Not Getting Caller ID on Asterisk Based Openvox UC 501 IPPBX

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Reliance Jio
I am working on an IPPBX project for my home (joint family). I have 3 PSTN lines (BSNL, Airtel and Jio), through Fibre to Home Connections. I could not figure out a way to connect these to the IPPBX through SIP trunk. So, I connected them through the FXO ports of my Openvox UC 501 IPPBX.

I have configured two extensions on the IPPBX. One is an Analog extension through the FXS port of my IPPBX.
The second one is an SIP extension through a Linksys PAP 2 Analog phone adapter.

I am able to call either of the extensions internally from each other. I am also able to make incoming calls through either of the trunks land on the SIP as well as the FXS extension.

On calling either of my trunks from outside when the call lands on the SIP phone, caller id is displayed as "out of area" while when it lands on the FXS extension, no caller id is displayed.
Caller ID works fine when calling internally between the SIP and FXS extensions.

I have checked all the PSTN lines individually. Caller ID is active on all of them.

So in short: THE FXS extension displays no caller id at all while the SIP extension shows the correct Id on internal calls and "out of area" on outside calls.

I have tried to interchange DTMF, Bell and v23 as Caller ID signal and Ring and Polarity for Caller ID Start.
It hasn't worked.

If there is someone experienced in IP telephony in the India System out there, please help me.

Thanks in advance.
I used to use a caller ID convertor with BSNL landline, DECT phone and SPA3012 long time ago. Without that the caller id did not show up in SPA3102.

These days I have Airtel feeding into my FreeSwitch server, while other have it going on into Asterisk. There are threads on this forum for BSNL as well. You can ditch the FXO altogether for these lines (no point in converting audio and lose quality). No idea about Jio though, havent looked into that.
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Thanks Rajil. Thats the basic issue. How do I ditch the FXO port? I need help in configuring the BSNL and Airtel SIP trunks. Can you guide me to the concerned forums you mentioned.
All the relevant information is on this forum.

For Airtel, look at this thread. You will gather all the tidbits in that thread. Essentially, you need to make sure that there is a route between your PBX (ATA) and the upstream proxy, and setup dns so that points to their sip server. Also, make sure only 1 device is registering to their sip server at one time, otherwise you get locked out and need a technician call to be unlocked.