[Asterisk-Users] Caller-ID for alphanumeric SIP uris

[David Beckemeyer]

My first post here, so a brief intro:

I'm somewhat new to Asterisk, but have been working with SIP
in depth for about 3 years.  I studied Asterisk for about a year 
and have been experimenting with it hands-on for the past 
month or so.  I've done 6 Asterisk installs in wildly different
roles/applications, some of them test systems, others in 
semi-production, so I know a little bit about it.  I've setup
voicemail, meetme, ENUM, and other Asterisk features, and I've 
written some AGI scripts and done some other semi-interesting 
tweaks.

That said, I'm curious about how others might solve the following
problem.  In a pure-SIP environment, if a user has an alphanumeric
SIP uri, say sip:joe at sipservice1.com, when that user calls another 
SIP phone, (a real IP phone, as opposed to an ATA), via a SIP proxy,
that phone can log the full URI, and 'call return' works because the
SIP phone calls that URI.  With Asterisk, such a call would come in 
with the SIP From: header (thus Caller-ID in Asterisk parlance) as
something like:

  From: "joe" <sip:asterisk at 204.16.112.70>;tag=as54f3792a

In this case, Asterisk doesn't know how to return the call, nor
does the SIP phone, because even if the SIP phone can dial full
alphanumeric URI's with some kind of a 'call return' feature, 
the sip:asterisk at 204.16.112.70 (where 204.16.112.70 would be the
IP address of the Asterisk server), isn't a valid URI and doesn't
route a call to the original SIP URI: sip:joe at sipservice1.com.

I've thought of some tricks for handling this, and I've looked
around the archives and Google searches, but haven't seen much
discussion of this issue.

TIA,

David