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

@PSX_Defector said:

You can also block it at the SIP level with your provider. Do that along with international dialing, save yourself some headache.

If the SIP provider offers this, definitely. Using a route to do it stops your extensions (if compromised) from using them. Blocking them at the SIP carrier stops your PBX if compromised from calling them.

Neither stops you if the SIP provider is the one making the calls (looking at you, Megapath.)

Roll20.net used WebRTC also. It was really gitchy when i first started my campaign back up in it. though, so we used hangouts for the voice/video.

@donaldlandru said in FreePBX Asterisk Music on Hold Streaming Sources:

Cleaning up my old post and marking it solved. After taking options and cost to the business we ended up staying with the bundled music.

Thanks for circling back 🙂

Thanks everyone!

@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.

It is a sales tactic...

Sounds like you phone is not putting the PC poet on VLAN 1

The one feature that I really loved with 3CX was the incredible ease of setting up soft phones over a VPN. They do that beautifully.

@JaredBusch said:

Great news of course, but I just do not know if I can trust them to keep it up to date.

I'd be curious to know that too, especially after your adventure with yum updates and going from Elastix 2.4 to 2.5.

Digging through my notes I also have a note that Cox had to switch from a Atrica to Alcatel SBC. At least that was their claim. In the end they didn't swap the box and instead made changes to the existing Atrica.

@scottalanmiller said:

They are a Spanish language company based in Ecuador.

I know this.

Any question like this would technically be dependent on the PBX in question and could not be answered generically. That being said, I do not know of any system where this would cause a problem.

@scottalanmiller Yes... Everything works fine when I dialed in and pointed inbound route to an extension, rings and audio is perfect. Web interface, logs, Asterisk functions, modules loaded everything works fine. This is what confuses me the most. Unfortunately I cant just erase the entire server and start from scratch.