I think I'd like to jump on the bandwagon here also....<div><br></div><div>As said by some of the other members on the list, this IS a very big project. The fact that you are looking for an ISO to test it out, suggests that you don't really understand how it all works. Without going into too much detail, let me just say that OpenSIPs won't do anything for you. You have to configure it from the ground up. So even if you had an ISO, it would give you binaries and modules that essentially wouldn't do anything at all. And honestly the compilation, installation bits are the easy part!</div>
<div><br></div><div>As for this whole Asterisk redundancy. This has been discussed for a long time and there are dozens of ways to do that. Everything from using UCARP, Linux HA, Asterisk "redundancy" patches, database backending, programming backup proxy IP in UACs. One thing I can tell you with reasonably certainty is that anything you do to make "redundancy" with asterisk is almost certainly a hack. That being said, I can also say that many many people (myself included) have come up with methods that seem to work to provide redundancy in a way that is reasonably painless for the end user. But there are always something like:</div>
<div>1. the backup server doesn't have the primary server voicemails synced</div><div>2. phones on the backup server can't call phones still on the primary server</div><div>3. call picked, user presence doesn't work from primary to backup</div>
<div><br></div><div>Just keep in mind that Asterisk itself really wasn't ever designed with the concept that another server would be it's redundant pair.</div><div><br></div><div>As for pairing OpenSIPs and Asterisk. There are a dozen ways to do that as well.. You can do it on the front end (phones register to opensips), on the back end (phones register to asterisk, calls go out to opensips), or somewhere in between even (ie: Asterisk as just a media server). It's all really dependent on how you are going to leverage the technologies of the individual platforms. </div>
<div><br></div><div>I personally try to make asterisk do as absolutely little as possible. I think you'll find scaling OpenSIPs much more enjoyable than Asterisk. ;)</div><div><br></div><div>That all being said, you're really only limited by your creativity and skill level with the platforms. It's best to think of them complementing each other. Pick the features you want to use from each.</div>
<div><br></div><div>As for NAT. Sure you can run these platforms behind NAT, but you're asking for a world of pain. :) Just don't do it. That's my opinion. Far end NAT is tricky enough to deal with to also have Near end NAT issues as well.</div>
<div><a href="http://en.wikipedia.org/wiki/KISS_principle">http://en.wikipedia.org/wiki/KISS_principle</a> :)<br></div><div><br></div><div>-Brett</div><div><br></div><div><div class="gmail_quote">On Wed, Jul 1, 2009 at 10:52 PM, <a href="mailto:lists@grounded.net">lists@grounded.net</a> <span dir="ltr"><<a href="mailto:lists@grounded.net">lists@grounded.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">I've come across this project a few times but have been having a bit of a time confirming just what the project does. I thought perhaps the best way would be to join the list and ask.<br>
<br>
My task is to put together a scalable asterisk based pbx system. Because the boxes will initially have more than they really should installed on them, we need to limit the number of users per box to perhaps 50.<br>
<br>
Right now, the plan calls for every box to have a second one for redundancy. I was planning on manually redirecting connections (for now) but it sounds like opensips could take care of a number of issues.<br>
<br>
I have multiple providers (WANs) at one location but was thinking that for highest reliability, that I might have three locations to be safe unless there are better ideas.<br>
<br>
One would be the location where the initial user connection is made, such as a proxy/load balancer.<br>
Then, two separate physical locations and networks for redundancy. The front end could use both sites as needed but if something went down, could re-route users/sessions to the redundant location.<br>
<br>
This of course is where my questions about opensips come in.<br>
<br>
-From what I can tell, opensips could act as a pbx on it's own but it can act as a proxy/load balancer/gateway to asterisk systems as well.<br>
<br>
-If this is the case, would there be a way of creating a distributed environment, like as in a web server farm, making scaling quite easy.<br>
<br>
-Does opensips handle only new incoming connections or could it actually move sessions from a down server to another which is still up?<br>
<br>
-Would there be any control, or even any need depending on how the back end can be set up, by which to control which pbx/pair that someone registers to?<br>
<br>
-Would I have some method of controlling how many people can register on any one box?<br>
<br>
Thank you very much for this information as it will help to first understand what the project can do.<br>
<br>
Mike<br>
<br>
<br>
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</blockquote></div><br></div>