Quote:
Originally Posted by martin
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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I guess you were right. It seems to be a Codec problem. Now it's ok. Thks again.
