[Asterisk-Users] RTP codec error???

[Derek Beaumont]

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