I do notice that Linphone provides real time feedback no individual calls, which is nice.

@tm1000 said in FreePBX 13 No Conference Room App:

This was happening because for some reason the conbridge.conf file (or an include therein) was not being created correctly and thus because conf files were missing confbridge was never loaded. I believe this was because conferences was not originally included in the tarball for 13 but we fixed this so it should be OK???? Not sure...

It was fixed this week? It was just six days ago that we saw it, but it's not been re-run since then.

@fuznutz04 @Reid-Cooper I'm logged in as the admin, never logged in as otherwise on this phone.

@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.

@Dashrender said in 3CX Linux Beta:

@scottalanmiller said in 3CX Linux Beta:

@Dashrender said in 3CX Linux Beta:

@JaredBusch said in 3CX Linux Beta:

@Dashrender said in 3CX Linux Beta:

@scottalanmiller said in 3CX Linux Beta:

@Pete-S said in 3CX Linux Beta:

So when you push a button and have all your phones upgraded to the latest 3CX/Yealink firmware you pretty much know that everything is going to work afterwards.

Now that is a pretty solid value that could make up for quite a bit of cost.

I was wondering how much value this provided. I've done some firmware updates via FreePBX - been a while though - and I didn't find it very easy, though not end of the world hard either.

A push button solution though would definitely be nice. Anyone know if the endpoint module for FreePBX is there yet?

Then you were doing it wrong. Assuming you have the paid EPM, it is all just built in. Always has been.

If you were going without the paid EPM, then it is still easy if you have setup auto provisioning manually during initial setup.

I've never used EPM.

That's the part of FreePBX that does that stuff.

Yes I know... JB has always suggested against it's value in the past.

Because setting up manually once does take some skill, but after that, it is simple file replication.

I am trying to set up an incoming access number and use it in a pinless scenario. I don't actually have it connected to any phone system per se.

@thwr said in MC16: VoIP with Jared Busch Session:

@JaredBusch said in MC16: VoIP with Jared Busch Session:

@scottalanmiller said in MC16: VoIP with Jared Busch Session:

@JaredBusch said in MC16: VoIP with Jared Busch Session:

@thwr said in MC16: VoIP with Jared Busch Session:

@scottalanmiller said in MC16: VoIP with Jared Busch Session:

Myths of VoIP:

No 911 Is not secure Requires dedicated Internet Needs special network configuration have to buy all new hardware (or any new hardware!)

Seriously? ...

Serious that those are myths of VOIP? yup. I hear those all the time.

Same here.

@thwr Just happened last night in fact when i gave a presentation to a SpiceCorp.

Really sad. But there are so many myths (and half-knowledge) in and around IT...

Seems to be the majority more often than not 😞

@RamblingBiped said in SIP Phone registers no audio...:

Here is the guide I followed to get the Yealink T23G to work with OpenVPN: http://www.sunstatetechnology.com/docs/YealinkOpenVPNGuide.pdf

When I disable NAT on the extension the phone is unable to register.

Something is wrong there, that sounds like the VPN is not working as there is no NAT when there is a VPN.

@JaredBusch said in The High Cost of Unlimited Calling:

@scottalanmiller said in The High Cost of Unlimited Calling:

@NetworkNerd said in The High Cost of Unlimited Calling:

There's also the idea that you can have unlimited concurrent calls and just pay per minute of use while you pay a certain fee for each DID you keep per month. As you said, there is a cap to this "unlimited" number of concurrent calls that gets applied (at least from what I have seen from certain providers).

Sometimes all you can eat is all someone says you can eat.

That's surprising that someone would cap that since normally those are charge per minute and the goal of unlimited concurrency is to encourage you to use as much as possible.

I'm not sure what he is taking about. All the pay per minute SIP trunk vendors I have used have no hard cap on this.

Some like VoIP.ms do put in a soft cap but will raise it to whatever you want.

Their reasoning is to prevent fraud

That I would understand, you don't want someone able to charge tens of thousands of minutes of usage per minute if you didn't know what you were doing. Lots of risk there.

@NetworkNerd & @JaredBusch - We would love to have you both. We can offer you 20% off this year's AstriCon registration with promo code "VUC" at checkout. Otherwise, make sure to check out the Early Bird pricing for next year's event (which ends in late June)

@wirestyle22 said in FreePBX Remote Extension Problems:

@scottalanmiller said in FreePBX Remote Extension Problems:

@wirestyle22 said in FreePBX Remote Extension Problems:

@RamblingBiped Hope this is the solution. I unfortunately have Cisco CM and its my first time managing VoIP

So sorry.

Complication for the sake of complication

And spending money for the sake of throwing away profits 😉

@JaredBusch I really wish I were going 😞

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

@scottalanmiller said

The more that you add, the better T-Mobile gets.

You mentioned, perhaps privately, to look into adding everyone in our company, including family members.

Do people actually do that? To me, it's always seemed like a PITA. Business users, sure. But family members?

@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 🙂

@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.