[Asterisk-Users] 499 Error on X-lite / asterisk setup

David John Walsh davidjohnwalsh at gmail.com
Mon Apr 11 13:03:41 MST 2005


Ah ha!

Thats is Robert, you are a genius!!!

Actually that gives a larger issue - it doesn't take a lot for a user
to click the codecs off and then its a call to the help desk.

I am not intending to use xlite in production, but it does beg the
question can it be forced??

On Apr 11, 2005 7:21 PM, Robert Keller <rkeller at ferndale.wednet.edu> wrote:
>  David, do you have all the codec's enabled:
>  
>  I had that problem until I highlighted all of them.  I doubt all are
> needed, but that helped me.
>  
>  Robert.
>  
>  
>  > From: David John Walsh <davidjohnwalsh at gmail.com>
>  > Reply-To: David John Walsh <davidjohnwalsh at gmail.com>, Asterisk Users
> Mailing 
>  > List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
>  > Date: Mon, 11 Apr 2005 18:34:46 +0100
>  > To: asterisk-users at lists.digium.com
>  > Subject: [Asterisk-Users] 499 Error on X-lite / asterisk setup
>  > 
>  > I have a fairly simple asterisk setup - A at H 0.8 in SIP.conf:
>  > 
>  > Extentions 200 - 204 - username, password, callerid all same as extension
>  > 
>  > Extensions.conf - default build from A at H 0.8
>  > 
>  > In x-lite all spaces are either the IP address of the asterisk box or
>  > the extension number.
>  > 
>  > On loading of x-lite, asterisk pipes up that the extension is seen
>  > 
>  > dialling anything from x-lite gets to asterisk (seen with sip debug)
>  > however nothing comes up in the console (verbose >4) and 2 seconds
>  > later x-lite returns an error of "499 Not Acceptable Here"
>  > 
>  > In the console I get :
>  > 
>  > SIP/2.0 488 Not Acceptable Here
>  > Via: SIP/2.0/UDP 
>  >
> 172.16.0.32:5060;branch=z9hG4bKAD3D66F2AAAF11D9A5AF000A95D3F194
>  > From: 200 <sip:200 at 172.16.1.5>;tag=1211608254
>  > To: <sip:201 at 172.16.1.5>;tag=as2a712e55
>  > Call-ID:
> ABC2A5CE-AAAF-11D9-A5AF-000A95D3F194 at 172.16.0.32
>  > CSeq: 16528 INVITE
>  > User-Agent: Asterisk PBX
>  > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>  > Contact: <sip:201 at 172.16.1.5>
>  > Content-Length: 0
>  > 
>  > The AAH was a clean install from the ISO on known to be good hardware,
>  > and its nothing I haven't done before
>  > 
>  > Have I missed something?
>  > 
>  > David
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