Topic: Advice needed on upgrade of old SAIL installation

[mdo]

Hi

We are familiar using SME server for filesharing, email, etc. (all its usual tasks) and due to this and the active community about SAIL here, we used SME as the base for an inhouse SAIL installation some years back. It's SME version 7.3 and SAIL version sail-2.2.1-651.

External communication is done through a tdm400p (?) card with 4 x pots lines, no VoIP to the outside world yet. One extension manages a fax machine. All extensions/internal phones are Linksys SPA942.

This setup is probably pretty simple but I had not been involved with the original installation (and with Asterisk/SAIL in general) so I don't know much about it.

We need to move forward and want to move (test) towards outside VoIP through a SIP trunk (?) which our ISP can offer.

I read about SARK 3.x (based on SME8b6 which I am familar with) and wonder what might be the best move to that? What might be a good way of migrating the old SARK install (what files, databases to backup?) and then start with a fresh install of SAIL 3.x (on a new HDD). Initially we want to use the existing tdm400p card and lines as they are.

Can the setup data be restored from the backup or will the 3.x version be so different that I need to start new and try to redo manually the former setup?

Thanks for any advice. Michael   

[mdo]

Reply (the only one so far) to myself:

http://sarkpbx.com/twiki/bin/view/Main/DocChapter03h#Converting_V2_databases_to_V3
should do the trick (?)

[SARK devs]

Hello Michael

2.2.1-651 is pretty old (released Jul 18 2008).  By SAIL standards that is ancient.  I recenty had occasion to attempt to convert a system of that vintage using the converter you referenced and it was not particularly successfull.   There were a lot of entities in the database back then that disappeared over the years and the converter had a hard time with them.  We got there in the end but for a small system it is probably easier just to start from scratch with a new install on a new box.

You don't say how many phones you have so I can only guess at the workload but I assume it won't be that many if you have 4 pots lines to feed from.

V3.0 will run quite nicely as a VM so you can simply grab the iso (http://sailpbx.com/isos)  and put up a VMware image for testing and conversion. Alternatively, you can use a spare unit if you have one lying around, it will run on pretty much any old PC.

The V3 database (which uses SQLite3) is all in one file and you can pretty much just move it from one machine to another.   

Kind Regards

S

 

 

[mdo]

Thank you very much for the reply and your thoughts on that.

I will probably try to set it up from scratch then. We currently have only 4 x pots (no sip trunk yet) and about 12 extensions plus some logic which extension to ring for incoming calls, etc. Should be able to figure it out.

As much as I would love to virtualise that system, as long as we need the card for the 4 pots I believe there is no chance (?) I was reading somewhere else (at the Proxmox forums?) that it seems very difficult to get this working reliable.

Michael

[SARK devs]

As much as I would love to virtualise that system, as long as we need the card for the 4 pots I believe there is no chance (?)

Yes that's true.  However, a VM is a good way to test and familiarize yourself with the system.  You can move the database over to the new system once you are ready to deploy.

Kind Regards

S

[mdo]

I have done a fresh installation of SME8A16V31 and then a manual configuration for this installation, trying to replicate our small setup (1 x TDM400 with 4 pots, [no SIP trunk yet], 10 extensions).

I have incoming calls working and internal (SIP phone/extension to SIP) but we cannot get outgoing calls to work:

[Mar 18 21:12:47] NOTICE[3409] chan_sip.c: Call from 'Ext704' to extension '54472255' rejected because extension not found in context 'internal'.
[Mar 18 21:12:59] WARNING[11498] chan_dahdi.c: Unable to determine channel for data /5447225
[Mar 18 21:12:59] WARNING[11498] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
[Mar 18 21:12:59] WARNING[11498] chan_dahdi.c: Unable to determine channel for data /5447225
[Mar 18 21:12:59] WARNING[11498] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)

I am still thinking it is a problem with the dialplan but that is so simple that I cannot imagine. In the old version we had 3 routes - "default' with dialplan _0X. "alternate default" with _XXXXXXX and "emergency calls" with _1XX which I have manually replicated in the new version setup.

Any idea where I am going wrong? I have to admit I am new to Asterisk.

Michael

[SARK devs]

It seems to me that the dialplan you quote ( _0X. _XXXXXXX _1XX)  will not match an 8 digit number (unless it begins with a zero).   54472255 is an 8 digit number that does not begin with a zero.

Without seeing your route and trunk statements I can't comment on the DAHDI warnings but it rather looks as though there is something wrong with the group trunk definitions.

Finally, I see you have elected to run asterisk 1.6; have you set the correct delimiter value (comma) in globals panel  (Admin tab)?

Kind Regards

S

[mdo]

Thank you for your reply.

We tried a lot over the weekend and with manual changes, we could get outgoing calls working by selecting a specific line in the dial command so I believe you are right that there is something wrong with the group trunk definitions (I still don't know how to set them properly).

We are struggling and missing an admin guide to understand a fresh setup. As far as I can see, there is no admin guide for the V3 version available? http://sarkpbx.com/twiki/bin/view

And yes, one of the issues were the old delimiter but we found and fixed that quickly.

For the moment I have given up on V3 and we will continue to run the very old version that we have. I need to get a better understanding what other products are available (and how much documentation) to be able to decide which way to go.

Thanks for your replies again.

Michael