What does "sip show peers" output ? Also set a timeout in millisec like qualify=200 instead of qualify=yes<br><br><div><span class="gmail_quote">On 14/06/07, <b class="gmail_sendername">randulo</b> <<a href="mailto:spamsucks2005@gmail.com">
spamsucks2005@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I totally puzzled by this situation. I have asterisk
1.4.4 behind NAT.<br>All SIP peers are working properly to place or receive calls.<br>Any SIP peer or friend whether NATted or not will become UNREACHABLE<br>if qualify=yes.<br><br>I have identical peers on the other asterisk
1.2.16 production server.<br>In fact, two of the phones (linksys 941 and Polycom ip500) are using<br>one line for each asterisk. The 1.2 one works normally, the 1.4 does<br>not.<br><br>The sip confgs from "sip show settings" are identical on the two servers.
<br><br>The sip.conf peer entries were moved over exactly.<br><br>Ports 5060 to 5065 are forwarded to the asterisk server.<br><br>Looking at sip debug, I notice a few differences:<br><br>REGISTER from phone:<br><br>"Authorization: Digest username="Poly", realm="asterisk",..."
<br><br>does not show on the 1.4 server.<br><br>Trying (sent by *):<br><br>"Supported: replaces"<br><br>The Via lines are the same (internal ip addresses) on both servers,<br>but there is a "Sending to 192.168..
." on the 1.2 message where there<br>is none on the 1.4.<br><br>What is "supported: replaces" ?<br><br>What config setting generates the "Authorization: Digest..." message ?<br>_______________________________________________
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