Hello,

I have a question on the caller-id interworking between PSTN and SIP. In
PSTN, there are two cases: "anonymous" and "no caller id". "Anonymous" is
used when the caller sets "presentation not allowed" in the Setup message.
The "no caller id" is the case when caller id is genuinely not available
(for example, interworking with decadic trunk, etc.).

My question is: when sending SIP Invite, is there a way to distinguish the
two cases. I checked in the SIP rfc and the only option I see is
"anonymous".

So how to handle it?

Thanks,
Shikhar Sarkar

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