Hi. Newbie here
I've installed PIAF from scratch a few times and am trying to configure the system in baby steps. I have a SIP phone working (I can call talking clock *60) but I can't get my analog phone or Bell line to work with TDM410p clone card. The card is installed and configured properly according to dahdi_tool, lspci, lsdahdi, etc.
I think the problem lies in the devices registering in Asterisk. I've properly followed online instructions for building a basic zap extension, zap trunk, and route but I can't make test calls with either the SIP phone or the analog phone.
I understand there to be a conflict between the dahdi-channels.conf and chan_dahdi.conf files. One is referenced by FreePBX and the other is populated by dahdi_cfg. In fact no chan_dahdi.conf file exists but /var/log/asterisk/full complains that it isn't found. What file is trying to call it?
Any ideas? Where can I confirm the logical configuration and registration of my FXO/FXS devices?
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I think I found a solution already.
I read on a forum somewhere to delete the FXS ports from dahdi-channels.conf, create chan_dahdi.conf (from the available template) and rearrange it so that channel=1-2 is defined first #including dahdi-channels.conf (my FXO's), then define group=1 and #include chan_dahdi_additional.conf which are my extensions (channels 3 and 4) defined in FreePBX. I suppose I could have created an empty chan_dahdi.conf and #included dahdi-channels.conf as a whole.
Anyway I can now call the SIP softphone from the analog phone (and vice versa) and with either phone I can call out on the Bell Line.
The sound quality is a bit scratchy so I'll see about rearranging the parameter settings. I also had an IRQ conflict that slowed me down for a while. I hope that doesn't pop up again!
I'll post the resulting conf files once I'm finished playing with them.