[Asterisk-Dev] SIP DTMF question

Peter Grace pgrace at fierymoon.com
Tue Aug 26 15:50:12 MST 2003


 
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Right, as of the latest x-lite/asterisk cvs versions,
rfc2833 is the good choice.

Pete

- -----Original Message-----
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] On Behalf Of
Eric Wieling
Sent: Tuesday, August 26, 2003 5:00 PM
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] SIP DTMF question


You cannot use inband unless you are using ulaw or alaw as
the codec.

On Tue, 2003-08-26 at 14:32, asterisk at thehowertons.net
wrote:
> I've been lurking for quite a while looking at all the
> SIP/DTMF  information passing through this forum.  My
> problem is after dialing a  number with the SIP client
> (XLite or SJPhone) no tones are recognize (sent?) to the
> far end.  For instance, I call the demo at digium and 
> try to select 2 for support and it does not select
> anything.  What do  I need to have in my setup?  Here is
> my current sip.conf setup for the  phone.
> 
> [phone1]
> type=friend
> host=dynamic
> username=ryan at internal.thehowertons.net
> secret=password
> dtmfmode=inband; Choices are inband, rfc2833, or info
> mailbox=9725 ;  Mailbox for message waiting indicator
> context=home
> callerid="Ryan V. Howerton" <9725>
> 
> Thanks for the help.
> 
> Ryan
> 
> 
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com 
> http://lists.digium.com/mailman/listinfo/asterisk-dev
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