Why There is No Caller ID for Analog Line?

The caller id issues often happen on PSTN trunks.

Please check the following first:

  1. The first thing we need to do is to make sure that the PSTN line’s caller ID service has been enabled. Users can connect the PSTN line directly to a normal desktop analog phone to check if the caller id works.
  2. If the caller id works on the phone, please try change the ‘caller id signaling’ and ‘Caller ID start’ on the analog trunk page.
  3. If you still fail to get the caller id works, please login MyPBX via SSH for further debug. Please follow below steps to get the detail caller id signaling.

Caller Signal Analysis

Step 1. Login SSH.

For S-Series: How to Log in SSH of Yeastar S-Series VoIP PBX

For U-Series & Gateway: How to Login the SSH of Yeastar MyPBX

Step 2. Capture the FXO port signal.

Followed by this guide: How to Get the FXO/FXS/GSM Port Signal Capture

Step 3. Download the rx.raw file and analyze it.

1.  Download  the package and open rx.raw file

2. Use Cool Edit or Audition to check the Caller ID signal waveform;
    Open the rx.raw file as

  • Sample Rate 8000
  • Channels Mono
  • Resolution 16-bit

Formatted the rx.raw file as 16-bit Intel PCM ( LSB, MSB)


Step 3. Analyze the Caller ID.

Examples of caller id signal waveform. 

1) FSK After first ring. If you hear it, the FSK sounds like zzzzzzzz.



2) FSK After Polarity. Actually it's hard to identify the polarity signal. As it is not special, and the sound could be flash signal or other voice jitter. However it seems this situation happens in few countries like UK or Japan.




* If you live in UK, please try ETSI-V23/V23-UK; if you live in Japan, try V23-Japan.

3) DTMF Before Ring. The DTMF would sounds like when you hit the key one your phone keypad. It's totally different with FSK sound, and it is easy to distinguish.


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  • 0

    It would be great if I could do this but Adobe owns cool edit....so what FREE audio editor can I use?

  • 0

    does any one know what to do with cid problem when non of these works?
    I have 3 analoge landline which are properly works on panasonic pbx and ahowing caller id well. but when connected to my voip call center(grandstream which have fxo line capability) the cid ia not working.
    I have captured the tone but its not like any dtmf however when I set my device to fsk it not even connect the call.
    the only workaround is to either disable the caller id or set it to dtmf which in both case no caller id is displayed.

  • 0

    James Kline You can use Audacity:

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