Showing posts with label asterisk. Show all posts
Showing posts with label asterisk. Show all posts

Thursday, 22 October 2015

Asterisk Wideband Music on Hold

15:24 Posted by Jurgens Krause , , , , , , , , , No comments

The default music on hold included with Asterisk leaves somewhat to be desired, and getting high quality replacement music in the correct formats can be a challenge. I have not managed to figure out how to convert to G722, but you can easily convert to the native Asterisk SLN format with the following commands:

Monday, 29 June 2015

Wednesday, 17 June 2015

Yealink T21P Review

11:55 Posted by Jurgens Krause , , , , , , 2 comments
The Yealink T21P is an entry level IP Phone. It is in the same product class as the Snom 300, but comes in at about 20% cheaper.

The T21 comes in two models, the T21P being the POE version, which is the one I will be looking at. The only difference between the T21 and the T21P is in fact that the T21P supports 802.11af Power over Ethernet.

Thursday, 28 May 2015

Top Free Windows Softphones Compared

14:57 Posted by Jurgens Krause , , , , , , , , No comments
Photo by Irita Kirsbluma

Which softphone you use may depend largely on why you want to use a softphone in the first place. For example, if you are looking for an alternative to FOP (Flash Operator Panel) then I would suggest the excellent Dialplate application, but if you are just a normal desktop user looking for a cheap way to make use of your company's existing VOIP infrastructure, then have a look at these.

When comparing features I will only be looking at the features available in the free version, and not the premium one where applicable.

Friday, 8 May 2015

Asterisk Dialplan Show and Tell 1

14:57 Posted by Jurgens Krause , , , , , , No comments

NEW FEATURE ALERT!!

I hope to make this a regular feature on my blog. If you have an awesome dialplan that you would like to share, please let me know and I will analyze and feature it here!


My first share is the dialplan that I use for clients who need added features beyond basic calling. This includes call diverts as well as voicemail.

Thursday, 7 May 2015

Monday, 9 March 2015

Asterisk blind transfer callback

10:30 Posted by Jurgens Krause , , , No comments

One of the problems with Asterisk, is that, when you blind transfer a call to an internal extension, and the remote side never answers, the call does not automatically go back to the caller. The following piece of dial plan logic should replace your normal internal call logic.

Thursday, 18 December 2014

Asterisk Out of Office Recording Extension

12:34 Posted by Jurgens Krause No comments
My clients sometimes need to put temporary messages on their phone lines. For example during the Christmas holidays or if everyone is away for a company retreat.

This is a simple edit to your dialplan, so it should not be too difficult.

Wednesday, 26 March 2014

Asterisk PIN based dialing

10:31 Posted by Jurgens Krause , No comments

As an Asterisk administrator, I often have clients requesting the ability to have users enter a PIN before dialing. This is useful on factory floors where users roam between different phones, but still need to be held accountable for their calls. Apart from just authenticating the user, the dialplan also has to ensure that the account code is correctly assigned in the CDR.
To this end Asterisk provides the AUTHENTICATE dialplan application.

Thursday, 13 March 2014

Asterisk - Outbound Whitelisting

16:07 Posted by Jurgens Krause , 4 comments
There are numerous ways to handle outbound whitelisting on asterisk, but I needed a solution where the end user could simply ftp a text file to the server to update the whitelist.

Tuesday, 11 September 2012

Asterisk Call Groups

18:31 Posted by Jurgens Krause , , 1 comment
A large part of my work centers around managing Asterisk based Virtual PBXes. Now, I know very little of Asterisk, so as I learn I will share what I learn in the home of helping out other people who struggle.

Intercepting Calls on Asterisk (or CallGroups) as the clever people call it.

Say that your client has a switchboard, and six extensions in the office. Someone calls the office, knows the extension they want to reach and by the magics of Asterisk IVRs they manage to connect themselves directly to Joe. Now, Joe is at the water cooler chatting with a colleague and is thus unable to hear his phone ring. Martha at the switchboard however does hear it ringing, and feels the urge to answer it. How would you set up the Asterisk server to allow this?

Well, it is really quite simple:

What you need to do is assign a callgroup and a pickupgroup as shown below. But first, what is a:
callgroup:
A callgroup is simply a way of grouping extensions that share some common function, ie. all the extension in the "Sales" department

pickupgroup:
The pickupgroup directive, when assigned to an extension, shows which callgroups' calls can be intercepted or picked up by the relevant extension

Practical:

Open up sip.conf (usually it can be found in /etc/asterisk/sip.conf)

Here you will find many sections, called "Contexts"

Find the context matching your Switchboard extension ie. [100]. It might look something like this:
-------
[100]
type=peer
host=dynamic
context=international
disallow=all
allow=g729
qualify=yes
secret=superawesomepassword
nat=yes
canreinvite=no
canredirect=no
----------

To the bottom of this add the pickupgroup directive:
pickupgroup=1

This tells Asterisk that extension 100 can intercept any call ringing on any phone in callgroup 1

Now, no extensions have yet been assigned to callgroup 1 so to remedy that, find  the context matching the extension that you want to place in the callgroup and simply add the callgroup directive, like so:
[101]
type=peer
.
.
.
callgroup=1

The switchboard (ext 100) will now be able to pick up a call ringing on extension 101 simply by pressing *2# on their handset (sometimes just *2)



And there you go, Asterisk callgroups in a nutshell. Of course there is much more you can do with callgroups, and we will take a look at that in the future.