[Asterisk-Dev] Re: [Asterisk-Users] SIP change...

[Mark Spencer]

> Normally the caller-id is taken from "remote-party-id" in the SIP
> INVITE.  We don't see that field poplated in this INVITE.  What is the
> originating gateway?  What device is sending the call to the 827?  We
> should be seeing "remote-party-id" in the INVITE.

The string "remote-party-id" does not even appear in RFC3261.  A little
bit of googling reveals it seems to be something "up in the air" and there
is no RFC which seems to reference it.  Further, in its absense (as I have
learned), RFC3261 states of the "From" header:

   The From header field indicates the logical identity of the initiator
   of the request, possibly the user's address-of-record.  Like the To
   header field, it contains a URI and optionally a display name.  It is
   used by SIP elements to determine which processing rules to apply to
   a request (for example, automatic call rejection).  As such, it is
   very important that the From URI not contain IP addresses or the FQDN
   of the host on which the UA is running, since these are not logical
   names.

Therefore, the Cisco should use it as CallerID.  If, however they want it
in a different form, let me know the document they're referencing and the
form it's needed in and I'll see what I can do.

Mark

p.s. Does anyone still not see what a bloated, vague, and overly
complicated pile of garbage SIP is (although H.323 still manages not to be
outdone)?