[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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