[asterisk-users] Problem: 2 second silence at the beginning ofmostcalls

[Mike]

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.  
 
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.
 
I am trying to figure it out with Ethereal (first thing I did) but I'm not
sure what to look for.
 
Mike


  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve
Langstaff
Sent: November 7, 2006 8:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Problem: 2 second silence at the beginning
ofmostcalls


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 at lists.digium.com
[mailto:asterisk-users-bounces at 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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