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SIP

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Which key types can I use with the SIP virtual port?   |   What is required to configure a SIP server connection?   |   The SIP auto dial keys overwrite a user label once the call is established?   |   What points should I check if a SIP connection is not working?   |   How do I install the Mobile App client?   |   How do I install the Mobile App on an Android phone?

 

What additional functionality is on offer when using SIP with Gemini or Messenger systems?

If the site has a SIP server, for example a Cisco Call Manager or Asterisk server then Gemini and Messenger systems can register with this third party SIP server.

Both the Gemini and Messenger matrices support 32 hardware ports and depending on the interface cards fitted to the matrix, these ports can support the following audio interfaces:

Trilogy panels, 4 wires, MADI* or FXO* analogue telephone lines. In addition to the 32 hardware ports each Gemini/Messenger matrix can also support a further 32 virtual ports. Each assigned virtual port will consume a DSP channel when active, so if virtual ports are to be used, the host must have sufficient DSP to support this.

* Gemini only

Once the host matrix is connected to a SIP server, up to 32 virtual ports per host can be registered with the server, each virtual port having a unique SIP alias supplied from the server. Below are some examples of how SIP virtual ports can be utilised:

SIP virtual port as a virtual FXO port:

A SIP bound virtual port can work in a similar manner to a traditional FXO phone connection. A third party caller can dial the SIP aliases and connect to the Trilogy system or a panel can dial a SIP phone and thus connect the SIP phone to the Trilogy matrix.

SIP virtual port as a virtual 4 wire port:

Once a SIP connection has been established, similar to a 4 wire port on the matrix, the SIP virtual port can be targeted to panel keys in the system and the panel can key to speak/listen to this SIP virtual port.

SIP virtual port as a dial–up IFB port:

The virtual port can be assigned as an IFB source or IFB destination. The SIP virtual port can thus be used for a dial-up IFB function replacing the dial-up IFB using traditional analogue phone lines and FXO cards within the Trilogy matrix.

SIP virtual port as a tie-line for remote site communications:

SIP is a useful tool when connecting to a remote site, be it another Trilogy system or a third party system which supports SIP. The SIP virtual port can be assigned as a dedicated tie-line for panel to panel intercom, when the remote panel is at another site and not part of the Trilogy system served by the local Trilogy Database Supervisor.

SIP virtual port as a conference member:

The SIP virtual port can be assigned as a conference member. The SIP connection can be established by either the SIP member dialling in or a panel dialling out to establish the connection. An automated method available in the Trilogy system is the use of reverse phone control; this feature can be set to automatically connect the SIP client when a conference has active members speaking into the conference.

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Which key types can I use with the SIP virtual port?

If the SIP bound virtual port is to be used in 4 wire, IFB or as a conference member then all of the standard key types are available to be used. If the SIP virtual port is to be controlled by a panel phone key the following three dial keys types are available.

Each phone key type behaves in a similar manner to establish the call, when the key is pressed down selects the dial pad and menu list of pre-stored numbers, including the last dialled number to help automate the dialling process. The incoming call can be set to ring the panel. Once the call is established, the SIP phone key can be put on hold and then any panel can speak to this call as if it is a standard 4 wire.

The SIP phone key can be set to auto dial a pre-set number entered in the Gateway Editor. So pressing the key down auto dials and connects to the remote SIP alias. SIP dialling is reasonably quick (typically a few 100mS depending on the network)

The SIP key can also be set to auto answer, so an incoming call is automatically answered (no ringing) and the callers’ voice is heard without the need to interact with the key. These last two features can be used together to offer an “intercom-like” user experience when connecting to third-party systems.

Phone action, combined listen and speak “hands-free”. On establishing a call the key is latched listen, latched speak. An upwards key press puts the call on/off hold, down key press clears the call.
Phone action, latched listen / latching speak. On establishing a call the key is latched listen, press to speak. The speak mode is either a quick press down to latch or press and hold down for momentary speak. An upwards key press opens a menu and gives the option to put the call on hold or clear the call.
Phone action, latched listen / latching speak. On establishing a call the key is latched listen, press to speak. The speak mode is either a quick press down to latch or press and hold down for momentary speak. An upwards key press opens a menu and gives the option to put the call on hold or clear the call.

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What is required to configure a SIP server connection?

To register a Gemini or Messenger matrix with a third party SIP server, the following information is required:

  • The IP address of the SIP server
  • The coding profile assigned by the server.

The Trilogy system will also need to register each extension or alias. This requires a unique name, user and password from the SIP server.  The SIP server is usually set to provide individual usernames.

  • Name – User name, typically this is the phone extension number registered with the SIP server.
  • User – User name for extension, does not need to be unique.
  • Password – Password for the extension, does not need to be unique.

By default Gemini uses the following ports for the SIP connection:

  • UDP 5060 for the SIP call port, this is configurable in the Trilogy Editor if this needs to be changed.
  • UDP 10000 for the RTP/RTCP port, this is configurable in the Trilogy Editor if this needs to be changed.

An example of SIP Aliases in the Trilogy Gateway Editor > Host Advanced Editor is shown below.

A typical coding profile is G711A with a 10mS packet size.

The default Registrar expiration time is 3600 seconds. Depending on the SIP server this may need to be reduced to 120 seconds.

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The SIP auto dial keys overwrite a user label once the call is established?

If the SIP key label is being overwritten with the auto dial number when the call is established this can be resolved by adding the key name you wish to appear on the key in the phone list.

Change the ‘Name’ entry in the Phone Number Editor to the required name, this will now prevent the auto dial number appearing on the key.

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What points should I check if a SIP connection is not working?

If the SIP Server Authentication / Registration is not succeeding, check the formatting of the User Names, Passwords, Number and Codec type. Any errors or typos will cause authentication and registration to fail.

Is the codec type chosen supported by the SIP server? If the call that is failing this could be a codec negotiation failure.

Check the Registrar expiration time in the Gateway Editor > Host Advanced Editor > Aliases page. The default expire is 3600 seconds; depending on the SIP server this may need to be reduced to 120 seconds.

Does the same SIP number fail each time? This could be due to the SIP alias in question not registering with the SIP server and would suggest an incorrect setting, may be a typo in the number, password etc. 

If you are making multiple SIP calls and it is the last call, regardless of the number dialled that does not connect then check the number of DSP that have been allocated for IP phone connections on the Host Advanced Editor > Audio page. 

 

Each active (dialled) SIP aliases will use an IP phone connection audio DSP channel; the Gemini has a total of 32 or 64 DSP channels, depending in the module installed. A number of DSP must be allocated for the SIP ‘IP Phone’ connections, depending on the number of concurrent calls expected to occur in the system. If for example you allocate 18 DSP channels for the IP phone connections, this will allow a maximum total of 18 simultaneously active SIP calls to be made on this Gemini host.

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How do I install the Mobile App client?

The Trilogy Mobile App software is included as part of the ‘TrilogyV5Setup.exe’ installation file supplied on the CD with the equipment. Before commencing, check that the CD installer version matches he current version of software reported in the Gateway Editor > Help About dialogue as the software may have been updated since the original installation. If the versions are different, contact Trilogy Support and we will supply a link to download the matching installer version.

Each active Mobile App connection will use a SIP IP phone connection audio DSP channel; the Gemini has a total of 32 or 64 DSP channels, depending on the module installed. A number of DSP must be allocated for the Mobile App, depending on the simultaneous connections expected. If for example you allocate 2 DSP channels for IP phone connections, this will allow a maximum total of 2 simultaneously connected Mobile Apps to this Gemini host.

The Mobile App is a key coded option, so first enter the new key codes for the Gateway Editor and Virtual Panels in Gateway Editor > Setup > Enter Software Keys. If you do not have a key but wish to purchase one please contact Trilogy Sales.

 

Next run the ‘TrilogyV5Setup.exe’ installation file on the PC hosting the Gateway Editor application.

On the “Installation or Configuration” screen choose the ‘Installation’ option.

 

On the selected components screen, check the ‘Virtual Panel Web server’ option.

 

Click ‘Next’ through the remaining screens of the installer.

Once the installation is complete you will see the Virtual Panel Web Server icon  on the system tray. Double click on this to open the Web Server GUI.

In the Web Server Setup > Database Settings dialogue, set the Database Supervisor Address to the fixed IP address of the Database Supervisor. The web server is normally installed on the Database Supervisor PC, so this address is usually the fixed IP address of this PC.

 

In the same dialogue, click on the “Web Server” tab and set the Interface to the Database Server IP Address entered in the previous step.

 

Now launch the Gateway Configuration Editor:

Each Virtual Panel will require a SIP connection. In the Enterprise Advanced Editor > SIP Connections page add the number of connections required (and allowed by the license).

 

Next add the Mobile App Virtual Panels in the Host Advanced Editor > Virtual Ports page.

Set the ‘Type’ to ‘Virtual Panel’ and add the 5/8 character name for each.

Once the virtual ports are assigned, a Vpan entry will appear in the host tree view, select the virtual panel in the tree view to add keys.

Next open the Host Advanced Editor > SIP Aliases tab.

  • Add a SIP alias for each Mobile App (Vpan). Enter the ‘Server’ IP address for each entry.
  • Under ‘Preferred Codecs’, set Profile 1 to Codec 4 and Profile 2 to Codec 5.
  • In the ‘VP connect’ entry add the virtual port you wish to tie to this SIP aliases.

Also in the Host Advanced Editor, select the ‘Audio’ tab.

  • Increment the ‘Min IP Phones’ number to match the number of SIP connections required.
  • Change the audio profile to 4 to match the preferred profile 1 set in the Host Advanced Editor > SIP Aliases page.

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How do I install the Mobile App on an Android phone?

Ensure the PC hosting the trilogy Web Server application and the phone are connected and visible to each other over the Wi-Fi. Once this is established the phone should appear as a client in the Web Server > Client page.

Next, download the Mobile App on the phone. Open the phones browser and type the address of the Web Server, for example:

http://192.168.17.50:80/android

This should now download the app to the phone. If not, on your Andriod phone, go to Phone Settings > Security and check the ‘Unknown Sources’ option.

Once the Mobile App is installed, launch the app and go to Settings > Virtual Panel Web Server page and set the IP address and port of the web server.

Choose the Gemini virtual port number you wish to connect the virtual panel to, for example ‘34’.

Once the app is running the TBC, SIP and DBS indicators on the Mobile App are green to indicate the app has connected to the web server and is communicating with the Gemini matrix.

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