This is probably a silly question but when I receive incoming calls from withheld numbers, the CID for the call changes to the CID I have of the pjsip trunk. Is there any way of stopping this, so that the call remains as anonymous? I would like to do this so that I can have an inbound route that will handle anonymous calls differently to all other calls.
Calls come in from a Grandstream HT813, which has its FXO port registered to the pjsip trunk. The trunk name is the name of my external landline number. All incoming calls are routed to the pjsip trunk and Caller ID works perfectly when the incoming call has Caller ID. If the call is anonymous/withheld, the CID of the call is setting to the name of the pjsip trunk.
I would like to stop this behaviour, who anonymous/withheld calls remain anonymous/withheld.
What happens, when you set this to no (advanced settings):
Display CallerID on calling phone
EDIT: Ohh…it’s about incoming calls…I had this problem on outgoing calls and the above setting fixed it.
Hey Charles, thank you so much for taking time to respond. It seems like this forum is quiet now, difficult to get a response. I haven’t tried this yet, but I will tomorrow. I’m not sure if this will stop caller-ID altogether though? I’d still want caller-ID to work for calls that have it, I just don’t want incoming withheld/anonymous calls having the CID of my landline number.
It seems like FreePBX is looking for the CID, not finding it so assigning the SID of the trunk instead. There must be a setting somewhere to stop it from using the CID / name of the trunk.
Noticed this in the syslog of my HT813 when making a call with Caller-ID turned off -
Trying to determine if the HT813 is setting the CID to the trunk CID or if FreePBX is doing that. Looking at the FreePBX log, it doesn’t see “unknown” at all.
Admin → Blacklist → + Blacklist Number
Add anonymous and unknown as separate entries.
Then adjust your Admin–> Blacklist → settings tab for the call treatment. But not sure if the incoming caller ID will change. It will handle the call differently to a caller showing their caller ID.
Test to see if this works.
What happens when you set “rewrite contact” to no…in your advanced settings of the pjsip trunk?
EDIT: Your gateway has a setting Caller ID scheme “sin 227-BT”. Is this UK-specific? Maybe the format and the transfer of the CallerID to freePBX is the problem. Maybe the gateway does not use “anonymous” (or whatever freePBX expects) for hidden CIDs.
Caller ID Transport Type is set to Relay via SIP From. You could set it otherwise, but that would require having the PBX e.g. fish it out of PAI and IMO wouldn’t help. On an anonymous call, the HT doesn’t set SIP From, so it contains what would otherwise be there, namely the value of SIP User ID / Authenticate ID, which in this setup must match the Trunk Name.
The Trunk Name can be anything, but Inbound Routes have a stupid requirement that the CallerID Number field be a number. So, you could set it to some value that would never appear in a real caller ID, for example 000000, then create an Inbound Route with DID Number left at ‘any’ and CallerID Number set to 000000, which routes to your desired anonymous call handling.
Another approach: get a number from (for example)
and you’ll be able to see the ‘withheld’ numbers. (UK law likely has restrictions on how you can use this information.)
It will be. Whilst I can’t find a URL for that Supplier Information Note, it will be the BT system where a call is signalled by a polarity reversal, the terminal responds with a partial looping fo the line, and data is transmitted, as V.21 (I think) before ringing current is applied.
Thank you for your response, I really appreciate it. I’ve set it up this way now with an inbound route pointing to IVR. It works, but I was wanting to try the privacy manager feature… but I’m unable to as the call has the CID of the trunk and it only works if there is no caller ID tied to the call.
I’ve renamed the trunk etc. just to 0000, instead of it being my landline number, so I can deal with that. I think I’d like to replace the Grandstream HT813 with something better through time, I just find this box really problematic in different ways.
Thank you for getting back to me. I gave this a try but nothing I change seems to have any effect. I think from what Stewart said, this is really by design from the way I’ve decided to set this up.
I still dont unterstand why you use your old phone line. Don’t you have internet access at this location? Aren’t there a few SIP-trunk providers in the UK?
I’m not using my old phone line any more, I removed that several years ago. The phone line is in a way, VoIP as it’s delivered over fibre broadband in digital but then converted to analogue I believe, by the broadband modem.
Do you pay an extra fee for the phone line? Can you keep the Internet/TV access without the phone?
If yes, it would be much better if you remove your “conversion cascade” and go with a SIP-trunk provider.
It’s one of these things where Internet & TV become cheaper if their phone service is included. I think through time though, I probably will move to a SIP trunk provider to eliminate this hardware as it is generally a nuisance to troubleshoot/maintain.
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