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<DIV dir=ltr align=left><SPAN class=941550613-07112006><FONT face=Arial
color=#0000ff size=2>I was wondering whether you have canreinvite=yes on those
phones, and that the audio between the phones is working, but not between the
Asterisk server and the phones - perhaps an Ethereal trace from your Hub might
help?</FONT></SPAN></DIV><BR>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of
</B>Mike<BR><B>Sent:</B> 07 November 2006 12:42<BR><B>To:</B> 'Asterisk Users
Mailing List - Non-Commercial Discussion'<BR><B>Subject:</B> [asterisk-users]
Problem: 2 second silence at the beginning of mostcalls<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial size=2>I have the
following setup in my test lab (which reflects very much my production
installation, just on a smaller scale)</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=System>Asterisk server
------------- Internet -------------- Home router (Linksys) ---------------Hub
----------------> Polycom 501 (Phone A)</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006><FONT
face=System> |------------------->
Polycom 501 (Phone B)</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial size=2>All calls go
through my asterisk server, even if its from one Polycom to the other. If I
dial from phone A to phone B, audio doesnt get passed for the first 1-2
seconds. I end up saying "hello? hello? hello?" and eventually I heard
something. It makes for a bad user experience.</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial size=2>What can be the
problem? I imagine the NAT isnt the problem, or there would be no audio at
all. My Asterisk is running 1.2.4, and my Polycom phones at running
bootrom 3.2.2 and SIP 2.0.1 (fairly recent).</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2>Mike</FONT></SPAN></DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=375062612-07112006></SPAN> </DIV>
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