SIP Trunk Between Asterisk Boxes
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,
Jerry
3 thoughts on - SIP Trunk Between Asterisk Boxes
“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.
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Joshua C. Colp Digium – A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW – Huntsville, AL 35806 – US
Check us out at: http://www.digium.com & http://www.asterisk.org
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I did not see anything printed when I pressed a key. I say the audio prints.
Jerry
That means either it was not negotiated or was not picked up by Asterisk.
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Joshua C. Colp Digium – A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW – Huntsville, AL 35806 – US
Check us out at: http://www.digium.com & http://www.asterisk.org
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