Linux Quick Install Guide
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- 1 Introduction
- 2 Who This is For
- 3 Prerequisites
- 3.1 Mandatory
- 4 Download
- 5 Generate Installation and Configuration Files
- 6 Compile the Source
- 7 Install FreeSWITCH™
- 8 Start FreeSWITCH™
- 8.1 Test a SIP Phone
- 8.2 Troubleshooting
Introduction
This is a generic quick install guide for Linux and Unix.
Who This is For
Experienced Linux/Unix users familiar with FreeSWITCH™ installation or want a quick overview without reading the Installation Page.
Prerequisites
Mandatory
These mandatory prerequisites provide for compiling the standard FreeSWITCH™ installation and test the supplied configuration and sample IVR. They are sufficient for many production systems.
- GIT or WGET
- AUTOCONF
- AUTOMAKE
- GCC-C++
- LIBJPEG-DEVEL Used by mod_spandsp for basic codecs
- LIBTOOL
- MAKE
- NCURSES-DEVEL
Optional
There are optional modules that can be built and if so may require one or more of the following;
-
libtiff for fax support
-
ODBC or UNIX-ODBC and ODBC-devel see the ODBC page for information
-
OpenSSL (libssl-dev / openssl-devel) for SIP SSL & TLS and Dingaling
-
python-devel for the python interface
-
libzrtp ZRTP encryption support, see the FreeSWITCH™ ZRTP IS NOW DEPREICATED and will be removed out of the docs
Download
Selecting a Version
There are two versions of FreeSWITCH™ to choose from: the stable version recommended for production systems, and _master_containing the latest fixes and features used duringdevelopment. The stable version is recommended unless you have experience with FreeSWITCH™. Additional Information is located at Source Options.
This documentation is tested with the master version to keep it as recent as possible and to provide feedback to the developers. If there are problems the developers may request that you install the master version for diagnosis.
Download Using Git Clone
Download FreeSWITCH™ by switching to the src directory and invoking one of the "git clone" commands below. The git command will create the freeswitch directory in src (/usr/local/src/freeswitch) and download the FreeSWITCH™ source files.
cd /usr/local/src
git clone https://github.com/signalwire/freeswitch.git
Generate Installation and Configuration Files
Create Configuration Files
cd freeswitch # <- the new freeswitch directory created by the git clone process
./bootstrap.sh
./configure
Edit modules.conf
This is optional but consider editing modules.conf.
Compile the Source
make
Install FreeSWITCH™
make all install cd-sounds-install cd-moh-install
Start FreeSWITCH™
Make sure no other instance of FreeSWITCH™ or Asterisk is running on the same computer. Run command:
/usr/local/freeswitch/bin/freeswitch
Test a SIP Phone
Configure a SIP phone or softphone with the IP address of the FreeSWITCH™ computer and user "1000" and password "1234". The default FreeSWITCH™ configuration provides pre-defined definitions for extensions 1000-1019, all passwords are 1234.
- Dial 9664 to hear music.
- Dial 5000 to test the sample IVR
- Configure a second SIP phone as user 1001, dial 1001 from 1000, and 1000 from 1001 to test between phones
- Check out more in Configuring FreeSWITCH at Some stuff to try out!
Troubleshooting
- If there is no communication between the phones and FreeSWITCH™ or between phones disable the firewall on your Linux server. If testing is successful activate the firewall and open only the specific ports needed. For more information about firewalls and ports to open, see the Firewall page.
- If you suspect networking issues look into networking tools such as Wireshark or TCPDump.