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" ;tag=as7985a3fc To: Contact: 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" ;tag=as7985a3fc To: Contact: 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" ;tag=as7985a3fc To: Contact: 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" ;tag=as7985a3fc To: Contact: 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" ;tag=as7985a3fc To: Contact: 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