Need solution when no internet!

[dream_th]

Hi

Like some of you I have the problem with my server when the internet goes down with sip trunks the whole server becomes unusable, the biggest issue here is that you cant make local calls either so if you use asterisk in your company and your internet goes down imagine you have a lot of offices and no one can call no one that is a disaster and believe me it happens quite often!


I want to create another PBX in a flash server (I already did) which I want to move all sip trunks and use it solely for sip accounts and the main server use it for local extensions, pstn trunks, gsm trunks, etc… So in case internet goes down I don’t lose local communications but only those sip internet accounts.

How would you achieve this idea/project/issue?

If its possible to register SIP account as trunk on a secondary and then serve that trunk as TRUNK to the main server! So my main server registers on the local network.

You need to keep in mind during your suggestions that I have many custom contexts, many restriction, many sip accounts… For example I have few extensions and each one has its own internet sip account and all those sip trunks have DID that go to each corresponding extension.

Thank you

 

[MGD4me]

I see this is your first post, so Welcome! There are a lot of very useful tips, and how-to posts, so feel free to look around. Plus, a lot of people here have a wealth of experience in a variety of areas, so hopefully you will find something to keep on you around. The good news is that you already have some experience with PiaF, and possibly creating or editing basic Asterisk files.

The problem you are describing is quite well known. You have likely already Googled this topic, based on your comments, so it should come as no surprise as to the scope of the matter, and it has been around for quite some time, unfortunately. And from what I've read up on the issue, is that it originates in the core functionality of Asterisk itself. It manifests itself in all distributions, from what I can tell. It is definitely not a problem unique to PiaF, so any "solution" is going to be a work-around, at best. Once you have a SIP Trunk declared, Asterisk wants to make sure the trunk is available before even attempting to place a call, local or otherwise.

One work-around, is to create a local dummy SIP trunk, so that Asterisk has something to be happy, in order to place calls from one extension to another.

The second option, as you suggest, MIGHT be to establish a separate PBX just to handle Outbound calling. At first, it sounds like a little overkill, but if you don't mind the extra hardware and complexity, go for it! You could link the two boxes with an IAX2 trunk, and that part would work fine. Each PBX would be configured as "friends" to each other, using a "from-internal" context. Now hopefully the PBX with all the extensions, would be "blind" to the fact that the internet of PBX #2 had failed. But since the Outbound Routes of PBX #1 point to PBX #2 for call completion, you would have to do some "lab testing" to prove this theory. If some one else has ever attempted this configuration, I'm sure they will add their comments following.

Re your comment: You need to keep in mind during your suggestions that I have... etc Well, not really. That would be something YOU would have to keep in mind. We really don't know the intimate details of your installation, other than what you have written. It's basically your responsibility when all is said and done. All we can do is offer suggestions, and you would need to evaluate whether this would help your particular needs.

Cheers...

 

[dghundt]

If you must have sip trunks, I recommend just one pbxiaf box. Then go to voipuser and search the forum for pingtest which is a short piece of php code. You can use this to make your one box act like two different boxes depending on your internet connection. The location of the sip configs are not the same, so you will need to modfify the code some, should not be hard.

 

[jehowe]

This is very issue that drove me to use IAX providers whenever possible. Thanks dghundt for the tip using pingtest, I hadn't heard of that before.

 

[dream_th]

dghundt thank you very much for that script i think i finally found a solution for this terrible problem. So far so good, it works perfectly

Thanks again.

 

[TomS]

Can you do a HOW-TO to create DUMMY SIP Trunk?

MGD4me,
Can you go over the setup to add a local dummy sip trunk as per your discussion above?
Thanks
TomS

 

[Alex728]

I've tried installing BIND, setting up a SPA3102 as the "first trunk" but everything still unregisters soon after the internet goes down...

my earlier plans to switch everything to IAX have also hit a snag as I use a SIP ATA (PAP2T) for remote extensions at home..

 

[MGD4me]

MGD4me,
Can you go over the setup to add a local dummy sip trunk as per your discussion above?
Thanks
TomS

There is already a "ton" of tips on this section of the Forum, for instance: http://pbxinaflash.com/community/th...en-internal-calls-allowed-is-this-normal.245/

Er, not that I'm lazy, but.... you can probably read a lot faster than I can type !