Rtp.c:500 ast_rtp_read: + probelm witch musiconhold

hi, could someone help me solve this problem?
i have this command in extension.conf
exten => _XX,1,dial(SIP/${EXTEN},50,m(vyzvaneni))

musiconhold.conf has this lines,


when i call for example extension number 10(it is an real extension), i get this in CLI

Executing Dial(“SIP/11-a425”, “SIP/10|50|m(vyzvaneni)”) in new stack
– Called 10
– Started music on hold, class ‘vyzvaneni’, on channel ‘SIP/11-a425’
– SIP/10-aa7f is ringing
Aug 8 00:25:30 WARNING[1973]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed
Aug 8 00:25:30 NOTICE[1973]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP:

could someone tell me where is problem?thanks you very much. bye

When you are talking on a normal analog phone (non-voip), you hear a quiet white noise type sound while the line is active. (look for it sometime, most people don’t notice it). This is usually a result of the long copper wiring runs to the house (IIRC).
In VoIP, there is no long copper runs to the house, thus some SIP phones will inject a similar noise artifically so the phone doesn’t sound ‘dead’.
It is saying that this feature is not completely supported by *,

so if you can, turn off comfort noise on your SIP phones :smile:

thanks a lot, now is everything ok:-)music is playing:Dthanks