:aao_pwner@192.168.1.109>
Content-Length: 0
---
Scheduling destruction of call '53A3BAF2-608A-4345-BC21-644D98D20857@192.168.1.110' in 15000 ms
asterisk1*CLI>
<-- SIP read from 192.168.1.110:1000:
OPTIONS sip:192.168.1.109:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.110:1000;rport;branch=z9hG4bKc0a8016e0000001045518f06000000f1000000df
Content-Length: 0
Call-ID: 0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110
CSeq: 83 OPTIONS
From: <sip:aao_pwner@192.168.1.109:5060>;tag=1323439022898
Max-Forwards: 70
To: <sip:192.168.1.109:5060>
--- (8 headers 0 lines)---
Looking for s in from-sip-external (domain 192.168.1.109)
Transmitting (no NAT) to 192.168.1.110:1000:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.110:1000;rport;branch=z9hG4bKc0a8016e0000001045518f06000000f1000000df;received=192.168.1.110
From: <sip:aao_pwner@192.168.1.109:5060>;tag=1323439022898
To: <sip:192.168.1.109:5060>;tag=as4de731b5
Call-ID: 0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110
CSeq: 83 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:192.168.1.109>
Accept: application/sdp
Content-Length: 0
---
Destroying call '0DA4C9FB-79F5-4874-998F-A3C0B3ED763C@192.168.1.110'