Testing DTMF with Asterisk

SIP DTMF Detection Problems
If you are having problems sending DTMF digits and are using a SIP phone, make sure the dtmfmode they have set is the same on the phone and in Asterisk. Also make sure you are not sending both inband and out-of-band (rfc2833) tones.


DTMF in band or out of band
as i have already mentioned i am new in Voip ans SIP protocol and i would like to know what is the difference between sending DTMF sounds in band and DTMF out of band.please help because i could not understand from the sites i found when i googled it.

Sending the DTMF codes out-of-band (OOB) means that codec used by the audio cannot mangle the DTMF codes beyond recognition.

Some codec (G.729a) don't have the fidelity needed to maintain the dual frequencies of a single DTMF tone. Sending the DTMF OOB means the other side will recognize the DTMF even if it doesn't "sound" right.

Testing DTMF with Asterisk

The D option to the Dial command transmits DTMF tones, with a 'w' causing a pause:
 Dial(<num>@<peer>,180,D(w*w1w3)) 

Manually extract and post DTMF digits

Q: I'm trying to make asterisk detect some DTMF digits during a call and post them (can't use WaitExten or Features.conf).
A: I would suggest that you implement that in logger.c and configure a line to send logs to an HTTP POST (via logger.conf), with the pbx_substitute_variables_helper function, using the ${CURL()} function directly. You may need to "preload => func_curl.so" in modules.conf, but that will work well. Or use a simple log watcher:
 
tail -n0 -f /var/log/asterisk/debug | \
   grep 'DTMF digit: [0-9#*]' | \
   your_custum_filter