Hello!
There is production environment based on Asterisk 1.6 with VoIP-gateway SPA-400 connected and updated with firmware 1.1.2.2 version. Regional settiing are employed according to
https://supportforums.cisco.com/servlet/JiveServlet/downloadBody/9873-102-3-44705/SPA400Configuration_060109.pdf
It all works fine, except of transmitting of DTMF-signals while calling from cell-phones. It takes 2-3 times to press the button to make SPA-400 "identify" DTMF-signal. When using station phohe, the situation is not reproduce, all works the right way.
We performed an experiment:
1. Calling from station phone to our dial exchange, and recording (in *.wav format) call while pressing buttons from 0 to 9.
2. Open wav-file with special signal-analizing tool www.sigview.com
3. Scaling signal for number "2", for example.
4. So there is graph:
Duration = 3,255-3,239 = 0,016 sec
spectrum is:
6. then we perform a call from cell-phone, recording call while pressing 0-9 buttons.
7. Open wav-file in analizing tool
8. The picture is:
Duration = 0,012 sec
spectrum is:
- When we compare the amplitude of the signals in both cases, it is clear we have a problem with defining DTMF because of its low value, which cause a distortion.
Then, according to http://www.gaw.ru/html.cgi/txt/app/micros/msp430/slaae16.htm - The "2" is 697Hz and 1336Hz respectively. We have 687Hz and 1343Hz for stationary phone, and 687 and 1312 for cell Nokia.
- The "line busy" signal:
We tryed to change "DTFM power" parameter many times from -400 to 20, and other regional settings, no right result.
What shell we tune to make it transmitt DTMF-signals (in case of cell-phone) with the first attempt?