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<DIV dir=ltr align=left><SPAN class=843221513-07112006><FONT face=Arial
color=#0000ff size=2>I _had_ canreinvite=yes, before I read your post. My
production environement though cannot handle reinvites (all phones are behind
different NATs, too messy). So I've set those to canreinvite=no.
</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=843221513-07112006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=843221513-07112006><FONT face=Arial
color=#0000ff size=2>Unfortunately, it's not making a difference. I still
get the 1-2 seconds silence at the beginning of my calls. My Asterisk
server is not behind a NAT, so in theory it should work flawlessly. Also,
the latency between my LAN and my Asterisk server is about 10ms, very
stable.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=843221513-07112006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=843221513-07112006><FONT face=Arial
color=#0000ff size=2>I am trying to figure it out with Ethereal (first thing I
did) but I'm not sure what to look for.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=843221513-07112006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=843221513-07112006><FONT face=Arial
color=#0000ff size=2>Mike</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>Steve
Langstaff<BR><B>Sent:</B> November 7, 2006 8:08 AM<BR><B>To:</B> Asterisk
Users Mailing List - Non-Commercial Discussion<BR><B>Subject:</B> RE:
[asterisk-users] Problem: 2 second silence at the beginning
ofmostcalls<BR></FONT><BR></DIV>
<DIV></DIV>
<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>
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size=2></FONT></SPAN> </DIV>
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size=2></FONT></SPAN> </DIV>
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