I am setting up a simple SIP/analog phone line gateway.
The SIP phones used to work, I could leave check voicemails, call other users and do echo test etc
Now I installed a TDM400P (1 FXO)
Had to re-compile asterisk in order to get it working
Had to get my soundcard working since suddenly I got complaints about a missing /dev/dsp device.
So I installed the alsa drivers for my (82801EB AC'97) onboard soundcard
Sound works now on my system
I can dial out using both the SIP phones and console dial commands
What does NOT work (any more) is the microphone/DTMF tones of the SIP phones. This must be a server related problem since dialing works and I can use the headsets/microphones with other applications. The SIP softphones are on otehr machines as the asterisk server.
Does anyone have an idea where to start? Is it most likely sound system or ZAP related? Did enyone encounter that problem before?
Thanks a lot Maik
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