FreePBX seems to ignore my ata's *67 feature code to block caller ID

I have a HT701 by Grandstream.
I have the trunk cid set to “hidden” <NXXNXXXXXX)
It works if I remove the extension’s CID, but that would block the CID from all outgoing calls.
But when I use *67 then the number or even the Send Anonymous option in the ATA’s menu it doesn’t block the CID.
I use VOIP.ms as my sip trunk if that matters.
I have no CID set with them to place on the call either.
Here the asterisk log when I try to make a CID Blocked call:
http://pastebin.com/cAmWY7AG
Please let me know if you need any other logs or captures

P.S. Here is log of when the extension has no CID set in the FreePBX settings
http://pastebin.com/MdADYx5U

@thomasb9511 These debugs are incomplete. The first one doesn’t show what happens to the call when it’s sent out and there are no SIP messages to see how the call is be presented to the provider.

I do see this though:
– Executing [[email protected]:22] Dial(“SIP/201-00000038”, “SIP/Voip/NOUTGOING,300,”) in new stack

You’re telling it to open a SIP channel on the Voip trunk and to send the call to the destination NOUTGOING that’s not going to end up anywhere.

Redo them with “sip set debug on” issued from the Asterisk CLI and show those please.

@BlazeStudios NOUTGOING is a redaction for a real number.

here is a log with sip set debug on with 867 dialed before the number:
http://pastebin.com/7xu1Qb6J

540626XXXX and 540922XXXX are redactions

and here is a normal call:
http://pastebin.com/kNpHfGfP

@thomasb9511

From the call that has *67, shows the call as anonymous.

From: “Anonymous” sip:[email protected];tag=1539221497

From the call without *67, your caller ID is shown.

From: “Thomas Brown” sip:[email protected];tag=743948028

Are you overriding the extensions caller ID at the trunk level?

@BlazeStudios
It is set to allow any cid

@thomasb9511 You will probably need to talk to your provider. Asterisk is respecting *67 and sending the call out as anonymous. You should check with the provider to see how they are seeing the call and sending it out.

I will, so it isn’t necessarily my system’s fault

Doesn’t look like it.

@BlazeStudios
-- Executing [[email protected]:20] ExecIf("SIP/201-0000002b", "1?Set(CONNECTEDLINE(name,i)=CID:540922XXXX)") in new stack -- Executing [[email protected]:21] ExecIf("SIP/201-0000002b", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)540922XXXX)") in new stack

Does this mean the CID was set to (Hidden)540922XXXX) or 540922XXXX?

That means that it is hidden…

Just wanted to make sure the fact it was listed two ways confused me.

in your trunk

fromuser=whatever

can be a problem

@dicko

I wonder why?

Here is my outgoing trunk config

`canreinvite=nonat
nat=yes
context=from-trunk
host=washington.voip.ms
username=100000
fromuser=100000
secret=*****
type=peer
disallow=all
allow=ulaw
trustrpid=yes
sendrpid=yes
insecure=invite
qualify=yes```

Also I try the call with and without fromuser=whatever
and still gets the CID passed through.

I would send a call log with sip debug to their support, and ask why.

Already have and I just got a reply:
"In the case of the call made with *67, I’ve noticed the From appears as sip:[email protected], I recommend you to confirm if this is the From that you’re receiving from the Device to your PBX, and if that’s the case to confirm that your PBX is accepting this information.

And also to make sure that is not the format you’re passing to our server. This is because that format is incorrect, and our server expects From: “CallerID Name” sip:[sipaccount]@[server].voip.ms, and since the format doesn’t match the call could fail."

That (sipaccount) is probably what your fromuser= should be

That is what is set to though.

I am probably going implement when asterisk gets a *67 call to change the CID.
to “hidden” <540922XXXX> which seems to work well.
The question is now how to do that.

@thomasb9511

Did voip.ms offer any suggestions on how to block caller id on outbound calls?