[asterisk-users] SIP trunk between asterisk boxes

Joshua C. Colp jcolp at digium.com
Tue Jul 23 12:20:20 CDT 2019

On Tue, Jul 23, 2019, at 1:47 PM, Jerry Geis wrote:
> I have a sip trunk between two asterisk boxes. 
> I can call into the first box, hit 499 for example and the call goes to 
> the second box and answers as expected plays me audio message just fine 
> etc... My issue is that DTMF does not seem to be working.
> Both sides are set for:
> dtmfmode=RFC2833
> What might I look at as to why DTMF digits are not transferred?
> Thanks,

"rtp set debug on" will show the RTP traffic flowing, and thus the DTMF. The "dtmf" option to the logger can also be used to provide a log message when DTMF is received. This can be used to narrow it down.

Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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