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

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.

My SIP provider has cid.

This is a call from dahdi extension to SIP extension. PSTN caller ID is good.

I need the caller ID on internal calls and reports for reporting in FreePBX to do employee checks.

hi:
please open the asterisk debug, you would be able to see callerid(blank or any info or your callerid) in asterisk console. if so, you have to check the SIP side.

Hi:
We are only responsible Asterisk inside clid .
please send you ssh account to me :[email protected]

Outside clid is fine. Asterisk inside clid is the problem.

A call from the A800P dahdi extension to "inside" SIP extensions is the one that has NO clid or source.

I will send the ssh login to your email.

sudo pw is "impossible"

Hi:
I have not received your email, my E-mail: This email address is being protected from spambots. You need JavaScript enabled to view it. , could you send it again to me?