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Today while doing a little support I had this strange issue: if a user dials out through a carrier, he/she will listen to the ringback tone right away; but if this same user calls to another extension he/she won't listen right away. Instead, you will have a variable dead air and then suddenly, the callee will answer.

When a caller listens to a ringback tone, it means that the callee has been reached and it has answered with a SIP signal. Usually, the normal SIP flow is as follows (it could vary):

  1. Caller sends the INVITE to the switch
  2. Switch sends the INVITE to the callee
  3. Callee answers with a 1XX answer (could be 100 or 183) to report it started the ringing
  4. Switch sends a 1XX answer to the caller followed by an RTP flow of the ringback tone
  5. Callee answers
  6. The switch sends a 183 (session in progress) followed by the voice RTP flow
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High Availability (also known as HA) is the capability of a system to remain online regardless the adversary events that might happen. Then, we will state that availability is the characteristic that concerns a service to be reachable when it is needed. And as you should guess, availability can be measured using percentages (from 0% to 100%); of course, we all know the more close to the 100%, the more expensive it is to deploy a system like that. In the folklore when you say a service has 3 nines of availability, you mean 99.9%., 

As a security consultant and CISSP certified, I have not found any hard definition that states how many 9's you should have in order to claim you have high availability. In my experience, people start calling high availability when you talk about 3 nines or better. But this is only a feeling.

When someone says in the cloud, it is a very gray term that means other's computer. When we speak about servers, we think about VPS'es in the cloud. A name such as Digital Ocean or Vultr jumps right away to my mind. If you are looking forward to having a non-expensive, reliable system in high availability with some load balancing, this article will help you to understand how this works.

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Today, I am publishing the RPMs for SNGrep 1.4.4 All you need to do is to add my OKay's RPM repository, and just install it. SNGrep is a tool for displaying SIP calls message flows from the terminal. It supports live capture to display real-time SIP packets and can also be used as PCAP viewer.

If you wonder what is the big difference from my RPM's, these have all the options enabled as much as possible. This RPM enables the HEP/EEP protocol, very handy if you want to use it to interact with Homer.

RPM's are available for Centos 6 and 7. And you can find it if you type yum search sngrep.

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Hi there. So this is the deal. As many of you know, I have been running a Billing for FusionPBX Funding to convert the code to Opensource. As a special, and a win-win to everybody, I have lowered down the funding goal from 22400 CAD to 7000 CAD (68.75% off). The challenge is the following: if by December 31st, 2017 23:59:59 this new low goal is reached, the source code will be published by January 6th, 2018 latest. If the goal is not reached the original goal will be restored on January 1st, 2018.

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The MOS is a measure to quantify the quality of the call based on the percentage of lost packets and the length of time they represent. It is not very clear how it is calculated, fortunately, FreeSWITCH does that for us.

This article shows how to configure Nagios to make a measurement of the MOS. In order to continue, you understand you must have installed a FusionPBX deployment, and you understand as well the concepts of FusionPBX. Your Nagios deployment needs sufficient permissions to reach the  MySQL / MariaDB database your VoIP server is using.


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