VOIP Archives | SysAdminMan

Monitoring your Peers (Asterisk extensions) and Trunks

25 February 2015JonAsterisk, Trixboxasterisk, extension, freepbx, pbx in a flash, sip, trixbox, trunk, VOIP

As an admin for a telephone system, possibly one of the most useful things you can do is monitoring your peers and trunks. There are 2 great reasons you should do so: 1. You can respond to and resolve issues with your system before your users know about it, and you can be in the know if someone reports “none of the phones are working” when in fact only 1 or 2 are not working 2. You can actually know when there is a problem with the system – where you otherwise might not know there is a problem until someone…

Test your broadband connection for VOIP

23 March 2011MattVOIPbroadband, test, visualware, VOIP

Most broadband connections work really well with VOIP now but any jitter or packet loss will ruin your call quality. Here’s a neat tool that checks your link for latency, packet loss and jitter. Bear in mind that if you’re running the test on a computer/laptop that’s connected via WiFi you are likely to get much worse results that a physically cabled connection. Running VOIP over WiFi is definitely not recommended.

No audio with certain Asterisk calls

11 December 2009MattAsteriskasterisk, no audio, sip, VOIP

I had an unusual problem recently with certain calls going to the PSTN via a SIP provider. The call would connect but with no audio at either end. I’ve seen this lots before and is often caused by NAT or a firewall blocking the audio stream but that wasn’t the cause this time. The problem was caused my trunk only being setup to allow the ulaw codec (allow=ulaw on the trunk). What I think was happening was that my provider was accepting, and connecting, the call but then when it tried to hand the call off to it’s upstream provider,…

asterisk.org gets a facelift

19 October 2009MattVOIPasterisk, VOIP

The home of Asterisk has had a nice makeover. With well over 1 million downloads already this year it is definitely a major player in the VOIP space. Check it out here – www.asterisk.org

Using an Iphone 3G with Asterisk and SIP

7 July 2009MattAsterisk3g, apple, asterisk, incoming calls, iphone, nat, router, sip, sipphone, VOIP, wifi

I’ve been testing an application called SipPhone for the Iphone (http://www.vnet-corp.com/iSip.htm) It works over WiFi only (obviously, as the mobile providers don’t want you using VOIP rather than paying for their minutes!) and it works pretty well. It was easy to get it registered as an extension on my Asterisk server and start making calls as you would with any other SIP or softphone. The quality was very good although it will obviously depend on how good your Wifi connection is. I had some trouble getting it to stay registered with my Asterisk server although that will be to do…

Calculating bandwidth for Asterisk calls

1 May 2009MattAsteriskasterisk, bandwidth, bandwidth calculator, elastix, freepbx, g 711, pbx in a flash, sip, trixbox, ulaw, VOIP, voip provider

One of the things you need to do when looking for a server to run Asterisk on is figure out how much bandwidth you need for the number of concurrent calls you’re expecting to have. A great tool for this can be found here – http://www.asteriskguru.com/tools/bandwidth_calculator.php Just set the codec you’re going to be using (check with your VOIP provider – g.711/ulaw is usual and the highest quality), the connection type (usually SIP or IAX2 with Asterisk) and the number of concurrent calls. It will then display the bandwidth required for that many calls. One thing to watch out for…

Least Cost Routing (LCR) with Asterisk and A2Billing

22 April 2009MattA2Billingasterisk, freepbx, import, lcr, least cost routing, VOIP

A2Billing is a great piece of call billing software for Asterisk. It can be integrated with FreePBX and used in lots of different ways. I run my own Asterisk system with FreePBX and use a2billing to do least cost routing. It’s possible to import rate tables from many different voip providers and then let a2billing route the call based on the cheapest route. One of the things you need to be aware of is that a2billing will route based on the ‘best matching’ rate. So, lets say you are trying to call +17061234567 and you have rate cards for 2…

Getting started with FreePBX – Part 6 Cheap phone calls using DISA and Callback

1 March 2009MattFreePBXasterisk, callback, cheap, configure, did, disa, extension, freepbx, import, inbound, international, ipkall, menu, number, password, server, setup, simple, VOIP

One of the great things about voip is that you can make international calls at local rates. Combine that with Asterisk/FreePBX and you’ve got the ability to make cheap international phone calls using your mobile phone. To do this we’re going to setup DISA (Direct Inward System Access). This will enable us to ring our Asterisk server, get a dial tone and then dial back out again. Then I will show you how you can combine this with callbacks if that works out cheaper for you. Installing the modules First we need to install the DISA (if it’s not installed…

Hackers targetting Asterisk boxes

2 December 2008MattAsteriskasterisk, freepbx, hacker, password, secret, sip, VOIP

I saw the first ‘externsion scan’ of my Asterisk box this week. That is, an external server tried to register as an extension, starting at extension 100 all the way up to extension 999. I’m assuming if they had found a valid extension number then this would have been been followed by a brute force password (secret) scan. This is an interesting article explaining the problem a little more – http://michigantelephone.wordpress.com/2008/11/28/why-didnt-freepbx-developers-implement-important-security-patch/ If you’re running Asterisk (and FreePBX) then the least you need to do is make sure that you’ve got pretty strong passwords for your extensions.

CallWithUs launch UK based SIP server

25 November 2008MattVOIPasterisk, callwithus, VOIP

*** UPDATE 23/6/09 – While callwithus still have a UK sip server you should use the server ‘sip.callwithus.com’ in your configuration settings. Check the callwithus website for details. If you’re based in the UK or Europe and looking for a cheap ITSP (VIOP provider) it might be worth looking at CallWithUS as they’ve recently launched a UK based SIP server. As well as the US based servers sip.callwithus.com, east.callwithus.com and west.callwithus.com you can now use uk.callwithus.com. I now get sub 6ms pings from my Asterisk server in BlueSquare to the CallWithUs server. I’ve been using CallWithUs for a while now…

Tag: VOIP