I’m setting up a vanilla asterisk machine as an answering machine using a X100P card (analog lines) and can’t figure out why the demo dialpan can’t detect my keypresses to enter into extensions.
I’m strictly using the direct demo config files. I did reinstall over an old version of asterisk - but made sure to do clean makes and reinstalls
I’m running on SUSE 10
The only changes I made where to add to the zapata.conf file
context=incoming
channel => 1
and to add to my zaptel.conf file
Any ideas?
Where are the DTMF tones coming from? A SIP phone (locally), or through the PSTN line after zaptel/asterisk picked up a call through the PSTN line?
They’re coming from the PSTN line. I’m using my cell phone to call in to test it out.
A few touchpoints come to mind
-
zaptel driver 1.2.3 did not work for me on my X100P clone card. Incoming audio was disturbed. I went back to zaptel 1.0.10. Works fine for me with Asterisk 1.2.4
-
rxgain might need adjustment. This is a parameter in zapata.conf for lowering or raisng audio levels (BTW, a corresponding dial for outgoing audio txgain is there as well). You can check audio levels like this:[code]# /usr/src/zaptel-1.0.10/ztmonitor 1 -v
Visual Audio Levels.
Use zapata.conf file to adjust the gains if needed.
( # = Audio Level * = Max Audio Hit )
<----------------(RX)----------------> <----------------(TX)---------------->
##################*
[/code]
This is the PSTN dial tone of my installation with 1db rxgain.
- Check what’s happening on the console
asterisk -r
I found the outputs of Asterisk 1.2.x very useful for debugging my dialplan