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FreeSWITCH integration HOWTO

This HOWTO will show you how to:

  • Install FreeSWITCH from scratch
  • Configure FreeSWITCH to bare-minimum level
  • Integrate your SIP provider to FreeSWITCH
  • Configure the bridge to connect to FreeSWITCH
  • Make a call from Matrix to the regular phone network

IMPORTANT: This HOWTO will not make any attempt in creating a secure install, only to give you the ability to test the bridge.

Requirements

  • Debian 8 system
  • Ability to support incoming UDP on port 5080, relayed to the system

Configure

FreeSWITCH

Add the FreeSWITCH repo GPG key for APT

wget -O - https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add -

Add the repos

echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main" > /etc/apt/sources.list.d/freeswitch.list
echo "deb http://ftp.debian.org/debian jessie-backports main" > /etc/apt/sources.list.d/debian-backports.list

Install FreeSWITCH and its Verto and RTC modules, for WebRTC support

apt-get update && apt-get install -y freeswitch-meta-all freeswitch-mod-verto freeswitch-mod-rtc

Change default password

IMPORTANT: If you do not do this, FreeSWITCH will delay each call!

Edit /etc/freeswitch/vars.xml and set another default password password by replacing 1234 on this line:

<X-PRE-PROCESS cmd="set" data="default_password=1234"/>

Add SIP provider

Create a file /etc/freeswitch/sip_profiles/external/default.xml with following content, adapting username, real and password as needed:

<include>
  <gateway name="default">
    <param name="username" value="00123456789"/>
    <param name="realm" value="sip.provider.example.com"/>
    <param name="password" value="abc@abc"/>
    <param name="register" value="true"/>
    <param name="register-transport" value="udp"/>
  </gateway>
</include>

Make SIP provider as default gateway

Edit /etc/freeswitch/directory/default.xml and replace ${{default_provider}} by default (gateway name) in the following line:

      <variable name="default_gateway" value="$${default_provider}"/>

Fix dialplan so it works by default

For outgoing calls, edit /etc/freeswitch/dialplan/default/01_example.com.xml and replace the following line:

    <condition field="destination_number" expression="^(011\d+)$">

with

    <condition field="destination_number" expression="^(\d+)$">

For incoming calls, edit /etc/freeswitch/dialplan/public/00_inbound_did.xml and replace the following line:

    <condition field="destination_number" expression="^(5551212)$">

with

    <condition field="destination_number" expression="^(\d+)$">

Restart Freeswitch

systemctl restart freeswitch

VoIP bridge

Given the following example values, adapt as needed:

  • LAN IP of the FreeSWITCH system: 10.1.2.3
  • New default password in FreeSWITCH: 12345
  • Your Matrix User ID is @user1:domain.tld

In application.yaml, set:

providers.freeswitch.verto:
    url: 'ws://10.1.2.3:8081'
    login: '1000'
    password: '12345'

bridge.mapping.users:
  user1: '1000'

Restart the bridge

Use

Outgoing calls

If you wanted to call 00123456789, assuming you've respected default values in README.md:

  • Create a new room
  • Invite @_voip_00123456789:domain.tld
  • Hit the call button
  • The call should go through

Incoming calls

If your DID number is 00123456789:

  • Call it from a regular phone
  • You will get an invite to a room on the Matrix side, if not in a room already. Accept it
  • Your Matrix client should ring

Troubleshoot

Bridge hangs up call directly

If you see unknown failure: ERROR_CODE for the following codes:

  • INVALID_GATEWAY: The SIP gateway name doesn't match the default_gateway value.
  • INCOMPATIBLE DESTINATION: Your SIP line/gateway is not connected