Just a newbie in FreePBX, I setup my FreePBX connected with 1 POTS line, now during my test when calling to the POTS number from outside instead of directly going to IVR it rings(estimated of 20 secs.) first before going to IVR.
What do I need to configure to have it go directly to IVR instead of ring then IVR?
Is the POTS line connected through a gateway? If so, there is a timer that controls when the call is presented to Asterisk. For example, on an SPA3102, it’s called PSTN Answer Delay. If you’re in a country where caller ID is sent after the first ring, set this to 4. If it’s sent after a polarity reversal, 1 should be ok.
If it’s connected through an FXO card in the server, check that the dahdi configuration is correct for your country.
If you still have trouble, provide details on how the line is connected to Asterisk, as well as the source (copper pair from central office, cable MTA, fiber ONT, etc.).
Q: How the line is connected to Asterisk
A: There is an installed DIGIUM Card(not sure for the model/part number) in the Server and that DIGIUM Card have 2 FXO Ports and 2 FXS Ports, but only 1 FXO Port is active which means that is where the POTS line is connected.
Then base on your suggestion to check the DAHDI Config, the settings that I only configure is System Settings for Tone Region and Modprobe Settings for Opermode set them both to my country and the others are in default. But still the issue occurs.
Yes the POTS line has callerID, actually this POTS line is one of the 6 POTS line connected to our old FreePBX with the following details:
FreePBX Version: 18.104.22.168
Asterisk Version: Asterisk 11.8.1
Then now they want to replace it with a newer version so I borrowed one of the POTS lines from the old ones to use for the new one. But this issue occurs. I also setup all the settings that can configure the country in the system(Under Connectivity). This POTS line is just from our local telco provider here in the Philippines.
claims “PLDT uses DTMF, whereas, Globe and Bayantel having newer equipment uses FSK.”
Wow, I’ve never heard of different caller ID formats in the same country.
Look at the settings for cidsignalling and cidstart in your working system and set the new system the same. See
If you still get Failed to decode CallerID errors, find out what format your line is actually using (by listening with a butt set or by asking the telco), set cidsignalling and cidstart accordingly, retest.
If no luck, post what you found and a new log using what you believe to be the correct settings.