i have a little problem to understand this warning message, it’s annoying
and it cause a lot of spurious in the log files.
Im working with this scenario:
a sip outgoing trunk (Trunk-out) that support only ulaw and all calls are
always routed to this.
a list of sip UAs that potentially can use any codec apart g729/g723.
I setup the asterisk to do as mediaproxy so directmedia=no and
canreinvite=no. I want the rtp traffic goes through asterisk.
I can reproduce the waring message below when a peer uses a different codec
when the UA send out a number to route through the trunk, the warning
message is not displayed initially, but only when the ring tone starts,
then the mesage appears repeatedly and stops when the called peer answers,
and the call is bridged successfully without problems.
Im using asterisk 1.8.0
== Using SIP RTP CoS mark 5