All pastes #414228 Raw Edit

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#414228 ·published 2007-03-28 22:20 UTC
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SIP Debugging Enabled for IP: 148.240.27.13:50012
12 headers, 0 lines
Reliably Transmitting (NAT) to 148.240.27.13:50012:
OPTIONS sip:5200001@192.168.0.229:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 72.232.225.42:5060;branch=z9hG4bK557d6356;rport
From: "Unknown" <sip:Unknown@72.232.225.42>;tag=as7985a3fc
To: <sip:5200001@192.168.0.229:5060;user=phone;transport=udp>
Contact: <sip:Unknown@72.232.225.42>
Call-ID: 52109c77409072b84b5baf257047ccc9@72.232.225.42
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 28 Mar 2007 23:15:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #1 (NAT) to 148.240.27.13:50012:
OPTIONS sip:5200001@192.168.0.229:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 72.232.225.42:5060;branch=z9hG4bK557d6356;rport
From: "Unknown" <sip:Unknown@72.232.225.42>;tag=as7985a3fc
To: <sip:5200001@192.168.0.229:5060;user=phone;transport=udp>
Contact: <sip:Unknown@72.232.225.42>
Call-ID: 52109c77409072b84b5baf257047ccc9@72.232.225.42
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 28 Mar 2007 23:15:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #2 (NAT) to 148.240.27.13:50012:
OPTIONS sip:5200001@192.168.0.229:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 72.232.225.42:5060;branch=z9hG4bK557d6356;rport
From: "Unknown" <sip:Unknown@72.232.225.42>;tag=as7985a3fc
To: <sip:5200001@192.168.0.229:5060;user=phone;transport=udp>
Contact: <sip:Unknown@72.232.225.42>
Call-ID: 52109c77409072b84b5baf257047ccc9@72.232.225.42
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 28 Mar 2007 23:15:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #3 (NAT) to 148.240.27.13:50012:
OPTIONS sip:5200001@192.168.0.229:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 72.232.225.42:5060;branch=z9hG4bK557d6356;rport
From: "Unknown" <sip:Unknown@72.232.225.42>;tag=as7985a3fc
To: <sip:5200001@192.168.0.229:5060;user=phone;transport=udp>
Contact: <sip:Unknown@72.232.225.42>
Call-ID: 52109c77409072b84b5baf257047ccc9@72.232.225.42
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 28 Mar 2007 23:15:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #4 (NAT) to 148.240.27.13:50012:
OPTIONS sip:5200001@192.168.0.229:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 72.232.225.42:5060;branch=z9hG4bK557d6356;rport
From: "Unknown" <sip:Unknown@72.232.225.42>;tag=as7985a3fc
To: <sip:5200001@192.168.0.229:5060;user=phone;transport=udp>
Contact: <sip:Unknown@72.232.225.42>
Call-ID: 52109c77409072b84b5baf257047ccc9@72.232.225.42
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 28 Mar 2007 23:15:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '52109c77409072b84b5baf257047ccc9@72.232.225.42'


SIP.CONF

[general]                       
context=default                 ; Default context for incoming calls
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)   
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
nat=yes                         ; Global NAT settings  (Affects all peers and users)
rtcachefriends=yes