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Asterisk configuration

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Asterisk Configuration(CHAN_SIP)

Configuration with UDP/TCP transport protocol and video support

[general]
context=default
bindaddr=0.0.0.0
videosupport=yes
port=5060 

//Extension
[5001]
type=friend
host=dynamic
secret=password
disallow=all
allow=ulaw,alaw,g722,g729

Configuration with TLS and SRTP

[general]
context=default
bindaddr=0.0.0.0
port=5061
transport=udp,tls
encryption=yes
videosupport=yes
tlsprivatekey=/etc/asterisk/keys/default.key
tlscertfile=/etc/asterisk/keys/default.crt
tlsbindaddr=[::]:5061
tlsdontverifyserver=no
tlsclientmethod=tlsv1
tlsenable=yes

[5001]
type=friend
host=dynamic
secret=password
disallow=all
allow=g722
allow=g729
allow=ulaw
allow=alaw

Turbine station configuration

Configure Station mode

  • Click Main Settings from the left menu and select 'Use SIP.'
  • From the Model: drop down menu choose TCIS 1-3,TCIS 4-5, TCIV-3/TCIV-6, TFIE 1-2 or Mini (TMIS-1) depending on which model type you are configuring.
  • Click Save when done. A screen will appear (not shown) to confirm the setting, click Apply and Turbine will reboot.

Sip station main.jpg

Configure SIP Settings

Account Settings

Click on SIP ConfigurationSIP Settings and configure the following in the Account Settings section.

Turbine stations support UDP, TCP and TLS configuration options.

For TCP and TLS options it is necessary to set Outbound Proxy to have the same value as Server Domain.
Account settings sip.jpg

UDP outbound transport
  • Display name: Enter the desired name.
  • Directory Number (SIP ID): Enter a user extension administered station extension section (sip_additional.conf).
  • Server Domain (SIP): Enter the IP address of Asterisk.
  • Authentication User Name: Enter a user extension administered in station extension section (sip_additional.conf).
  • Authentication Password: Enter the Secret from station extension section (sip_additional.conf).
  • Outbound Proxy (mandatory for TCP and TLS, optional for UDP): Enter the IP address of Asterisk and 5060 as the Port for UDP/TCP

Turbine sip udp.png

TCP outbound transport
  • Display name: Enter the desired name.
  • Directory Number (SIP ID): Enter a user extension administered station extension section (sip_additional.conf).
  • Server Domain (SIP): Enter the IP address of Asterisk.
  • Authentication User Name: Enter a user extension administered in station extension section (sip_additional.conf).
  • Authentication Password: Enter the Secret from station extension section (sip_additional.conf).
  • Outbound Proxy (mandatory): Enter the IP address of Asterisk and 5060 as the Port for TCP


Turbine sip tcp.png

Secure SIP configuration with Secure RTP
  • Display name: Enter the desired name.
  • Directory Number (SIP ID): Enter a user extension administered station extension section (sip_additional.conf).
  • Server Domain (SIP): Enter the IP address of Asterisk.
  • Authentication User Name: Enter a user extension administered in station extension section (sip_additional.conf).
  • Authentication Password: Enter the Secret from station extension section (sip_additional.conf).


  • Outbound Proxy (mandatory): Enter the IP address of Asterisk and 5061 as the Port for TLS


  • SIP Scheme: Choose sips from the drop down.
  • RTP Encryption: Select srtp_encryption from the drop down.
  • choose SRTP crypto type from the drop down and TLS private key

Turbine sip tls srtp sips.png

How install certificate on Turbine station

Upload Certificate

Relay settings

Relay Settings (Pulse/SIP)

Configure Direct Access Key

Direct Access Key & Ringlist Settings (Pulse/SIP)

Verification Steps

Verify Turbine SIP Registration

From the Stentonfon web interface, select Information from the left menu. Verify that the Registration state shows Registered. Place a call to another endpoint to verify basic call operation.
Turbine registration.jpg

Verify registration on Asterisk

The following commands can be used to verify registration.

sip show peers
sip show users
sip show DIRNO

Verify Successful Calls

Place a call to and from the Turbine endpoint. Verify 2-way audio is heard and validate call terminates successfully.


Related Articles

This section references the Asterisk and Zenitel product documentation that are relevant to these Application Notes.

These documents form part of the Asterisk reference documentation suite.

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+Started

The Zenitel Turbine documentation can be found at http://www.zenitel.com.

[1] A100K11013-Pulse-Getting-Started.pdf.