DO NOT RESTORE (be careful)

telecomtechnician

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#1
Hi there

I installed elastix 1.3-2 on a very robust server and configured the call center module. After a lot of effort and frustation, the installation went fine. Everything went fine until an important request came up: outbound campaign with sip trunks. That was not possible on this version. That´s when we decided to wait until the next version of elastix came up that supported autodialer with sip. When the elastix 1.5-1 rc came up, I decided to install it on another server that is exactly the same as the first one. We did a backup of the first server (elastix 1.3-2) and copy it at the second server (elastix 1.5-1 rc)and restore it. I assumed that everything should work like a snap, but it did not. Different conclusions and possible explanations came up, to justify why this was not working. As I knew it was not a stable version, we left the problem like that and waited for the stable version to come up. When it finally did, we did everything as I previously explain but with elastix 1.5-9 stable, and asssumed once again that everything should work like a snap. IT DID NOT.

Our worries and concerns went to another level, thinking about the network, the sip trunk server, bla, bla, bla.

We came up with an interesting idea, (why not do a fresh install and DO NOT RESTORE anything) just configure the basics to put it to work and test it?. So we did that and !WELLA!, the call when out on the first try.

So we conclude a few things:

- The beta testers did not test a restore from 1.3 to 1.5
- If you want to restore from 1.3 to 1.5, it should be done carefully and do it step by step, because the database part of asterisk and mysql once they are restored in the new version they simply do not work and break the entire IPPBX functionality.


All of this, is based on my personal experience and there is no intention to hurt or criticize the hard work that all of the elastix team has done to put a very stable IPPBX on the top. But an issue like this is important to be shared.

Waiting for your comments

David Medina
 

dicko

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#2
Thank you David for the caution,

May I ask if the restore was an "Elastix" restore or a "FreePBX" restore? (the FreePBX backup/restore being less pervasive and doesn't include users accounts etc., nor of course the callcenter stuff, just the IPPBX)
 

Patrick_elx

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#3
As I wrote on another post, I had the same issue for an elastix gui restore from 1.4 to 1.5.

I had to go in free pbx and make a submit for each extension and ring group, if not they were always busy.

Maybe a free pbx database or something that was not taken into account by elastix. (for info the version of free pbx used on the 1.4 and 1.5 were exactly the same)
 

pawels

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#4
Outbound campign via sip truk not depend on elastix version.
It depend on cc module. Version 1.4 supports such dialling.
 

telecomtechnician

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#5
Hi there, it was an Elastix restore, I did not do it through freepbx.

David Medina
 

dicko

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#6
David:, Hi there
I ask because I was able to restore through FreePBX, (only what it does of course) successfully, whereas the Elastix whole kaboodle, just plain broke things. (same set of of FreePBX and it's modules on old and new before the restore).

There are couple things i'd like to get with dahdi proper on a couple of systems, but my instincts and sense of caution tell me to wait for 1.5's growing pains to be settled. (basically, if it works don't mess with it)
 

rafael

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#7
Hi David,

The thing about backup and restore would only work well among the same version of Elastix at this time. The reason for this is that all what the backup does is tar all the files you backup and put them in a file. There exists incompatibility with stuff like freepbx, zaptel/dahdi, etc....

So we need too do is document this (I would start by making your post sticky) and improve the backup restore module.

Anyway if you do a backup in the same version of Elastix that it was installed, then it would work well to get to the state where it was before. It won't work well for upgrades yet :(

Regards,

Rafael
 

izrunas

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#8
Is there a plan to create some sort of "verify data version and upgrade if needed" routine to the restore process?

If I had known that one MUST have identical versions, I could have saved myself a total of about 16 hours sitting overnight in a client's office futilely trying to restore from a semi-upgraded box to a fresh install.
 

izrunas

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#9
I have now concluded that the backup system is USELESS when trying to move from one machine to another one. I have tried every possible trick and the "restored" copy is broken and unfixable on the newer machine every time. Forgive my aggressive tone, but I have lost about a week of productive time on this one box and the issues with it.
 

ramoncio

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#10
Try doing a mondobackup and restoring it in the other machine. There is a script somewhere here to install mondo.
 

telecomtechnician

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#11
Hola, me imagino que hablas español

La idea cuando creé este post hace dos meses fue para dar a conocer mi molestia, frustación y creo que nadie lo habia hecho en ese momento, intentar restaurar elastix 1.3-2 de una máquina a otra máquina con elastix 1.5-2 y por supuesto no funcionó.

Esto fue una inversión de tiempo importante pero a su vez frustante y con un aprendizaje. Es un hecho que restaurar dentro de una misma versión SI funciona bien, pero cuando lo quieres hacer de una versión a otra TE GARANTIZO QUE ALGO SE VA A ROMPER.

Hay muchos cambios a nivel de archivos y de librerias entre la versión 1.3-2 a la 1.5-2, por ende si no haces backups selectivos e intentas subirlos por parte, es posible que te funcione como es posible que no.

Inclusive dentro de la misma versión, si las máquinas que tienen el elastix 1.5-2 y no están exactamente iguales (todos los módulos y dependencias de freepbx actualizado) una vez mas te digo, ALGO SE VA A ROMPER.

Por ende lo repito DO NOT RESTORE,(be careful)

Suerte

David Medina
 

iraj.kheirati

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#12
isolate sip trunk

hello dears

how i can isolate sip trunk channels?
i am using of 4fxo gateway(external device) and i maked sip trunk for dialin and dialout.
in example i have 2 extensions (301,302),i want for dialout; extension 301 used of port 1 of gateway and 302 used of port 2 of gateway for dial out.
please help me how i can isolate and analyze sip trunk channels for dial out.
thanks
 

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