Analog phone on FXS port - No ringing -> straight to voicemail

hi, mwilson75
I think your extensions.conf has some problem,the type of channel is dahdi,not zap,please check it.

I think you are using dahdi, so when your dialplan should like this:

Dial(dahdi/1/xxxx)

Please notice " not zap "

I changed the "Dial" setting in FreePBX extension to dahdi/2 and now it rings and accepts calls.

But, ...caller ID shows unknown when I call from Analog phone to SIP extensions. Also in FreePBX reports clid and source are blank.

Any ideas on those 2 problems??

Hi:
1) set loadzone=your country, defaultzone=your country in /etc/dahdi/system.conf
2) set country=your country in /etc/asterisk/indications.conf
3) load driver with your country code:modprobe opvxa1200 opermode=YOUR COUNTRY
4) set this in chan_dahdi.conf
cidstart=ring
cidsignaling=dtmf
usecallerid=yes
if it still has issue, you can check whether your PSTN line is provided with caller ID function.

you should check the extensions config in web gui

It also still shows blank clid and source in FreePBX report.

1) set loadzone=your country, defaultzone=your country in /etc/dahdi/system.conf
2) set country=your country in /etc/asterisk/indications.conf
3) load driver with your country code:modprobe opvxa1200 opermode=YOUR COUNTRY
4) set this in chan_dahdi.conf
cidstart=ring
cidsignaling=dtmf
usecallerid=yes
if it still has issue, you can check whether your PSTN line is provided with caller ID function.