[Serusers] rewriting the outgoing URI to its alias

Andy Thomas andy at andythomas.co.uk
Mon Jun 12 20:50:48 CEST 2006


Greger,

 

Im already doing this for calls coming into my SER installation.

However I want to do it the other way around, i.e. from SER/UA to PSTN

 

Therefore I want to rewrite the username if the outgoing INVITE message to
be the alias, not the location, when sending to the PSTN.

 

Andy

 

  _____  

From: Greger V. Teigre [mailto:greger at teigre.com] 
Sent: 12 June 2006 09:33
To: Andy Thomas
Cc: serusers at lists.iptel.org
Subject: Re: [Serusers] rewriting the outgoing URI to its alias

 

I see a lot of confusion on the different user related ids in SIP. Let me
try to explain (and answer your question, Andy, at the end of the post...)

Let's start at the user agent(UA) side:
--------------------
AOR: Address of record is the sip:myuser at domain.com address that you are
known as, just like your email address.  Many user agents don't ask for AOR,
but builds it from other parameters you need to set
Username: The username is normally the user (before @) of AOR)
Realm: Often the domain portion of the AOR, thus AOR = sip:username at realm
Friendly name/displayname: "My name"

Out of this info, the From header is constructed:
From: "My name" <sip:username at realm>
---
Entirely separate (but sometimes confused in user agents):
Authentication username: The user used for authentication
Authentication realm: The realm to authenticate within
Very often the authentication realm is implicitly assumed to be the same as
the AOR realm
---
Then, the user agent will create the Contact header. The Contact header
should be the public contact address of your current location. Thus:
Contact: sip:username at mypublicip:5060
If the user agent is behind a NAT, the mypublicip will be a private address.
---
A sidenote: Unless registration server is explicitly specified, the realm in
the AOR will be used for looking up the SIP registration server using DNS
SRV or A lookups. You should avoid putting the FQDN of your SIP server in
the realm. 
Also, you may in some user agents specify outbound proxy. This is the proxy
where the user agent will send INVITEs (and other outbound messages).
-------------------

And on the SER side:
-------------------
Authentication user/realm are used to do Digest authentication, but are then
forgotten (i.e. not stored).
The AOR is registered in the location table, together with the Contact
header, as well as the source ip and port (if different from Contact). The
fix_nated_register() function handles this setting of the so-called received
parameter.

So, to the routing: 
- Messages that need to be routed (i.e. do not have Route headers) will have
a Request URI; the first line and the part after the message type: INVITE
sip:username at domain.com
- It is by changing this request URI, you do routing.  The t_relay() command
uses the URI to forward the message correctly
- Before forwarding to a user agent, you want the request uri to be the same
as the stored Contact header for the AOR you are looking up. If not, the
user agent may reply with a 404 User not found
- The From header is NOT used for routing, and for backwards RFC
compatibility, you should not change the From header as some UAs will use
the content of the From header to match a dialog (however, if your UAs from
experience still work, there should not be a problem doing it, it's just not
RFC-compliant and may pop up and kick your butt later ;-)
- The AOR in location table is used for looking up incoming messages if you
can find a direct match between the Request-URI in the incoming message and
the stored AOR. If not, you can use the aliases table (and
lookup("aliases")) to match the Request-URI with something in the aliases
table, that again will map to the AOR in the location table
---
So keeping this (fairly simple) concept in focus: Routing (regardless of
LCR, avps or whatever) should focus on finding the correct Request-URI
before you call t_relay().

BUT, there are ways of "messing" up this...
---
- There are several commands in SER you can use to override the Request-URI
(forward_*). They should be avoided, unless you have a valid reason for
having a Request-URI in the message you are forwarding that is NOT
resolvable (either IP address or DNS name or DNS SRV/A resolvable) to the
party you are forwarding to. The reason can be if you want the R-URI to
contain the AOR and then forward the message to a server handling voicemail
- The dst_uri parameter (implicitly set by lookup) will tell t_relay() to
send the message to dst_uri instead of the Request-URI. lookup() will set
this when it finds that the Contact stored for the AOR also has an
associated received ip:port (because the user agent was NATed)

In general trying all sorts of tricks o fix-up things the way you need it
may not be so smart. Stick to the basics and question yourself: Is this
something I really want to do?
----------------------

To Andy, you want the aliases table to map to the AOR (as registered by the
UA in the REGISTER command). You do lookup("aliases") to resolve your DID
into an AOR. Then, later you can do lookup("location") to map the AOR to the
location of the UA (i.e. Contact/dst_uri). Then your Request-URI will be
correct.

g-)

Andy Thomas wrote: 

Does anyone know how I would do this-

 

For all users who have PSTN access, they are assigned a number in the MySQL
alias table which matches a PSTN DDI number.

e.g. user 8000 has an alias of 2071231234, so on an incoming call the
lookup("aliases") function correctly matches the DDI to the user.

 

I want my ser.cfg to rewrite the user on an outgoing call, if a number
exists in the alias table for that user

Obviously, the rewriteuser function will be used, but what do I put in after
that?

I have tried rewriteuser (lookup ("aliases"))  but that doesn't work

 

Can anyone help?

 

 
 
 





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