Production Migration

**VOIPoTim** · 05-31-2008 09:31 AM

Yeah, X-Lite with the default settings worked for me as well. Phone number as User and Auth User. No other changes outside of defaults.

Are you seeing issues with it?

**gbh2o** · 05-31-2008 11:38 AM

Now that my password has been reset, it plays nicely on each of the Zoom, PBXiaF system and PBXes account (the normal resting place for it) again.

mercy buckets!

**burris** · 05-31-2008 11:59 AM

If I hook up another adapter to the test bed and use the password I am now regestered with, will it mess up my VOIPo line when I go back to it??

**VOIPoTim** · 05-31-2008 12:05 PM

Originally Posted by burris

If I hook up another adapter to the test bed and use the password I am now regestered with, will it mess up my VOIPo line when I go back to it??

Only one at a time will work, but it won't affect your ability to switch back to codeblue.

**chpalmer** · 05-31-2008 01:04 PM

Version 3 build 41150 Im getting registration error 408 Request Timeout.

I have the newest build.

I have another ATA on here with the line Im trying on also. Could that affect it?

**burris** · 05-31-2008 01:19 PM

Fired it up on my SPA 2102...it registered immediately...calls worked in and out...

Only unusual thing was as soon as it registered and I went off hook to dial out, I got a stutter tone. I would imagine that you might have VM enabled in the test bed as I have mine disabled in my VOIPo account..

Not too shabby.

**VOIPoTim** · 05-31-2008 02:11 PM

Originally Posted by chpalmer

Version 3 build 41150 Im getting registration error 408 Request Timeout.

I have the newest build.

I have another ATA on here with the line Im trying on also. Could that affect it?

It shouldn't. You're using zeus.voipwelcome.com and not zeus.voipo.com?

**VOIPoTim** · 05-31-2008 02:11 PM

Someone asked how many servers they can choose from....

For those wondering, all core machines are now going to use the voipwelcome domain in their hostname. It's nothing special, just an anonymous domain to be used in headers so resellers won't have clients potentially seeing VOIPo.

Here's some additional background... When we move everyone, you'll be using sip.voipo.com and that'll be the only option to connect to. Once we fully roll it out, it will then automatically distribute people if needed and handle failover. No 50 servers choices like VT unfortunately.

Our setup is a little different in that VT using Asterisk is VERY limited in terms of how many users can use each server. Many of VT's problems (my opinion/theory only) are likely caused by loads. Asterisk just isn't designer for carriers and the creator of it will even tell you that. Ever notice how things are going well with VT and then someone "outs" the working server and it goes downhill? It's because Asterisk can't handle large loads. Ever wonder why VT moves people without them knowing sometimes? They have to fight to distribute loads. Since it's user-controlled, there's another variable that makes it hard to do this. Asterisk doesn't do well when it hits a few hundred simultaneous calls. It then gets very flaky and inconsistent. Ever wonder why random issues come up and then later they just go away on their own?

We use Asterisk for voicemail which it's fine for, but for everything else, we use a carrier-grade softswitch (OpenSER) which can handle tens of thousands of requests per second and just proxies everything that's designed for carriers. Voicepulse uses it, Cisco uses it as the base for some of their commerical systems, etc.

OpenSER does not handle media at all...it's just a proxy that sets up the connections. In most cases, we are handing off the media now to remote gateways. In some cases that we do handle media, OpenSER doesn't handle it, but it can be distributed to a cluster of weighted telco-grade switches or softswitches vs having to go through the server (like Asterisk). That's where load issues typically come into play.

We've been using OpenSER since day 1 and hopefully you've seen how mcuh more reliable it can be. It doesn't do as much in some ways and that's the beauty of it.

So with us, just a few telco grade softswitches on the frontend with the way we have things setup can addresss the same load that 100 Asterisk servers could. Asterisk is great for businesses, PBX solutions, and for functionality/features. It's not good for large-scale setups though.

**chpalmer** · 05-31-2008 02:30 PM

Originally Posted by VOIPoTim

It shouldn't. You're using zeus.voipwelcome.com and not zeus.voipo.com?

Right. This is probably a network issue on my end and to prove that Im going to evict the "gamer" off my desktop shortly and try from there...

I copy paste everything just because I dont trust my eyeballs sometimes.

The RT31P2 Ive got on that line is behaving.