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
@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 [s@macro-dialout-trunk: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.
@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.
@BlazeStudios -- Executing [s@macro-dialout-trunk:20] ExecIf("SIP/201-0000002b", "1?Set(CONNECTEDLINE(name,i)=CID:540922XXXX)") in new stack -- Executing [s@macro-dialout-trunk: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?
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."
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.