I am having problem diagnosing a call problem. On both a Cisco phone
and a Linksys 942 I am only getting one side of the call when connected
over a WAN link or internet connection. I have set nat=yes and qualify
in sip.conf and the phone registers fine. I can hear the other end, but
they do not hear anything, no voice or dtmf. I found a tip about
changing the RTP rate from .03 to .02 on Sipura phones to match
Asterisk rate and did that. I also made sure the RTP range for the
phone and the server was set to 10000 thru 20000. These phones work
fine when on the same subnet as the server. The server shows the
following message:
NOTICE[24975]: rtp.c:330 process_rfc3389: Comfort noise support
incomplete in Asterisk (RFC 3389). Please turn off on client if
possible. Client IP:
192.168.10.56
But I have silence suppresion off on the linksys phone.
--
Bruce
Nortex Networks
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