I'm going to try this out. Looks interesting.

Been working with Yealink support on this and after confirming this happening with the T46G and the T42G phones, their support upped the severity and I got an answer back to add the below setting to the configuration files.

transfer.hang_up_after_success_trans = 3

I did so on the T46G, disabled the custom firewall rules, and had the operator repeat the transfer.

No ghost ringback.

I'll leave the custom firewall disabled and see if it stays good all day.

I'll test in more detail over the weekend or on Tuesday, whichever time I have a tech on site.

@JaredBusch said in Wazo 16.16 First Look:

Not seeing anything in the Wazo documentation on how to set this up either.

In the early release, definitely is looking like it is a bit hard to use. It's early though, hopefully that stuff will follow soon.

@scottalanmiller fwconsole starts asterisk, and it makes sure it's started and running with the correct UID

Or it should. It did last time I checked 8)

Edit: Saying that, I believe I was the last one that messed with that code, so if it's NOT working, it's probably my fault.

https://github.com/FreePBX/framework/blob/release/13.0/amp_conf/htdocs/admin/libraries/Console/Start.class.php#L147

@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.

Thanks Jared, that was informative video about VOIP.

@scottalanmiller So since I posted the OP, I found out what the issue was. (or at least eliminated it by changing a setting.) On old versions of FreePBX, in Advanced SIP settings, in order to enable TCP connections, you had to manually type:

tcpenable =yes

In new versions, there is now a button to enable TCP. Having both settings enabled seems to cause these issues. Most likely, the config file for sip.conf gets messed up having these similar/identical settings. Removing the old entry eliminates the errors.

@quintana said in XiVO - Open Minded Telecom System:

@scottalanmiller Can you give me more detail. How did you install it?

It's on the other thread as well, but it is just a generic ISO based install on Vultr.

@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.

I think that we need a post highlighting the new phones, too!

@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)

@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.

Asterisk AGI?

As with any other language, it's learning by doing IMHO. Define something you want to accomplish, learn the basic syntax and just try to get the first steps done.

@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?

@aaronstuder said in 8 port FXO gateway needed:

@JaredBusch sent via email. Sorry.

I'm a little frustrated on why this thing is not working. Not your problem. thanks for sending your config for comparison.

Circling back to this... I need more damned time in a day....

As far as I can tell the firewalls are all correctly forwarding traffic.

WTF

Very nice.

@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.)

Ah ha, okay. Yeah, just copy/paste into crontab, should be all set.

You'll want to run manually first and see if the emails come through. I tested on a FreePBX system and it went straight through to my Office 365 no problem.

If you are on FreePBX, you will need mailx installed. All dependencies are met by a default install. It's a tiny binary package.

yum -y install mailx

That is what provides the mail command.

@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.

@AlyRagab said in Using Rsync to Replicate Elastix 2:

all passwords are matched

And you've tested them with the MySQL client, for example?

Long ago I had an article on this on my own blog but that is no longer up, so I am copying it here...

https://web.archive.org/web/20150221062029/http://www.scottalanmiller.com/linux/2012/09/02/improving-elastix-memory-usage/

The default installation of Elastix has more services running than are typically needed or desired on a PBX. These services eat far more memory that is necessary and can very easily be cleaned up to improve memory utilization.

First we will stop a series of unnecessary services from starting at boot time (this will disable shared storage, local email handling, new hardware detection, etc. so be aware that this does stop some things but any service that proves to be needed is trivial to re-enable.)

chkconfig nfslock off chkconfig cyrus-imapd off chkconfig iscsi off chkconfig iscsid off chkconfig netfs off chkconfig kudzu off

Further, if your system is like mine you likely use the web server very lightly but will find that the default configuration of Apache is set to spawn, by default, eight processes. This is far too many for a normal deployment. Each process uses memory. For an average deployment of Elastix, three is more than enough. You need only raise this number if web performance suffers. This will not impact telephony performance regardless.

In the file /etc/httpd/conf/httpd.conf we need to edit the section:

<IfModule prefork.c> StartServers 2 MinSpareServers 2 MaxSpareServers 8 ServerLimit 256 MaxClients 256 MaxRequestsPerChild 4000 </IfModule>

to something more like this:

<IfModule prefork.c> StartServers 3 MinSpareServers 2 MaxSpareServers 10 ServerLimit 256 MaxClients 256 MaxRequestsPerChild 4000 </IfModule>

You can wait for the system to reboot or restart Apache manually:

service httpd restart

And finally, to control swapping activity on the box, assuming that you want to avoid swapping when unnecessary, which I do because my box is virtualized, simply add this line on to /etc/sysctl.conf:

vm.swappiness = 10

You’ll want to test that number carefully. A setting of “10” is quite standard for virtualized systems. The default is “60”. For a physical deployment the higher value is likely better as it allows CentOS to make better decisions about how to utilize memory for maximum throughput. But on a virtualized system we really want to avoid, typically, any additional contention at the storage IO layer.
[Testing on Elastix 2.0 and 2.3]

@JaredBusch said:

@art_of_shred said:

I'd say that is a good bit of information to have, but who is going to be moving PBX VM's midday? :P

For me it was good to know that it works that smoothly, not because I plan to ever move one during the day. But because if I HAVE to move one during the day, I will be confident that it will work.

You did notice I up-voted your post? I was just messing with ya. Yeah, that was a valuable test, for sure.

@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.

@JaredBusch said:

@scottalanmiller said:

That makes sense. This is a FreePBX install from FreePBX ISO as far as I can tell from looking at it and all files come from the Schmooze repos and it is causing it to be very out of date compared to standard CentOS 6.

Wait, what? Does the FreePBX distro change what repos yum uses for base CentOS updates?

Yeah, it uses Schmooze repos instead of CentOS ones. I can change that, of course, but by default...