Installing

Table of Contents [Hide/Show]

Downloading the software
Installing the software
Installing the software as a Buddy
Starting Solo for the first time
Audio Settings
   ASIO
   Configure inputs/outputs
SIP Providers
   SIP settings
      Display
      SIP Server
      NAT/Firewall
      Audio
      Number rules
Using a handset
   Configuring Solo for use with Handsets
   Configuring the handset (Linksys SPA9xx example using web based configuration tool)
Using GPIO cards
      Advantech
      Axia Element

Downloading the software¶

Once you have your hardware in place and setup correctly you are ready to download your Solo software. Once Broadcast Bionics have received your order, you will be emailed a login for our download site. Simply go to http://download.bionics.co.uk which will take you to the screen shown below. Log on using the details provided and you will be able to download your PhoneBOX Solo software and accompanying manual.

Installing the software¶

The software will be downloaded from the bionics website as an .exe file. You can run this straight from the browser, though we would recommend saving the file to a local location first. Once the .exe file is downloaded, double click it (accepting any dialogue boxes that appear) and you will be presented with the following screen:

Once you have agreed to the licensing terms of PhoneBOX Solo, you will then be presented with the default install path for Solo. You can now click the Browse button to choose a different path, or click Next to proceed with the installation.

In the next install, screen, select "Master" (Buddy installs will be covered later).

Finally, click Install on the following screen to start your PhoneBOX Solo installation.

Installing the software as a Buddy¶

The process for installing Solo as a Buddy is similar to the previous process that was used to isntall the Master. However, when asked in the isntaller, please select Buddy, as shown here:

You will then be prompted to enter the machine name or IP of the Solo Master machine. Once this is entered, click Next.

Once the install is complete, please click Finish. Now when you start up the Buddy, it should try and connect to the Master installation and display Lines, Devices, etc. as dictated by the Master.

Starting Solo for the first time¶

Now you have downloaded your software, to start Solo, simply click on the start menu icon as you would to start any other application.

The Solo software will load and, if Solo has not already been licensed, will prompt you to request a license. Click the 'Request/Download' button, enter your details to submit a license request. Following this Solo will close. Until a license has been issue it is not possible to start Solo.

Once a license has been issued, you will receive an email from support@bionics.co.uk informing you that your licence request has been approved. Upon receipt of this email you will now be able to Solo as a licensed user. Simply open Solo again and the license will be downloaded automatically.

Solo will then present you with some initial choices that will allow you to configure your new installation.

The first choice is the network adapter you will be using on this machine to communicate with the SIP gateway/provider:

You can also tick the box in the bottom left of this dialog to tell Solo to remember this choice of network adapter on future occasions.

You will then be presented with the Location settings. Once these are completed, click "OK".

Audio Settings¶

You will be prompted to set up your audio settings using the following dialogue box.

By clicking the Configure button next to each option, you will be able to set applicable devices, and then click OK to apply the changes, or cancel to abandon the changes. There is also a checkbox that will allow you to specify whether each device is shown on Buddy machines or not.

ASIO¶

If you wish to use ASIO devices, select an ASIO driver from the drop down list and select 'change'. To configure a selected ASIO driver, select 'ASIO settings'. This will show the selected ASIO drivers own configuration dialog. Any changes made here may require Solo to be restarted before they take effect.

Pleae note that before you can select an ASIO driver, one will have to be installed. We recommend that you use the ASIO driver provided with your souncard, but if required, ASIO4ALL can be used with a wide range of sound cards. This can be found here.

Click 'Configure' for each device and you will be presented with the following to make you selections.

Configure inputs/outputs¶

Select your input and output channel devices from the drop down boxes and click OK. Note that if using an ASIO driver (which takes exclusive use of the selected soundcard), it is likely the inputs and outputs of the soundcard will appear in the drop down list as both as WAVE devices and ASIO devices (as shown in the image above) - make sure you select the 'ASIO:xxx' device at the bottom of the list and not its equivalent WAVE device or Solo will be unable to open the input or output.

Simply repeat this process for each device and for your Music on Hold (MOH). Your MOH will only require an input channel. Should you need to edit these at a later date you can find the 'Audio Settings' dialogue box in the Tools > Admin menu Once completed click OK and you will be prompted to set up your SIP settings.

If the device should be controllable from a buddy, select 'Show device on buddies'.

SIP Providers¶

Each 'provider' represents a set of lines either on a SIP gateway or SIP provider. You can add as many 'providers' as required but are limited to a maximum total number of lines by your license.

If no providers are configured you will be prompted to set up at least one before using Solo. To make changes later go to Tools->Admin->SIP Providers

The SIP Providers dialog allows you to add and remove providers. To add a provider click the Add Provider button and follow the prompts on screen. Use the up and down buttons to change the order providers are displayed on the lines.

It is also possible to activate 'Demo' mode, which simulates a SIP provider for testing and demonstration purposes. Please ensure demo mode is not selected when you are ready to use Solo with real providers!

SIP settings¶

Display¶

Lines - The number of lines available on the gateway/with the provider. This will directly control how many calls can be made and received in Solo from the provider.
Dial priority - When clicking the dial button in Solo a provider is chosen to dial on according to the dial priority. Providers with a lower number will be tried in turn until a provider is found with spare lines.
Line label - The line label will be shown on each line for the provider. It is a good place to enter a description (e.g. XD) or the actual phone number of the line.

SIP Server¶

Most of the settings in this section will be specific to the SIP gateway/provider.

Registrar/Proxy - The IP address and port of the gateway/provider.
User @ domain - The user name and domain name required for authentication to connect to the gateway/provider.
Auth id - (Optional) If the auth id is different to the user name for the provider/gateway specify it here.
Password - If a password is required by the gateway/provider enter it here.
Registration - tick register if the gateway/provider supports registration, and specify an interval that registration should occur.
Allow direct calls - if ticked, calls to alternate SIP domains will be made directly from Solo, not via the gateway.
Local IP address - leave set to '(auto)' except for advanced IP configurations involving firewalls and port forwarding.

NAT/Firewall¶

STUN Server - Set the address of a STUN server, if known. STUN allows discovery of paths through NAT firewalls. Not usually required - set this value only if inbound calls or their audio is not received when using a SIP provider on the Internet. STUN should only be used a a last resort method of negotiating NAT and in most cases is not required.
'Keep alive' interval - To keep NAT translation active a keep alive can be sent on a regular basis.

Audio¶

Max jitter (ms) - The maximum jitter controls the variation in interval of audio packets received over IP. If the jitter is too small glitches in audio may be heard. If the jitter is set too high, the latency of the call will be excessive.

Number rules¶

Dial prefix - a number to insert in front of any dialled number, e.g. 9 for an outside line. This would then not need to be included in any dialled number within Solo. The minimum digits controls when the prefix will be applied. E.g. numbers with less than 5 digits are internal calls - don't apply the 9.
Strip prefix - a number to strip from the front of received calls. If a gateway/provider is prefixing calls with an unrequired digit specify it here to remove it. The minimum digits controls when the prefix will be applied.
Strip characters - any characters specified here will be removed from any number (dialled or received). E.g. enter ',-' to have commas and dashes removed from numbers if caller id is presenting numbers which contain these characters.
Strip local area code when dialling - if the provider cannot dial local numbers when the local area code is included in the dialled number, select this option to remove the area code before passing it to the provider.

Using a handset¶

Only Cisco SPA502 and SPA504 handsets are currently supported by this feature. PLease contact us if you require these handsets and we will be happy to help.

Linksys SPA921 and SPA922 legacy handsets can also be used with this feature though have now been dicontinued.

Configuring Solo for use with Handsets¶

Navigate to Tools - Admin - General Settings - Handset.

Tick 'handset enabled'.

The 'auto answer' option will cause the handset to go directly to speaker phone/handset as soon as a call is answered/dialled/unheld in Solo. This is the recommended way of using a handset in Solo as otherwise the user will be required to click, for example, answer in Solo, and then the handset will ring until the user lifts it.

Handset user - The user name configured on the handset (the default is 'handset').

Handset domain - The domain the handset is configured to register with (default is the name of the Solo PC). It may be necessary to use the IP address here if the handset is unable to resolve the name of the Solo PC.

Local port - The port on the Solo PC which will listen for the handset connection. Default is 5060 but if using 5020 in the proxy field then set it to 5020.

Configuring the handset (Linksys SPA9xx example using web based configuration tool)¶

These handsets can be configured via a web interface that runs over the network from the handset itself.

To find the IP Address of the phone, press the Setup button, and then press 9. A set of numbers should be shown under the "Current IP" heading.

By entering this IP address in a browser on the same network, you will be able to configure tha handset easily via your computer.

Once you have loaded the web interface, you will be presented with a set of configuration tabs. To configure the handset appropriately you will require access to the "Ext 1" tab. If you cannot already see this, then please click "Admin Login", located in the top right hand corner of the page, as shown in the screenshot at the end of this section.

The following settings are the most important ones that will need to be configured for proper use of the handset with Solo:

Proxy and Registration > Proxy

In this case, "proxy" means the PC that is running Solo. The handset needs to connect to Solo to be used with it, and this field will tell the handset where to find the Solo installation.

In this field you should first enter the name or IP address of the PC running solo (the name of the PC may not work in this field on some network configurations in which case, the IP address should be used).

The IP address should then be followed by a colon, and then a port number. Broadcast Bionics recommends the use of port 5020. However, if the 'local port' setting has been changed in the handset configuration of Solo, this value should be entered after the coloon instead.

Subscriber Information > User ID

Enter the username configured in Solo's handset settings (the default is 'handset') into this field.

The following settings can only be found when using Advanced mode.

Advanced mode can be accessed by clicking "Advanced" in the bottom left of the web interface. This is also shown in the screenshot at the end of this section.

User tab

Please set "Auto Answer Page" to 'yes', and then set the Preferred Audio Device 'speaker' or 'headset', depending on preference.

Regional tab

To remove the triple beep which you hear when answering/unparking a call to the handset, change the 'Page Tone' to '600@-16;.3(0.0)'

SIP tab

To make the handset attempt registration more frequently, set the 'Reg Retry Intvl' and 'Reg Retry Long Intvl' (both found under 'SIP Timer Values') to 30 seconds.

Using GPIO cards¶

Advantech¶

Solo supports the Advantech PCI-1750 card which has 16 isolated DI and 16 isolated DO channels.

The GPI/O's are not configurable in Solo - they will work automatically if an Advantech card is detected.

The inputs and outputs and are defined as follows (and are steady not pulsing);

Outputs 1-4 – SIP Provider 1-4 ringing (note - provider not line number)
Outputs 5-8 - Device 1-4 occupied (no handset)
Outputs 9-12 - Fader up/down notification (call on device turns red when fader is up).

Inputs 1-4 - Answer next to Device 1-4
Inputs 5-8 - Drop on device 1-4

Please note that an Advantech driver is required for your card (including Advantech device manager). Once installed, please open Advantech device manager and confirm card is installed and working using the test button. ActiveDAQ pro is not required.

Axia Element¶

Also supported is the Axia Element console. The IP address of the console, and 2 livewire stream numbers can be configured in settings.

The channel on button for the configured stream will answer next to the device.
The channel off button for the configured stream will drop a call on the device.
Fader up/down notification will be shown on the device - the device turns red when the fader is up and returns to its normal colour when the fader is down.