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
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