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<DIV><FONT face=Arial size=2>According to your console output it looks like
there is some major latency. What is the average ping time from your
asterisk machine to the polycom phone?</FONT></DIV>
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style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=astuserlist@gmail.com href="mailto:astuserlist@gmail.com">Rana
Dutt</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">Asterisk Users</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Sunday, July 16, 2006 6:51 PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [asterisk-users] Polycom phone
cycles between UNREACHABLE and REACHABLE</DIV>
<DIV><BR></DIV>
<DIV>I have a customer with a Polycom 501 phone behind a NAT. His phone
is connected to his Netgear router at home which in turn is connected to
his cable modem. The phone is set up to register with our remote Asterisk
server which is on a public, static IP address, with no NAT. </DIV>
<DIV> </DIV>
<DIV>If we set qualify=yes, our Asterisk console shows his extension becoming
UNREACHABLE for a minute, then REACHABLE for a minute, then UNREACHABLE again,
in an endless cycle. If we try to call the phone while it is UNREACHABLE, the
phone never rings and the call goes straight to voice mail. This is very
annoying. </DIV>
<DIV> </DIV>
<DIV>If we set qualify=no, then if we try to call the phone, the phone
sometimes does not ring at all, and we hear silence. The call eventually goes
to voice mail. This is equally annoying to the customer.</DIV>
<DIV> </DIV>
<DIV>What is the solution to this problem? We have other customers with
Polycom phones behind NAT, and they don't have this problem. Will we have
better luck if we replace the Polycom with a Linksys 942 phone? </DIV>
<DIV> </DIV>
<DIV>Here is some console output:</DIV>
<DIV> </DIV>
<DIV>Jul 16 21:44:24 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer
'280' is now UNREACHABLE! Last qualify: 174<BR>Jul 16 21:45:33
NOTICE[19981]: chan_sip.c:9697 handle_response_peerpoke: Peer '280' is now
REACHABLE! (3181ms / 5000ms) <BR>Jul 16 21:47:37 NOTICE[19981]:
chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last
qualify: 175</DIV>
<DIV> </DIV>
<DIV>Here is the way the phone is set up in sip.conf:</DIV>
<DIV> </DIV>
<DIV>[280]<BR>type=peer<BR>username=280<BR>secret=280<BR>host=dynamic<BR>dtmfmode=rfc2833<BR>callerid="John"
<280><BR>context=company_x<BR>mailbox=280<BR>nat=yes<BR>canreinvite=no<BR>qualify=5000<BR><BR>We
are using Asterisk 1.2.5 with standard .conf files. We are not using realtime
or databases. Any help would be highly appreciated. </DIV>
<DIV> </DIV>
<DIV>Rana Dutt</DIV>
<DIV>Softel Solutions</DIV>
<DIV><A href="mailto:rdutt@softelinc.com">rdutt@softelinc.com</A></DIV>
<DIV> </DIV>
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