<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman, new york, times, serif;font-size:12pt"><DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">Thanks for the reply Alex! I really appreciate it.</DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif"> </DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">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?</DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif"> </DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">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.</DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif"> </DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">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?</DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif"><BR>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)</DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif"> </DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">Thanks again for the reply,</DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif"> </DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">--Doug<BR></DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">----- Original Message ----<BR>From: Alex Balashov <abalashov@evaristesys.com><BR>To: Doug McLetchie <dougmcletchieatwork@yahoo.com><BR>Cc: users@lists.openser.org<BR>Sent: Wednesday, February 6, 2008 1:23:51 AM<BR>Subject: Re: [OpenSER-Users] newbie questions<BR><BR>Doug,<BR><BR>Doug McLetchie wrote:<BR><BR>> For inbound calls (calls coming from another carrier to my big expensive <BR>> gateway & destined for a specific PBX), I'd like my gateway to send the <BR>> call to the proxy, who will determine which PBX to send the call to, and <BR>> then send the call to to correct PBX via the Gateway. I don't <BR>> understand how to set up OpenSER to send the call via the Gateway. If I <BR>> provision the static ip address of the PBX in the proxy, won't it try to <BR>> send directly to that IP address instead of sending it to
the Gateway? <BR>> I think that the feature that I'm looking for is something like an <BR>> outbound proxy, for the proxy. (does that make sense?)<BR><BR>OpenSER can certainly do what you are trying to accomplish.<BR><BR>SIP routing is done by URI, which consists of a "user" part and a <BR>"domain" part. The "user" part is the number (or alphanumeric <BR>identification, in the case of pure-VoIP peering) and the "domain" part <BR>is the IP "place" at which the "user" part is reachable.<BR><BR>When you route a call to some URI, what you are really saying is, "Here, <BR>domain, you must know what to do with this 'user' part - i.e. have <BR>reachability information for it (a SIP contact bound from a <BR>registration, for example)."<BR><BR>A proxy by itself isn't enough. If you need to reach these PBXs, you <BR>clearly need to know how to reach them. This requires a SIP registrar <BR>somewhere, so that the PBXs can register
against it and say, "Here, you <BR>can reach me at such and such IP and port." Or, I suppose, you can <BR>define these contacts statically with a database interface from the <BR>proxy, in which case you don't need to use a registrar.<BR><BR>I don't know what this Big Expensive Gateway is, but if it's something <BR>like a Session Border Controller, it should be able to forward SIP <BR>REGISTER requests to your proxy/registrar. Or do they register against <BR>the gateway?<BR><BR>Either way, you can perform this resolution with OpenSER.<BR><BR>The problem you *might* run into is sending the same logical call leg <BR>back to the Gateway. Depending on what it is, it may not like that and <BR>perceive a call routing loop, because the call that went through it to <BR>the proxy is the "same" call (in terms of SIP Call-ID, and other things <BR>that make up a logical call "leg") that is now being sent back around to <BR>it. This problem is
often solved with the introduction of a <BR>back-to-back user agent which can re-originate a different call leg.<BR><BR>-- <BR>Alex Balashov<BR>Evariste Systems<BR>Web : <A href="http://www.evaristesys.com/" target=_blank>http://www.evaristesys.com/</A><BR>Tel : (+1) (678) 954-0670<BR>Direct : (+1) (678) 954-0671<BR>Mobile : (+1) (706) 338-8599<BR></DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif"><BR></DIV></div><br>
<hr size=1>Looking for last minute shopping deals? <a href="http://us.rd.yahoo.com/evt=51734/*http://tools.search.yahoo.com/newsearch/category.php?category=shopping">
Find them fast with Yahoo! Search.</a></body></html>