@jaredbusch said in Issabel PBX " Elastix Alternative ":

@scottalanmiller said in Issabel PBX " Elastix Alternative ":

@alyragab said in Issabel PBX " Elastix Alternative ":

I think that the purpose of the Issabel is to continue creating the Add On Modules that was supported by Elastix , also it is based on CentOS 7 which will be more reliable than what Elastix 2.5 was.
anyway i think we need to wait till we find Issabel is stable distro and we can depend on it as we did with Elastix.

I assume that they are working form the nearly released Elastix 4 work. That was on CentOS 7 at the end.

And it did work with a lot of patience and knowledge of the underlying systems. But it was a hard fail for the layman.

True. I had it working, but it wasn't ready for prime time when they gave up on it all.

Now log into your new FreePBX system and it will show the Apply Config warning.

Do so.

0_1486444059378_upload-fea00064-3888-40ea-af94-ee5a8ca2979c

At this point you can go through the new system one piece at a time and make sure you are good.
Change a couple extensions to point to your new PBX and make sure they register and can call each other.
Once you are sure that the basics are working, disable the trunks on the old system, enable on the new system, and verify outside connectivity works.
Finally, you can change the rest of the extensions.

3CX's behaviour as a company on the Spiceworks forums is additionally reason to not deal with the product.

@scottalanmiller said in panic button for IP phone:

@Mike-Davis said in panic button for IP phone:

@Dashrender I'd be pitching what ever kind of PBX @Mike-Ralston wants to build and support and what ever kind of phones would work well for them.

That'll be FreePBX.

I assumed as much, but wanted to make sure.. so get a list of functions then compare to functions on old system.

@scottalanmiller said in True Open Source PBX:

@JaredBusch said in True Open Source PBX:

I am interested in Xivo (whatever their new name is).

It is not production ready like FreepBX yet, but their goal of making everything accessible via REST API will make it a very flexible thing.

Me too, certainly lots of potential and I hate the FreePBX interface, so am hopeful there.

Let me know what you don't like if you have time!

Also the Wazo(XiVO) developers are great guys. A few of the FreePBX team (Rob Thomas ( @xrobau ) and I) talk to them ( @quintana ) weekly on IRC/Twitter and we've conversed over a hangout session and at Astricon.

@scottalanmiller said in Elastix 5 Dials 3CX for its Telephony Engine:

And PIAF was the goofy "build it yourself" version of FreePBX. Just silly. Why build it yourself and have things go wrong when FreePBX was totally free and open and did it all for you and had the ability to update, get support, has a community, etc.?

One point of note here @scottalanmiller PBX in a Flash existed a long time before FreePBX was a distro of its own. But yes, once FreePBX existed as a distro, there was never any other reason to do anything else.

@JaredBusch said in Elastix Time Restriction:

@AlyRagab said in Elastix Time Restriction:

Here We can do it per Extension :

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-6-SECT-3.html

and

http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime

Yes, but that is getting way beyond Elastix and into raw Asterisk programming.

What was said above was reference of how to handle it inside the GUI itself.

Both methods work, but dropping to manual editing of configuration files should normally be avoided whenever possible as that makes it harder to use the built in tools for backup and recovery.

Also, upgrades will often over write manual changes like this, meaning you have to remember and reply them after updates/upgrades.

@scottalanmiller said in PBX Update routines:

@fuznutz04 said in PBX Update routines:

I've seen that you can do it via a command, which of course could be scheduled, but I'm always Leary of that with FreePBX.

sudo -u asterisk /var/lib/asterisk/bin/module_admin upgradeall
sudo -u asterisk /var/lib/asterisk/bin/module_admin reload

That's the module part. Don't forget the OS, too.

Right. The OS part (at least in FreePBX) can be set for automatic updates via the GUI in System Admin.

@scottalanmiller said in Elastix Business Edition:

Who the heck wants MailChimp on their PBX?

I have no idea.

@poorizad said in Storing Elastix 2 Call Recordings on NFS NAS Share:

@scottalanmiller hi again....i guess the nfs drive is unmount from the server, because no file record on Nas after the 2016/07/28.....nothing changed on settings.....but a few server shutdown for hardware maintenance.....how can i fix it again ?

Try the mount command again. It might mount right up. If the NFS server has restarted, it would disconnect.

@poorizad said in Storing Elastix 2 VoiceMails on NFS NAS Share:

@scottalanmiller it means i use your previous command with this path . right ?

Correct

@Jimmy_K said in No Outbound calls even TRUNK is registered:

@JaredBusch . Can't be.

It most certainly can be, as that is the error in your log.

@Jimmy_K said in No Outbound calls even TRUNK is registered:

When I use X-lite it is still working well. However when it comes to PBX server, it can't work.

The softphone should not be affecting the dialed number in any way. So whatever digits you dialed there need to be what your trunk is sending.

@Jimmy_K said in No Outbound calls even TRUNK is registered:

They have provide me the prefix 20 + ( international call ) 00 + local number. But it still doesn't work at all

This is not what is in your log. So you screwed something up.

As I said way back in post 5:

@JaredBusch said in No Outbound calls even TRUNK is registered:

This is the call as presented to the trunk.

-- Called SIP/PQT/254202020

Is that a valid telephone number?
Assuming it is not NANPA, then that means it is country code 254, Kenya and the number being dialed is 202020.

Is this how your SIP trunk provider is expecting the call to be presented?

Added: https://mangolassi.it/topic/9519/voip-pstn-gateways

@scottalanmiller said in Elastix login page gives error HTTP ERROR 500 when login:

@dbeato said in Elastix login page gives error HTTP ERROR 500 when login:

@denny said in Elastix login page gives error HTTP ERROR 500 when login:

systemctl stop firewalld disable selinux find /var/www/html/var/ -type d -exec chmod 755 {} ; find /var/www/html/var/ -type f -exec chmod 755 {} ;

🙂

Way to necro a post, security and firewall ar of the upmost importance. No system ever needs to be setup open this way....

I think that he was being tongue in cheek.

Okay, I didn't get it LOL

@JaredBusch Yuck, thanks for the "stay away from".

@scottalanmiller said:

@Dashrender said:

@scottalanmiller said:

Sounds like your Mitel just isn't up to snuff if it requires that and you are using a kludge to get around a hobbled system.

Now you're assuming facts not in evidence. I have no idea if the Mitel can have a one button transfer to a conference bridge (more likely two button, conference and the conference location).

As for the setup, I've never used a setup with a one button transfer to a conference bridge, so I couldn't reference it. I learned something 🙂

So if you haven't used the one button transfer, and you have a Mitel, is it because you've just not bothered to use it, even though it was the driver that brought you to the Mitel, or do you feel that the Mitel is not up to snuff? Or is there a third option I am missing?

The company was using Inter-Tel (bought my Mitel) when I joined the company. Conferencing at the phone level I'm sure was not why they went with Inter-Tel - it's just a feature they discovered and continued to use.

For those stumbling in here down the road, the current consensus is that FreePBX is the better option for your new PBX. Elastix 4 did release, but is not being updated enough to remain a contender at this point.

Cool, thanks.