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?
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: 07 November 2006 12:42
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Problem: 2 second silence at the beginning of mostcallsI have the following setup in my test lab (which reflects very much my production installation, just on a smaller scale)Asterisk server ------------- Internet -------------- Home router (Linksys) ---------------Hub ----------------> Polycom 501 (Phone A)|-------------------> Polycom 501 (Phone B)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.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).Mike
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