[asterisk-users] Audio going one way for a few seconds during thecall

[Jason Backshall]

Two reccomendations:

 

1)       Enable nat for the SIP channels of the phones in SIP.conf.

 

Or

 

2)       If all the remote phones are in the same location, an IPSEC tunnel
between the remote router, and your Asterisk machine.

 

Jason.

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan
Zakaria
Sent: Saturday, 23 June 2007 1:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Audio going one way for a few seconds during
thecall

 

Hi,

This question was posted earlier, but there was no satisfactory answer to
it. Afterwards I tried everything but to no avail.

The problem of audio going one way during the call for a few seconds is
still there. 

Its Asterisk 1.2.18 hosted Dell server with no NAT.
Phones connect remotely through a hi-speed Internet connection, they are
behind NAT on a D-Link router, UDP ports 5060, 10001-20000 are forwarded to
LAN,*, which means they are forwarded to all the IPs. 

How can I fix this problem.

-- 
Zeeshan A Zakaria 

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