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Unread 10-13-2006, 09:45 PM   #8
martin
 
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This is usually a Codec or NAT issue. If 2 end points can not agree on a Codec the call is dropped.

Likewise if the RTP stream can not be established due to your firewall the call will be lost.

First thing I would check would be Codec.

Also are you able to dial 600?

If you dial a SIP Broker PSTN access number then *010<your VoXaLot number> does your phone ring and provide full audio?
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