I updated the version of asterisk I was using and the problem seems to have been solved.
Thanks for the help -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Thursday, June 05, 2003 3:31 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] RTP codec error??? Sorry for my terminology assumptions. UA = User Agent, which is what the ATA-186 and 7960 are. Anything that normally is what the "end user" has on their desk or on their computer (in the case of a softphone) is considered a "UA". So, since you have both an ATA-186 and Cisco 7960, make the changes I describe below. If you don't understand what they are, take a look at the configuration guides for each piece of equipment, located on the Cisco website, or alternately use Google (which I find to be more useful at finding things than Cisco's terrible search interface.) JT >What is a UA? I am not using an ATA-186 or a Cisco 7960. The only >Asterisk related hardware that I am using is TDM 400P and X100P. > >-----Original Message----- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of John Todd >Sent: Wednesday, June 04, 2003 5:27 PM >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] RTP codec error??? > >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 >> >>_______________________________________________ >>Asterisk-Users mailing list >>[EMAIL PROTECTED] >>http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
