Hi, i'd like to know more about this issue, i'm always getting this message while in call with anyone from sip to zap or zap to sip. ast_rtp_read: Unknown RTP codec 72 received
here is my current setup: client side, x-lite, with the transmit silence to yes, using ulaw,alaw on asterisk server side: sip.conf contain allow=ulaw and allow=alaw dtmfmode=inband So i always get this anoying notice and i cannot find any doc about fixing it. I have try to put rfc2833 or info for dtmfmode, still giving this result. Plus for the one on the zap side (using a regular phone) , he is hearing me like crystal. Me, using a sip x-lite phone software, i am always hearing parasite. Thank in advance, i would really appreciate some help about this issue. Here is my email for the one who know some of the answer : [EMAIL PROTECTED] Sincerely JF _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
