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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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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