Caller ID - BT PSTN & Freepbx

Hi ,

I am having trouble with a Grandstream Ht813.

Does anybody have the UK settings for Caller ID for BT Openreach lines?

I have created a PJSip trunk in Freepbx and added an in inbound route. The problem is Caller Id is showing up as the trunk number instead of the PSTN number.

Any ideas?

Thanks

No idea, but do you mean that incoming CallerID is wrong or is it the caller ID which is seen by anyone you call?

I have called the Freepbx trunk 5000 for example.

This is also the Sip username used on the Grandstream Ht813.

When you call from the PSTN Freepbx via the Ht813 answers the call bur the caller ID displayed is the Sip username (5000) instead of <area code > <Telephone number>

I am seeing a few references to the UK needing Line polarity reversal. I have that currently disabled should it be enabled?

It says check with PSTN supplier before enabling.

Edited by dlucas46 (Wed 09-Oct-24 23:55:58)

Hmm. All a bit random. If caller display is showing something meaningful albeit not the expected number, then I doubt line polarity reversal will have any bearing on it. At the moment, you don't know whether the display is the SIP username or the trunk name. Change one of them and try again to get a handle on this.

In reply to a post by dlucas46:

Does anybody have the UK settings for Caller ID for BT Openreach lines?

Openreach SIN227.

Sipura/Linksys/Cisco devices call this 'ETSI FSK with PR(UK)' or similar, the manual for the Grandstream HT813 suggests Caller ID Scheme should be 'SIN 227 – BT'

I already have that set.

Does the number of rings before the HT813 answers matter?

I read one setup for France and they said it the rings were below one you did not get caller ID working correctly.

Not for UK signalling, the caller ID handshake starts the detection process whereas other formats send the ID between first and second rings.

The image of the status page in the manual doesn't appear to display any CLID info, can you see any information in the Freepbx logs as to what information is sent in the SIP messages?

Not sure I understand your setup:

Do you have a BT analogue line (and some sort of FXS/FXO line card to convert analogue to digital in your Freepbx) or do you have a BT Digital Line?

If it is a digital line, then turn on logging on PJSIP to get the incoming SIP messages. The calling line ID can be in various SIP parameters. e.g. "From" header or p-asserted-identity. See whether you can find the CLID that you want to display and then change your dialplan to modify the message accordingly.

In reply to a post by essex_man:

Not sure I understand your setup:

Do you have a BT analogue line (and some sort of FXS/FXO line card to convert analogue to digital in your Freepbx) or do you have a BT Digital Line?

If it is a digital line, then turn on logging on PJSIP to get the incoming SIP messages. The calling line ID can be in various SIP parameters. e.g. "From" header or p-asserted-identity. See whether you can find the CLID that you want to display and then change your dialplan to modify the message accordingly.

I have a BT PSTN line that is running into a Grandstream HT813 with an FXO port.

I can get incoming and outgoing calls across the POTS line but the incoming caller ID is the Trunk name / Sip user of the HT813 ATA and not the caller ID from the BT PSTN.

In reply to a post by tdw42:

Not for UK signalling, the caller ID handshake starts the detection process whereas other formats send the ID between first and second rings.

The image of the status page in the manual doesn't appear to display any CLID info, can you see any information in the Freepbx logs as to what information is sent in the SIP messages?

I assume Pjsip logging would be enough to get the incoming sip messages from the FXO gateway?

You need to get the CLI being read correctly by the Grandstream gateway before you have any hope of passing it through to the other end of the SIP trunk. Is there anywhere in the gateway that would show a call log? Might also be worth asking Grandstream support.

I think the grandstream has the ability to syslog, could provide some information.

I will see what grandstream have to say. Probably worth asking over the the freepbx forums as well.

maybe this helps

https://community.freepbx.org/t/configure-grandstrea...

The problem is Caller Id is showing up as the trunk number instead of the PSTN number.

These are PSTN lines, not VOIP … is that correct ?

Sounds to me like they be ‘Feature Lines’ which a main hunt group number, which is what have been displayed, but each had a direct dial in number too.

Hi ,

It is a single BT PSTN line terminated to a single FXO port on the HT813 ATA.

I am not convinced that the Caller ID is getting relayed to Freepbx from the POTS line.

Well Pjsip logging shows this:

[2024-10-11 11:28:39] VERBOSE[462063][C-00000026] pbx.c: Executing [5000@from-pstn:9] ExecIf("PJSIP/5000-00000066", "1 ?Set(CALLERID(name)=5000)") in new stack

I think that is the root problem.

In reply to a post by dlucas46:

Hi ,

It is a single BT PSTN line terminated to a single FXO port on the HT813 ATA.

I am not convinced that the Caller ID is getting relayed to Freepbx from the POTS line.

In your original post you said ….

The problem is Caller Id is showing up as the trunk number instead of the PSTN number.

What do you mean by he ‘trunk number’ ? And what is the ‘wrong number’ you’re receiving instead .. (not the actual numbers themselves, {obv} just a description.

Do you have a standard analogue handset which has the ability to show caller display ?

You could connect this to the NTE and disconnect its current connection to the handset .. if this then works as you expect , the problem IS as you first guessed, your kit or its set up … if it doesn’t do as expected, it’s time to contact the voice service supplier.

In reply to a post by dlucas46:

Well Pjsip logging shows this:

[2024-10-11 11:28:39] VERBOSE[462063][C-00000026] pbx.c: Executing [5000@from-pstn:9] ExecIf("PJSIP/5000-00000066", "1 ?Set(CALLERID(name)=5000)") in new stack

I think that is the root problem.

The entry in the log file is only a symptom. As I said earlier, you have 2 things named 5000, so if you change one of them, you might get closer to pinning it down.

Hi,

Yes I get you point, but FreePBX wants the Trunk SIP name to match the Sip account on the ATA.
I am not sure I can change one or or the other and have them different if you get my meaning.

Edited by dlucas46 (Fri 11-Oct-24 23:00:19)

In Freepbx I have defined an incoming trunk. This is a PJSip extension with the number 5000.

On the Grandstream HTA the FXO port sip user is 5000 and pointed at Freepbx by IP address.
I have also used an option called Unconditional forward.

The Handsets I have are Analogue units on another H814 ATA , They are displaying internal caller ID correctly.

But if I get a phone call on the landline which is connected to a Grandstream H813 ATA FXO port,
the caller ID displayed is that of the PJSIP trunk / extension (5000) instead of the Mobile or other landline number. This information does not seem to be being feed through to Freepbx and then onto the handset.

In reply to a post by dlucas46:

But if I get a phone call on the landline which is connected to a Grandstream H813 ATA FXO port,
the caller ID displayed is that of the PJSIP trunk / extension (5000) instead of the Mobile or other landline number. This information does not seem to be being feed through to Freepbx and then onto the handset.

Zarjaz's point is that - at least in your narrative to us - you have not established that the CallerID is actually being sent on your PSTN line

In reply to a post by dlucas46:

Yes I get you point, but FreePBX wants the Trunk SIP name to match the Sip account on the ATA.
I am not sure I can change one or or the other and have them different if you get my meaning.

If it is set in one place and one place only, then what you say makes sense. But if it is set in 2 places, then the strength of the concept that the 2 must be the same is uncertain. Sometimes it is for no more than the convenience of the user documentation.

Sometimes it is helpful to disobey the instructions to find out where something is going awry, for the sake of learning something, rather than for the sake of a permanent setup.

I will check the PSTN lines caller ID directly by plugging in a phone directly to the line. If that handset gets the correct information, I will move on to the ATA, which is where I think the real problem probably is.

I will contact Grandstream support for the correct UK settings and see what they say.

In reply to a post by dlucas46:

I will check the PSTN lines caller ID directly by plugging in a phone directly to the line.

This is exactly what Zarjaz was saying, you need to first simplify and test before going all complicated with your setup.

Turn on the SUP logging on Freepbx or use wireshark to capture the SIP Invite arriving from the Grandstream and then see whether you can find the CLID number you are looking for. If it is no there, then the problem is in the Grandstream, otherwise you need to change the dialplan to pull out the correct field

In reply to a post by dlucas46:

In Freepbx I have defined an incoming trunk. This is a PJSip extension with the number 5000.

On the Grandstream HTA the FXO port sip user is 5000 and pointed at Freepbx by IP address.
I have also used an option called Unconditional forward.

The Handsets I have are Analogue units on another H814 ATA , They are displaying internal caller ID correctly.

But if I get a phone call on the landline which is connected to a Grandstream H813 ATA FXO port,
the caller ID displayed is that of the PJSIP trunk / extension (5000) instead of the Mobile or other landline number. This information does not seem to be being feed through to Freepbx and then onto the handset.

Ah … that’s me out. Apologies for being unable to help.

In connectivity - DAHDI config - analog hardware - global settings, make sure you have the following:

Other Global Dahdi Settings:

rxwink = 300

signalling = fxs_ks

context = from-pstn

sendcalleridafter = 1

hidecallerid = No

callwaitingcallerid = yes

callerid = asreceived

ukcallerid = yes

cidsignalling = v23

cidstart = polarity

Is that in Freepbx?

If so Freepbx 17 does not seem to have same menu.

All I have is DAHDI Channel DIDs

It is FreePBX but mine is an earlier version. I remember I had the exact issue with no CID showing when I set it up years ago, and adding all those settings sorted it out. Not sure where it might be in version 17 unfortunately.