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<DIV>I have a asterisk server running on site listening on a public ip.&nbsp; 
Tonight I attempted to connect a Cisco 7960 phone from my home location via sip 
but failed.&nbsp; My home network is simple, Cox cable connection hooked to a 
linksys&nbsp;wrt router.&nbsp; The firewall on the linksys router is disabled 
and I even setup dmz to the phones ip as a last resort.&nbsp; I removed the 
linksys router and plugged the phone directly into the cable modem and now the 
phone can connect fine and works.&nbsp; I pasted below the sip debug output, 
anybody know what's going on or have experience with this?</DIV>
<DIV>&nbsp;</DIV>
<DIV>------------ sip.conf ----------------</DIV>
<DIV>[general]<BR>context=default&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR>bindport=5060&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR>bindaddr=0.0.0.0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 
<BR>srvlookup=yes&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>[1002]<BR>username=1002<BR>secret=********<BR>type=friend<BR>host=dynamic<BR>allow=all<BR>context=default<BR>nat=yes<BR></DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>&nbsp;</DIV>
<DIV>---------- SIP DEBUG -------------</DIV>
<DIV>&lt;-- SIP read from 68.5.xxx.xxx:51065:<BR>INVITE sip:0@204.10.xxx.xxx 
SIP/2.0<BR>Via: SIP/2.0/UDP 192.168.1.102:5060;branch=z9hG4bK1accca5f<BR>From: 
"1002" 
&lt;sip:1002@204.10.xxx.xxx&gt;;tag=00115cd9d0370002128504be-71e03bcb<BR>To: 
&lt;sip:0@204.10.xxx.xxx&gt;<BR>Call-ID: <A 
href="mailto:00115cd9-d0370002-799a069f-51955597@192.168.1.102">00115cd9-d0370002-799a069f-51955597@192.168.1.102</A><BR>Max-Forwards: 
70<BR>CSeq: 101 INVITE<BR>User-Agent: Cisco-CP7960G/8.0<BR>Contact: 
&lt;sip:1002@192.168.1.102:5060;transport=udp&gt;<BR>Expires: 180<BR>Accept: 
application/sdp<BR>Allow: 
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE<BR>Supported: 
replaces,join,norefersub<BR>Content-Length: 274<BR>Content-Type: 
application/sdp<BR>Content-Disposition: session;handling=optional</DIV>
<DIV>&nbsp;</DIV>
<DIV>v=0<BR>o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102<BR>s=SIP Call<BR>t=0 
0<BR>m=audio 25584 RTP/AVP 0 8 18 101<BR>c=IN IP4 192.168.1.102<BR>a=rtpmap:0 
PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:18 G729/0<BR>a=fmtp:18 
annexb=no<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 
0-15<BR>a=sendrecv</DIV>
<DIV>&nbsp;</DIV>
<DIV>--- (16 headers 13 lines)---<BR>Using INVITE request as basis request - <A 
href="mailto:00115cd9-d0370002-799a069f-51955597@192.168.1.102">00115cd9-d0370002-799a069f-51955597@192.168.1.102</A><BR>Sending 
to 192.168.1.102 : 5060 (non-NAT)<BR>Reliably Transmitting (NAT) to 
68.5.xxx.xxx:51065:<BR>SIP/2.0 407 Proxy Authentication Required<BR>Via: 
SIP/2.0/UDP 
192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx<BR>From: "1002" 
&lt;sip:1002@204.10.xxx.xxx&gt;;tag=00115cd9d0370002128504be-71e03bcb<BR>To: 
&lt;sip:0@204.10.xxx.xxx&gt;;tag=as0ed772bf<BR>Call-ID: <A 
href="mailto:00115cd9-d0370002-799a069f-51955597@192.168.1.102">00115cd9-d0370002-799a069f-51955597@192.168.1.102</A><BR>CSeq: 
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, 
BYE, REFER, SUBSCRIBE, NOTIFY<BR>Contact: 
&lt;sip:0@204.10.xxx.xxx&gt;<BR>Proxy-Authenticate: Digest realm="asterisk", 
nonce="74f7630a"<BR>Content-Length: 0</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>---<BR>Scheduling destruction of call <A 
href="mailto:'00115cd9-d0370002-799a069f-51955597@192.168.1.102'">'00115cd9-d0370002-799a069f-51955597@192.168.1.102'</A> 
in 15000 ms<BR>Found user '1002'<BR>localhost*CLI&gt;<BR>&lt;-- SIP read from 
68.5.xxx.xxx:51065:<BR>INVITE sip:0@204.10.xxx.xxx SIP/2.0<BR>Via: SIP/2.0/UDP 
192.168.1.102:5060;branch=z9hG4bK1accca5f<BR>From: "1002" 
&lt;sip:1002@204.10.xxx.xxx&gt;;tag=00115cd9d0370002128504be-71e03bcb<BR>To: 
&lt;sip:0@204.10.xxx.xxx&gt;<BR>Call-ID: <A 
href="mailto:00115cd9-d0370002-799a069f-51955597@192.168.1.102">00115cd9-d0370002-799a069f-51955597@192.168.1.102</A><BR>Max-Forwards: 
70<BR>CSeq: 101 INVITE<BR>User-Agent: Cisco-CP7960G/8.0<BR>Contact: 
&lt;sip:1002@192.168.1.102:5060;transport=udp&gt;<BR>Expires: 180<BR>Accept: 
application/sdp<BR>Allow: 
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE<BR>Supported: 
replaces,join,norefersub<BR>Content-Length: 274<BR>Content-Type: 
application/sdp<BR>Content-Disposition: session;handling=optional</DIV>
<DIV>&nbsp;</DIV>
<DIV>v=0<BR>o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102<BR>s=SIP Call<BR>t=0 
0<BR>m=audio 25584 RTP/AVP 0 8 18 101<BR>c=IN IP4 192.168.1.102<BR>a=rtpmap:0 
PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:18 G729/0<BR>a=fmtp:18 
annexb=no<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 
0-15<BR>a=sendrecv</DIV>
<DIV>&nbsp;</DIV>
<DIV>--- (16 headers 13 lines)---<BR>Ignoring this INVITE 
request<BR>Retransmitting #1 (NAT) to 68.5.xxx.xxx:51065:<BR>SIP/2.0 407 Proxy 
Authentication Required<BR>Via: SIP/2.0/UDP 
192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx<BR>From: "1002" 
&lt;sip:1002@204.10.xxx.xxx&gt;;tag=00115cd9d0370002128504be-71e03bcb<BR>To: 
&lt;sip:0@204.10.xxx.xxx&gt;;tag=as0ed772bf<BR>Call-ID: <A 
href="mailto:00115cd9-d0370002-799a069f-51955597@192.168.1.102">00115cd9-d0370002-799a069f-51955597@192.168.1.102</A><BR>CSeq: 
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, 
BYE, REFER, SUBSCRIBE, NOTIFY<BR>Contact: 
&lt;sip:0@204.10.xxx.xxx&gt;<BR>Proxy-Authenticate: Digest realm="asterisk", 
nonce="74f7630a"<BR>Content-Length: 0</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>---<BR>localhost*CLI&gt;<BR>&lt;-- SIP read from 
68.5.xxx.xxx:51065:<BR>INVITE sip:0@204.10.xxx.xxx SIP/2.0<BR>Via: SIP/2.0/UDP 
192.168.1.102:5060;branch=z9hG4bK1accca5f<BR>From: "1002" 
&lt;sip:1002@204.10.xxx.xxx&gt;;tag=00115cd9d0370002128504be-71e03bcb<BR>To: 
&lt;sip:0@204.10.xxx.xxx&gt;<BR>Call-ID: <A 
href="mailto:00115cd9-d0370002-799a069f-51955597@192.168.1.102">00115cd9-d0370002-799a069f-51955597@192.168.1.102</A><BR>Max-Forwards: 
70<BR>CSeq: 101 INVITE<BR>User-Agent: Cisco-CP7960G/8.0<BR>Contact: 
&lt;sip:1002@192.168.1.102:5060;transport=udp&gt;<BR>Expires: 180<BR>Accept: 
application/sdp<BR>Allow: 
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE<BR>Supported: 
replaces,join,norefersub<BR>Content-Length: 274<BR>Content-Type: 
application/sdp<BR>Content-Disposition: session;handling=optional</DIV>
<DIV>&nbsp;</DIV>
<DIV>v=0<BR>o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102<BR>s=SIP Call<BR>t=0 
0<BR>m=audio 25584 RTP/AVP 0 8 18 101<BR>c=IN IP4 192.168.1.102<BR>a=rtpmap:0 
PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:18 G729/0<BR>a=fmtp:18 
annexb=no<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 
0-15<BR>a=sendrecv</DIV>
<DIV>&nbsp;</DIV>
<DIV>--- (16 headers 13 lines)---<BR>Ignoring this INVITE 
request<BR>Retransmitting #2 (NAT) to 68.5.xxx.xxx:51065:<BR>SIP/2.0 407 Proxy 
Authentication Required<BR>Via: SIP/2.0/UDP 
192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx<BR>From: "1002" 
&lt;sip:1002@204.10.xxx.xxx&gt;;tag=00115cd9d0370002128504be-71e03bcb<BR>To: 
&lt;sip:0@204.10.xxx.xxx&gt;;tag=as0ed772bf<BR>Call-ID: <A 
href="mailto:00115cd9-d0370002-799a069f-51955597@192.168.1.102">00115cd9-d0370002-799a069f-51955597@192.168.1.102</A><BR>CSeq: 
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, 
BYE, REFER, SUBSCRIBE, NOTIFY<BR>Contact: 
&lt;sip:0@204.10.xxx.xxx&gt;<BR>Proxy-Authenticate: Digest realm="asterisk", 
nonce="74f7630a"<BR>Content-Length: 0</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>---<BR>localhost*CLI&gt;<BR>&lt;-- SIP read from 
68.5.xxx.xxx:51065:<BR>INVITE sip:0@204.10.xxx.xxx SIP/2.0<BR>Via: SIP/2.0/UDP 
192.168.1.102:5060;branch=z9hG4bK1accca5f<BR>From: "1002" 
&lt;sip:1002@204.10.xxx.xxx&gt;;tag=00115cd9d0370002128504be-71e03bcb<BR>To: 
&lt;sip:0@204.10.xxx.xxx&gt;<BR>Call-ID: <A 
href="mailto:00115cd9-d0370002-799a069f-51955597@192.168.1.102">00115cd9-d0370002-799a069f-51955597@192.168.1.102</A><BR>Max-Forwards: 
70<BR>CSeq: 101 INVITE<BR>User-Agent: Cisco-CP7960G/8.0<BR>Contact: 
&lt;sip:1002@192.168.1.102:5060;transport=udp&gt;<BR>Expires: 180<BR>Accept: 
application/sdp<BR>Allow: 
ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE<BR>Supported: 
replaces,join,norefersub<BR>Content-Length: 274<BR>Content-Type: 
application/sdp<BR>Content-Disposition: session;handling=optional</DIV>
<DIV>&nbsp;</DIV>
<DIV>v=0<BR>o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102<BR>s=SIP Call<BR>t=0 
0<BR>m=audio 25584 RTP/AVP 0 8 18 101<BR>c=IN IP4 192.168.1.102<BR>a=rtpmap:0 
PCMU/8000<BR>a=rtpmap:8 PCMA/8000<BR>a=rtpmap:18 G729/0<BR>a=fmtp:18 
annexb=no<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 
0-15<BR>a=sendrecv</DIV>
<DIV>&nbsp;</DIV>
<DIV>--- (16 headers 13 lines)---<BR>Ignoring this INVITE 
request<BR>Retransmitting #3 (NAT) to 68.5.xxx.xxx:51065:<BR>SIP/2.0 407 Proxy 
Authentication Required<BR>Via: SIP/2.0/UDP 
192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx<BR>From: "1002" 
&lt;sip:1002@204.10.xxx.xxx&gt;;tag=00115cd9d0370002128504be-71e03bcb<BR>To: 
&lt;sip:0@204.10.xxx.xxx&gt;;tag=as0ed772bf<BR>Call-ID: <A 
href="mailto:00115cd9-d0370002-799a069f-51955597@192.168.1.102">00115cd9-d0370002-799a069f-51955597@192.168.1.102</A><BR>CSeq: 
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, 
BYE, REFER, SUBSCRIBE, NOTIFY<BR>Contact: 
&lt;sip:0@204.10.xxx.xxx&gt;<BR>Proxy-Authenticate: Digest realm="asterisk", 
nonce="74f7630a"<BR>Content-Length: 0</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>---<BR>Retransmitting #4 (NAT) to 68.5.xxx.xxx:51065:<BR>SIP/2.0 407 
Proxy Authentication Required<BR>Via: SIP/2.0/UDP 
192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx<BR>From: "1002" 
&lt;sip:1002@204.10.xxx.xxx&gt;;tag=00115cd9d0370002128504be-71e03bcb<BR>To: 
&lt;sip:0@204.10.xxx.xxx&gt;;tag=as0ed772bf<BR>Call-ID: <A 
href="mailto:00115cd9-d0370002-799a069f-51955597@192.168.1.102">00115cd9-d0370002-799a069f-51955597@192.168.1.102</A><BR>CSeq: 
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, 
BYE, REFER, SUBSCRIBE, NOTIFY<BR>Contact: 
&lt;sip:0@204.10.xxx.xxx&gt;<BR>Proxy-Authenticate: Digest realm="asterisk", 
nonce="74f7630a"<BR>Content-Length: 0</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>---<BR>Retransmitting #5 (NAT) to 68.5.xxx.xxx:51065:<BR>SIP/2.0 407 
Proxy Authentication Required<BR>Via: SIP/2.0/UDP 
192.168.1.102:5060;branch=z9hG4bK1accca5f;received=68.5.xxx.xxx<BR>From: "1002" 
&lt;sip:1002@204.10.xxx.xxx&gt;;tag=00115cd9d0370002128504be-71e03bcb<BR>To: 
&lt;sip:0@204.10.xxx.xxx&gt;;tag=as0ed772bf<BR>Call-ID: <A 
href="mailto:00115cd9-d0370002-799a069f-51955597@192.168.1.102">00115cd9-d0370002-799a069f-51955597@192.168.1.102</A><BR>CSeq: 
101 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, 
BYE, REFER, SUBSCRIBE, NOTIFY<BR>Contact: 
&lt;sip:0@204.10.xxx.xxx&gt;<BR>Proxy-Authenticate: Digest realm="asterisk", 
nonce="74f7630a"<BR>Content-Length: 0</DIV>
<DIV>&nbsp;</DIV>
<DIV><BR>---<BR>Destroying call <A 
href="mailto:'00115cd9-d0370002-799a069f-51955597@192.168.1.102'">'00115cd9-d0370002-799a069f-51955597@192.168.1.102'</A><BR></DIV>
<DIV><BR>-- <BR><BR>~Shaun</DIV></FONT></DIV></BODY></HTML>