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Freeswitch IP Telephony (Zenitel Connect Pro)

From Zenitel Wiki

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Introduction

Freeswitch is an Open Source IP Telephony software. Zenitel Connect Pro can integrate to Freeswitch using a SIP trunk.

Licensing

Integration using SIP Trunking is a licensed feature. The license has item number 1002720100, ZCL-PBX, iPBX, per defined trunk. A single license is required per SIP trunk configured.
All defined trunks together support a total of 8 simultaneous calls.

Capabilities

  • Bi-directional Audio Calls
  • Bi-directional Video Calls
  • TLS transport for secure data transmission
  • SRTP transport for encrypted audio and video transmission

Configuration in Zenitel Connect Pro

Please refer to SIP Trunk (Zenitel Connect Pro)

  • From the Devices and Connections menu, select External Communication
  • Press the + icon for External iPBX communication to add a new line
  • Choose a Prefix number to access the Trunk
  • Give the trunk a name
  • Enter the Peer IP Address
  • Choose the Call Service to define users allowed to access the trunk
  • Set the SIP Transport to TLS
  • Enter the pre-defined Username and Password. This must match the configuration in Freeswitch
  • Set the Media Encryption to SRTP
  • Press the Tick icon to insert the Trunk

Configuration in Freeswitch

In the relevant SIP Profile, insert the following Gateway details 

<gateway name="<Name>">
<param name="username" value="<username>"/>
<param name="realm" value="<ZCP IP Address>"/>
<param name="from-user" value="<username>"/>
<param name="password" value="<password>"/>
<param name="register" value="false"/>
<param name="register-transport" value="tls"/>
</gateway>


In the relevant Dialplan, insert the following call routing details. Note that this provides for 0 + 3 digits as the access code and extension number within the Zenitel Connect Pro.

<extension name="Zenitel Connect Pro">
<condition field="destination_number" expression="^0(\d{3})$">
action application="set" data="effective_caller_id_number=${caller_id_number}"/>
action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/>
action application="set" data="rtp_secure_media=mandatory"/>
action application="export" data="rtp_secure_media=true"/>
action application="set" data="rtp_secure_media_outbound=true"/>
action application="bridge" data="rtp_secure_media=true,sofia/gateway/<name>/$1"/>
</condition>
</extension>


Due to the formatting of the XML and incompatibility with the Wiki, please add the < to the start of each action application command line.

From the Freeswitch console, reload the xml data and rescan your relevant profile.

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