Thanks for the reply Alex!  I really appreciate it.

 

I've been reading manuals & sip specs all day, and I think that I found my answer, and it really is an easy one.  here's my own answer to my question, can you let me know if this sounds right?

 

When a message is send from a user agent (on the public internet) to my gatweway & then to the proxy (on a private network), the gateway will add a via on to the message.  that's how open SER (or any other proxy) can send a response message back to user agent on the public internet via the gateway.

 

It's really the same thing as setting up a registrar, or any other kind of proxy/user agent that's going to communicate via a SBC of any kind.  So, it that right?  is that how the SIP via header is used?

I understand the part about registrations.  We're not currently doing that in our network, we're using all static IP addressing.  I plan on using registrations in the future.  the gateway that I'm using can act as a session border controller, but it can also act as a class 4/5 switch (TDM or IP).  so it's some serious overkill for what I want to do here.  :o)

 

Thanks again for the reply,

 

--Doug

----- Original Message ----
From: Alex Balashov <abalashov@evaristesys.com>
To: Doug McLetchie <dougmcletchieatwork@yahoo.com>
Cc: users@lists.openser.org
Sent: Wednesday, February 6, 2008 1:23:51 AM
Subject: Re: [OpenSER-Users] newbie questions

Doug,

Doug McLetchie wrote:

> For inbound calls (calls coming from another carrier to my big expensive
> gateway & destined for a specific PBX),  I'd like my gateway to send the
> call to the proxy, who will determine which PBX to send the call to, and
> then send the call to to correct PBX via the Gateway.  I don't
> understand how to set up OpenSER to send the call via the Gateway.  If I
> provision the static ip address of the PBX in the proxy, won't it try to
> send directly to that IP address instead of sending it to the Gateway? 
> I think that the feature that I'm looking for is something like an
> outbound proxy, for the proxy. (does that make sense?)

OpenSER can certainly do what you are trying to accomplish.

SIP routing is done by URI, which consists of a "user" part and a
"domain" part.  The "user" part is the number (or alphanumeric
identification, in the case of pure-VoIP peering) and the "domain" part
is the IP "place" at which the "user" part is reachable.

When you route a call to some URI, what you are really saying is, "Here,
domain, you must know what to do with this 'user' part - i.e. have
reachability information for it (a SIP contact bound from a
registration, for example)."

A proxy by itself isn't enough.  If you need to reach these PBXs, you
clearly need to know how to reach them.  This requires a SIP registrar
somewhere, so that the PBXs can register against it and say, "Here, you
can reach me at such and such IP and port."  Or, I suppose, you can
define these contacts statically with a database interface from the
proxy, in which case you don't need to use a registrar.

I don't know what this Big Expensive Gateway is, but if it's something
like a Session Border Controller, it should be able to forward SIP
REGISTER requests to your proxy/registrar.  Or do they register against
the gateway?

Either way, you can perform this resolution with OpenSER.

The problem you *might* run into is sending the same logical call leg
back to the Gateway.  Depending on what it is, it may not like that and
perceive a call routing loop, because the call that went through it to
the proxy is the "same" call (in terms of SIP Call-ID, and other things
that make up a logical call "leg") that is now being sent back around to
it.  This problem is often solved with the introduction of a
back-to-back user agent which can re-originate a different call leg.

--
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599