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?

@scottalanmiller said in 8 port FXO gateway needed:

@JaredBusch said in 8 port FXO gateway needed:

The unit worked perfectly for 2 years (June 2018). At that time it was honorably retired as the site was able to get a solid fiber connection thanks to @Phil-CommQuotes.

Good reminder, I needed to email him today!

I'm here to help! Let's talk!

@Jason said in One way audio on an IAX2 trunk:

Also did you add the network for the remote site in asterisk?

Found this topic again after checking the IAX2 tag.

Thought I would update with the answer.

This was the problem

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? 😛

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...

@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.

@scottalanmiller Ok, I think I got it. Per http://bugs.elastix.org/view.php?id=2549:

Starting with PHPMailer v5.2.7, the default HELO hostname is not set anymore to 'localhost' if not supplied on the Hello() function... This generates that the Elastix remotesmtp configuration checker ends up sending an empty HELO/EHLO command, which makes Gmail (and probably many other SMTP servers) really unhappy.

A possible solution would be to modify /var/www/html/modules/remote_smtp/libs/paloSantoEmailRelay.class.php on line 156:

if(!$smtp->Hello()){

change to:

if(!$smtp->Hello(getHostname())){

If you are using the Multi Tenant version the location is /usr/share/elastix/apps/remote_smtp/libs/paloSantoEmailRelay.class.php

Changed that and it worked. However, whenever I create a new organization and the email is sent to the "admin", I never receive anything... what could it be now? Any ideas? Thanks!

Yes, the number of people who are out there looking for "whatever is called Elastix" is way more than enough to keep the website and brand name around. Elastix has a huge presence in South America and in LATAM countries in general alone.

Sounds like.... fun?