asterisk users - Nov 2006 - Problem: 2 second silence at the beginning of most calls

I 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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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: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] 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 mostcalls
	
	
	I 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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