Hi, I wonder if there is some standarized header/parameter to get the 
behaviour I tell now:

Imagine a SIP user  sip:[EMAIL PROTECTED]  registers itselfto a proxy that 
routes him calls from a PSTN gateway and others proxies.

sip:[EMAIL PROTECTED] has assigned two PSTN numbers:
  +111111
  +222222

sip:[EMAIL PROTECTED] registers the address:
  sip:[EMAIL PROTECTED]:5060

When the proxy receives a call from the PSTN gateway for +111111 it will 
generate an INVITE as:
  INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
  From: <tel:+10003345>
  To: <+tel:+111111>

So bob phone can know the PSTN dialed number by examining the To header.

But imagine that the proxy receives an INVITE from other proxy with peering 
relationship, and this INVITE was diverted by the original destination 
(+333333) to +111111.
This is, the INVITE arriving to bob phone will be:
  INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
  From: <sip:[EMAIL PROTECTED]>
  To: <+tel:+333333>
  P-Asserted-Identity: <tel:+92390923>

So in this case bob has no way to know that the real PSTN destination was 
+111111.


Well, I solve this issue by adding a custom header "P-E.164" added by the 
proxy responsible of "bob", so the above INVITE would be:
  INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
  From: <sip:[EMAIL PROTECTED]>
  To: <+tel:+333333>
  P-Asserted-Identity: <tel:+92390923>
  P-E.164: <tel:+111111>

This is, the only SIP node that must add (or replace) this header is the proxy 
responsible of the final user. In this way, bob will know that a call 
originally to +333333 was diverted to him number +111111 (not +222222).


I wonder if it exists a standar way to do this (not very complex please). As 
you see, in my custom solution the final device just must inspect the 
header "P-E.164", no more.


Thanks a lot.


-- 
Iñaki Baz Castillo

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