I hav been trying to debug this for a while and I am drawing a blank. I
have an Asterisk PBX with a Adtran 750 and a Digium T1 card.
Sip to Sip calls work great.
Incoming PSTN calls through the CB have one way to audio when sent to a
Sip phone on the local network. What could cause that?
host*CLI> show version
Asterisk 1.2.5 built by root @ host.dyndns.org on a i686 running Linux
on 2006-03-15 19:21:51 UTC
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Chris Mason
NetConcepts
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Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271
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