[Asterisk-Users] RTP codec error???

[John Todd]

What kind of UA are you using?  ATA-186?  Cisco 7960?  If the former, 
set AudioMode: 0x00140014  and if the latter, set "enable_vad: "0"  "

Try that - it sets the Voice Auto Detect to "off".  I don't know if 
that will solve the problem, since it seems to relate to Comfort 
Noise Generation, but my phones no longer produce buckets of codec 
errors with those settings.

JT


>When I make a call using sip, I get the line
>NOTICE[327696]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec
>19 received
>Repeated many times on the console
>
>; SIP Configuration for Asterisk
>;
>[general]
>port = 5060                     ; Port to bind to
>;bindaddr = 0.0.0.0             ; Address to bind to
>context = outgoing              ; Default for incoming calls
>allow=gsm
>allow=ulaw
>allow=alaw
>
>
>[iconnect]
>type=friend
>username=********
>password=****
>host=sipauth.deltathree.com
>;host=213.137.73.178
>
>
>
>All I have been able to find about this topic is that 19 is supposed to
>be comfort noise (whatever that is)
>
>Any help is appreciated
>
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