[Asterisk-Users] RTP codec error???

[Derek Beaumont]

I updated the version of asterisk I was using and the problem seems to
have been solved.

Thanks for the help



-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of John Todd
Sent: Thursday, June 05, 2003 3:31 PM
To: asterisk-users at lists.digium.com
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: asterisk-users-admin at lists.digium.com
>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of John Todd
>Sent: Wednesday, June 04, 2003 5:27 PM
>To: asterisk-users at lists.digium.com
>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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>
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