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Upgrade FreeSWITCH from compiled to package 1.5.8

Installation

First, add repository to system

vim /etc/apt/sources.list

add line

deb http://files.freeswitch.org/repo/deb-master/debian/ wheezy main

Then add GPG key to repository

gpg --keyserver pool.sks-keyservers.net --recv-key D76EDC7725E010CF
gpg -a --export D76EDC7725E010CF | apt-key add -

Now, update repository

apt-get update

FreeSWITCH is ready to install. Just run

Upgrade Siremis 3.2.0 to 4.1.0

Download Siremis 4.1.0

Go to your desired folder (for example /var/www) and download Siremis from site http://siremis.asipto.com/pub/downloads/siremis/.

cd /var/www/
wget http://siremis.asipto.com/pub/downloads/siremis/siremis-4.1.0.tgz

Then extract Siremis folder from archive

tar -xvf siremis-4.1.0.tgz

Folder siremis-4.1.0 appears here.

 

Installing WebRTC2SIP gateway - tutorial

System preparation

apt-get update
apt-get upgrade

As the first step we need to install packages necessary to build the main webrtc2sip gateway:

apt-get install build essential libtool automake subversion git pkg-config screen libxml2-dev / 
libssl-dev libsrtp0-dev

to support for libspeex (audio codec) and libspeexdsp (audio processing and jitter buffer) add

Configuring Kamailio 4.x for WebSocket

Author: Patrik Formanek 2014


This tutorial instruct how to add the WebSocket support for your kamailio SIP server. As the prerequisities we need to have successfully installed and working kamailio server (described within several tutorials in this site, for example Installing Kamailio 3.1 within SIP/Kamailio section of this site).


As the first step we need to install websocket modules:

Kamailio configuration to provide load balancing and failover for media services

In many setups Kamailio is used as a PROXY server that takes care of routing calls to servers providing voice services, e.g. voicemail, IVR or conference calls.

There are a few things an administrator must keep in mind.

Kamailio Call establishment permission rules

This article talks about deploying permission control mechanism for call establishment in Kamailio SIP Proxy.

In many VoIP solutions, it is crutial to deploy numbering scheme and write down rules where users are/aren't allowed to call.
On top of that, a company can allow the people to call outside, for example to PSTN. The rules can change over time as well as the numbering scheme itself.

Installing SIPp 3.2 on Debian Squeeze 6.0.5 32 bit

SIPp version 3.2 hasn't precompiled binary packages. There are some instructions provided at the sipp home page, however small comment about limits.h is missing and therefore the compilation will not be succesfull. Here we provide instructions how to compile Sipp with SSL, pcap play and distributed pauses features enabled.

Install prerequisities

Three pre-requisites are necessary to compile SIPp:

Enabling SCTP support in Kamailio 3.3.x - debian squeeze

Preprepared binary package of the Kamailio server hasn't enabled the SCTP support. You may check this on your running platform both ways:

First running kamailio command with -V options and checking the output for the sctp flag:

kamailio -V

or

kamailio -V | grep SCTP

and if nothing to see, there is no SCTP support enabled.

Second, using sercmd tool, but we have to load and enable ctl module in our kamailio.cfg. Then run

How to install Asterisk on Debian - APT repository

Latest Asterisk packages we should install now directly into our Debian/Ubuntu servers using the apt-get tool. More info at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-APT%28Debian%2FUbuntu%29

 

Instructions Options 1) (debian and Ubuntu)

First, we will import Asterisk project apt key:

Kamailio 3.3 and FreeSWITCH 1.2.2 interconnection for voicemail and conference services on Debian Squeeze (6.0) 64bit - TUTORIAL

This tutorial will, hopefully, guide you on configuration of interconnection between Kamailio and FreeSWITCH. We will use Kamailio as proxy and registrar server and use FreeSWITCH only for services.

 

All of the configuration files that have been changed are part of attachment of this tutorial. In Original.zip you will find the original files and in Modified.zip the modified version. So you can actually compare them side by side. This could be really useful and I hope it will help you in every way possible.

 

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