To configure VOP with Asterisk follow the tutorial here: Asterisk / Tutorial

To configure VOP with sipXecs follow the tutorial here: sipXecs / Tutorial

To configure VOP with PBXnSIP follow the tutorial here: PBXnSIP / Tutorial

To configure VOP with 3CX follow the tutorial here: 3CX / Tutorial

VOP is available in four editions:

VOP Standard, for medium and large company receptionists.

VOP Lite, for small company receptionists and call center agents.

VOP Micro, for desktop computers.

VOP Nano, for personal computers (free non commercial use) and for ITSP (one time license) who wants to offer a brandable softphone to all their customers.

You can review all the editions prices, features and limitations by clicking here.

VOP can be evaluated for 30 days without any limitation.

During your evaluation you can switch between versions (Standard - Lite - Micro) in the Settings window: Settings > License > Version.

To receive a fully functional 30-day trial version: click here.

You will also receive a link to our purchase page with standard quantities and licensing.

For higher quantities or custom licensing we will send you a custom invoice.

VOP can be used as a standalone softphone or tethered to a hard phone.

When used as a standalone softphone, the audio is going through the software and your computer sound card (or headset).

When tethered to a hard phone, the calls are still managed by the application but they are connected with the hard phone so the audio is not going anymore through the software but directly from/to the hard phone.

The hard phone can be any phone, analog or VoIP or even a cell phone, as long as it has its own phone number (different from the VOP one) and can be called by the software.

Switching from standalone softphone mode (default mode) to hard phone mode is as easy as entering the hard phone number in: Settings > Operator > Operator phone number.

To switch back to standalone softphone mode, just remove the hard phone number.

There are several advantages of using a hard phone.

Since no audio is going through the application, even if the computer has a slow CPU or is overloaded the audio quality is still fine.

Many receptionists/operators are used to handle calls using a regular phone and it helps them to move smoothly from analog PBX to IP PBX or IP Centrex.

The operator can still handle incoming calls when his computer has a failure by configuring the VOP SIP account to redirect incoming calls to the hard phone when not registered.

In case of separated data/voice network, the application can control calls through the data network (computer network) while the audio is going through the voice network (hard phone network). Even if the data network is overloaded the audio quality is still fine.

VOP provides two types of directories, local and external directories.

When a directory is local it means that all the users in that directory are local to your company (company staff).

Some directories are created manually or imported from vCard files, some are loaded from a LDAP server, some from Outlook contacts, some through the internet.

When you create a directory you decide if it is a local or external directory.

Some directories are automatically set to local or external depending on their source.

Outlook contacts directory is set to external because it contains external users. We also provide a parameter that lets you specify your company name so outlook contacts from that company are placed in a local directory while others are placed in an external directory.

LDAP directories contain both users and contacts so when a LDAP directory is loaded, users are loaded in a local directory and contacts are loaded in an external directory. You can tweak the default LDAP users/contacts filters to better differentiate the records.

When you receive an incoming call, the caller phone number is searched in every directory.

If the caller is identified from a local type directory, the call will be shown in bold (in the queue and in the logs). This helps the operator to quickly identify if the caller is from the company.

The operator can also decide to automatically reject/forward anonymous or external calls.

He can also decide to automatically call anonymously external users.

Phone presence (extension monitoring) is also restricted to local directories.

Monitoring extensions is what we call phone presence.

To enable phone presence for the phone contact of a user in a local directory, you need to right-click on the phone contact and select: Contact > Presence > Enable.

If the SIP proxy is supporting presence the icon of the phone will become green.

If your SIP proxy supports presence resource list, you can instead select the phones you want to monitor at the SIP proxy and enter the resource list name in: Settings > Account > Presence resource list.

In this case you can also ask the application to automatically populate its local directory from the presence resource list content by checking: Settings > Account > Populate local directory from presence resource list.

If your SIP proxy supports basic phone presence you will only be able to see the phone status. (ringing, busy, available)

If your SIP proxy supports full phone presence (RFC4235) you will also be able to see every active call on the phone. (direction, state, caller/ee phone number and name)

If your SIP proxy supports it you can even enable phone presence for conference room extensions to see every participant, for parking room extensions to see who is waiting and also for agent group extensions to see every call waiting in the queue.

If your SIP proxy supports full phone presence (RFC4235) our application will offer you to intercept a call from a phone by right-clicking on a phone (or on a call on a phone) and selecting: Intercept

Otherwise you need to enter the call intercept code to dial in: Settings > Account > Call intercept code. ('$' will be replaced by the phone number)

For example if to intercept a call on the phone number 900 you need to dial *14900, the intercept code should be "*14$".

User presence is available using XMPP or/and MSN.

To use XMPP you need to enter your XMPP server and account information in: Settings > XMPP

To use MSN you need to have a Windows/MSN/Live Messenger installed.

To enable user presence for a user, you need to add a XMMP or MSN contact to this user.

When this user will be online his name will turn to bold in the directory.

You will also be able to see his full status next to his XMPP/MSN contact address.

You will be able to send an instant message to the user by right-clicking on him and selecting "Send Message".

If you are using XMPP you will also be able to transfer files to/from the user by clicking on the up arrow in a message window.

You can also display both icons (phone presence and user presence) next to each user name by checking: Settings > Account > Show user presence icon.

The CRM is integrated in VOP through a dedicated web browser window (Internet Explorer).

Most CRM offer a web interface, you just have to enter its URL in: Settings > CRM > CRM URL.

If your CRM provides an URL with parameters to display a customer record based on its phone number, you should to enter this URL instead. (in the URL, $u will be replaced by the caller/ee phone number)

Once done you will be able to right click on a call and select CRM to display the caller/ee record in the CRM window.

Using the CRM settings, you can also decide to display the CRM on a call double click or automatically when there is a new call.

Configuring this will let you quickly handle customers calls by interacting with their CRM records.

Let's say for the example that Google is your CRM web interface.

To display a search result for a 'word' the URL used by Google is: http://www.google.com/search?q=word

So to configure Google as your CRM tool you need to enter the URL "http://www.google.com/search?q=$u" in: Settings > CRM > CRM URL.

As you can see the 'word' has been replaced by '$u' which will be later replaced by the caller/callee phone number before loading/displaying the URL.

Now for each call a new right-click menu item "CRM" will be available, which once selected will display the Google search result of the caller/ee phone number in the CRM window.

For a real CRM example, you can create a trial account at Salesforce and use this CRM URL:

https://emea.salesforce.com/search/SearchResults?searchType=2&str=$u&search=Search

For the CRM URL and many other places where a URL is used, we support these variables:

$l (login) = proxy user name.
$m (me) = proxy user name and host. (user@host)
$p (password) = proxy password.
$P (password) = proxy password in MD5 format.
$o (operator) = operator phone number.

For an incoming call:

$u (user) = caller phone number.
$d (display) = caller name.
$a (address) = caller address. (user@host)
$U (user) = callee phone number.
$D (display) = callee name.
$A (address) = callee address. (user@host)

For an outgoing call:

$u (user) = callee phone number.
$d (display) = callee name.
$a (address) = callee address. (user@host)
$U (user) = caller phone number.
$D (display) = caller name.
$A (address) = caller address. (user@host)

For the phone contact of a user:

$u (user) = phone number.
$d (display) = phone user name.
$a (address) = phone address. (user@host)

The users calendars can be viewed in the built-in web browser window (Internet Explorer).

Most calendar softwares offer a web interface. For example Outlook offers the (O)utlook (W)eb (A)ccess.

For each user, you just have to enter his personal calendar URL as his web contact.

Once done you will be able to double click on a user web contact to view his calendar in the web browser window.

VOP offers configurable keyboard shortcuts to quickly manage calls.

By default the shortcuts are configured to be used with the numeric keypad of the keyboard (with Num Lock set to OFF).

Up/Down keys will let you move within calls or directory users. Left/Right keys will let you move from/to the selected user contacts.

Clear key (the 5 key) will let you switch between incoming calls and directory.

Insert key will let you activate a call (pick it up or put it on hold) or activate a user contact (call, e-mail or browse it).

Delete key will let you hang up a call.

Add key (the plus key) will offer you the most logical transfer, through a menu, depending on where you pressed it and how many calls are active. You will be able to validate the offered transfer by pressing Enter or change it using Up/Down keys or Cancel it using the Escape key.

You can also press F1/F2 on an incoming call to tag it by adding additional From/To text.

Finally you can press F3 to quickly move to the directory search field of F4 to move the outgoing call dial field.

With a little practice you will be able to quickly manage calls without using the mouse.

We also added non-configurable key shortcuts to quickly control the current call:

To Answer/Pick-up the current call press Alt-PageUp.
To Reject/Hang-up the current call press Alt-PageDown.
To Put On/Off Hold the current call press Alt-Home.

You can dial any phone number on your screen from any application by selecting it, like you would do to copy/paste the phone number, but instead of pressing Ctrl-C to copy it, you press Windows-C to call it. (the Windows or Start key on your keyboard)

This will bring VOP to front and call the selected phone number.

VOP also supports callto URL (to initiate a call) and xferto URL (to initiate a transfer) embedded into web pages.

You can also make VOP call a phone number using the utility "VOPCtrl.exe", located in the application folder, using the syntax: "VOPCtrl /call phone-number"

VOPCtrl offers additional commands that can be used to control the current call:

"VOPCtrl /transfer phone-number" will transfer the current call to the phone number.
"VOPCtrl /pickup" will pickup or answer the current call.
"VOPCtrl /hangup" will hang up or reject the current call.
"VOPCtrl /hold" will put the current call on/off hold.

Our application only registers to one SIP account.

You can associate/redirect multiple phone numbers to the SIP account.

When there is an incoming call the application shows the call destination (dialed phone number) in the To field of the incoming call list.

The operator knows before answering the call which company or service is called.

You can link a directory to a phone number by right-clicking in it and selecting: Directory > Phone.

Once a directory is linked to a phone number, when the operator will pick up a call destined to this phone number that directory will become the top/front directory on his screen.

Our application relies on the IP PBX to put the called phone number in the SIP "To" or "P-Called-Party-ID" fields of the INVITE.

For example, using Asterisk, you need to add the "P-Called-Party-ID" field using SIPAddHeader() before issuing the Dial() to the VOP extension.

VOP offers several ways to handle situation where there are too many incoming calls.

You can limit the number of concurrent incoming calls. Extra calls will be rejected to voicemail or forwarded to another phone number.

You can also set a delay after unanswered incoming ringing calls will be rejected to voicemail or forwarded to another phone number.

You can also set a delay after unanswered incoming ringing calls will be automatically answered and put on hold.

These parameters are available in: Settings > Incoming calls.

Voice effects let the user modify the pitch of his voice, add reverb or add a background sound to his conversation.

Voice modification can be used to call someone without being recognized (for example to evaluate an employee or to call a competitor).

Background sounds can be used to handle special events (for example using a christmas music) or to remind the company theme (for example beach/sea sounds for a travel agency)

To enable voice effects you need to click on the microphone icon in the status bar and select which effects you want to enable.

You can use any USB headset/handset with our application, but if you want to have the keys/buttons/hook natively supported you have to select one from our validated list.

We select USB phones based on their audio quality and ease of use.

We currently support these models:

Jabra GN9330e OC (Part# 9327-509-505)
Plantronics Savi Go OC (Part# WG101/B)
Plantronics C220-M (Part# 80299-02)
Plantronics C435-M (Part# 85801-05)
Plantronics C620-M (Part# 79930-42)
HP Internet Handset
VoIPvoice Cyberphone W
Polycom CX200
IPEVO Free.1

When we validate a USB phone, we natively support every key/button with on/off hook detection.

It means that you don't have to install any extra driver or software.

You just have to plug the USB phone and select it in: Settings > Softphone > USB phone.

When you have a VOP Standard at your reception and several VOP Lite/Micro at the other computers you can interconnect them for additional features.

VOP Standard can act as a server (VOP Server) receiving connections from others VOP Lite/Micro acting as clients (VOP Clients).

The interconnection brings enhanced features:

VOP Server becomes a presence server for all the other VOP Clients. VOP Server proxies the phone presence between VOP Clients. This enable users to have phone presence even if their SIP proxy doesn't support it.

VOP Server becomes a messaging server for all the other VOP Clients. VOP Server proxies all the messages between VOP Clients. This enable users to message each others even if their SIP proxy doesn't support it.

VOP Server becomes a note server for all the other VOP Clients. VOP Server proxies all personal notes between VOP Clients. This enable users to add/modify a personal note to their record in real time. (for example to indicate transfer restrictions, availability periods, location change, backup contacts ...)

To enable this interconnection you need to specify the VOP Server host or ip address to all the VOP Clients in: Settings > Interconnection > VOP server.

In order to work the VOP Server and VOP Clients local directory must at least contain each others and at best be common.

We provide two types of license:

The first type of license is a serial number generated from the address of your network card.

The license is bound to your computer. When you change your computer we generate a new serial and we black-list the old one.

For multiple licenses we create you an account at our license manager where you can generate your licenses yourself.

The second type of license is a USB dongle.

The license can be moved from one computer (or even customer) to another.

The USB dongle and its shipping require an additional fee.

VOP provides a multi-level and multi-layer debug window.

You need to press Alt-D to see the debug window.

Using the toolbar you will be able to raise the debug level from 0 to 5. (level 0 or 1 is usually enough to debug most common problems, where level 5 will display every SIP packet)

Using the toolbar you can also write debug information to file or copy the whole window content to a notepad to e-mail it to us so we can investigate your problem.

VOP is compound of:

VOP.exe - It's a small bootstrap that loads the firmware and the configuration file then starts.

VOP.bin - It's the firmware who contains the application binaries and data which will be loaded or extracted when needed.

VOP.xml - It's the XML configuration file with all the parameters.

These are the 3 files required to run VOP.

You also have other files created later by VOP and loaded at start-up if they are present:

settings.xml - It's the XML file containing the user settings.

state.xml or state-compact.xml - It's the XML file containing the application state when it last exited.

When VOP starts it loads the configuration file named VOP.xml located in the application folder.

This configuration file offers more than 200 XML parameters to control the application.

All the XML parameters are described at the beginning of the file in a comment section.

If you want to pre-configure the application you need to edit this file to enter your desired values.

Our application can load customers directories from your LDAP server if you have one.

Our application can also download directories through HTTP or HTTPS in our XML format.

All our directories uses the same XML format. The format is described in the configuration file (VOP.xml) at the end of the comment section.

If you need examples you can look at the directories that our application uses.

They are stored in the application folder "directories" for user managed directories and "cache" for imported/downloaded directories.

To make our application download a directory through HTTP/HTTPS at start up you need to enter its URL in the configuration file (VOP.xml) using the parameter <directoryurl>.

You can add as many downloadable directories as you need.

Using this parameter, you can for example enter the URL of a PHP/ASP script hosted at your web server that dynamically generates the customer XML directory from your database.

The configuration file (VOP.xml) contains a section named <settings>.

The settings window of the application is dynamically built from this <settings> section.

By adding/removing XML parameters to this section you can decide which settings the user can see/modify within the application.

You can change the order of the XML parameters and create/rename categories (<header>) to fully design the settings window.

You can also set a password to a category using the password parameter, like this: <header password="secret">.

When you exit the application the windows layout is stored in a file named sate.xml in the application folder.

If you want to create a default layout for your customers you need to first design your layout within the application then exit it.

Next, you need to edit the file state.xml and copy the <layout> section to the configuration file VOP.xml.

This layout will become the default layout when the user starts the application for the first time or when he resets the layout.

Adding or modifying a translation is a very easy process.

You need to press Alt-T in our software, this will create a XML language file named "language.xml" in the application folder.

This language file will contain every word/sentence of our application.

This file will be loaded when the application starts and replace the current translation.

You can translate/modify this file from time to time, partially or fully.

Once finished, you can either redistribute this language file (language.xml) with the application, or include the <language> section in the configuration file (VOP.xml).

If this is a new translation you can even send us your translation file so we add it to our next release.

You create your configuration file (VOP.xml) with your desired default values and the settings section of the application.

You create a setup containing the three files VOP.exe, VOP.bin, VOP.xml.

And you are ready to deploy our application to your customer.

We also offer extra services when needed: installer creation, logo addition, (A)utomated (P)rovisioning (S)ystem setup.

We offer an (A)utomated (P)rovisioning (S)ystem.

We create a custom bootstrap (VOP.exe) which points to the URL (HTTP or HTTPS) of your APS server.

We also create an installer containing only this bootstrap for deployment.

When the bootstrap starts it asks for a username/password (different from the SIP account) managed at the APS server.

If the username/password is valid the bootstrap loads the firmware and the configuration file through the network.

The username/password and configuration file are encrypted and stored on the customer computer. (the username/password can not be stolen and the configuration file content can not be seen)

At each new start-up the username/password is checked, firmware/configuration updates are checked and downloaded when needed.

When running, the application also checks regularly for new firmware/configuration updates.

The ITSP can manage all his customers binaries/configuration updates remotely at his APS server.

We also provide an APS server script based on PHP/MYSQL.

The application provides a multi-level and multi-layer debug system.

The debug information can be sent to a remote syslog server at the ITSP.

The debug can be enabled and the level can be adjusted through the configuration file.

When our application starts it builds the list of every IP associated with the SIP Proxy.

You can specify a comma separated list of IPs for the SIP proxy manually in: Settings > Account > Proxy address.

In that case the application will use that list of IPs.

Otherwise, the application will get every IPs associated with the SIP proxy host.

You can also enable the DNS SRV resolution of the SIP proxy host/domain in: Settings > Account > Use DNS SRV to get proxy address/port.

In that case the application will get the list of every host/port associated with the SIP proxy host/domain respecting the priority.

For each host, the application will get the list of every IPs associated with that host.

When the list of all the IPs will be constructed, the application will start by registering to the first IP and if it fails move to the next one.

At each re-registration, if the registration fails it will move to the next IP.

When all the available IPs will fail, the application will notify the user and the application will start over at the beginning of the list.

Failing means having no reply at all from the registration packet (if the registration is rejected with an error, like wrong authentication or else, the application will not move to the next IP but retry the same one later)

Our application can be configured to use any network interface available on the computer.

You can either create a virtual VLAN interface or add a dedicated network card to the computer.

You can also configure interfaces and manage routes directly/remotely from the configuration file.

You can also set the SIP/RTP TOS (Type Of Service: DSCP + ECN) to prioritize SIP/RTP packets on the network.

We offer logo addition in the application and we create a custom installer with pre-configured settings or an APS bootstrap.

When the logo is added to our software it is available in every edition/version.

Depending on your order quantity this is either offered or payable through a setup fee.

We also offer ITSP the ability to give to all their customers our VOP Nano edition for a one time license fee.

The user selfcare is usually a web page that the user can browse to modify its general (company wide) or personal telephony settings at the IP PBX or Softswitch.

Our application offers a dedicated web browser window named "Online" which can display this web page.

To display this online web settings page you need to enter its URL in: Settings > Web > Online web settings URL.

The ITSP can build/manage a web page with additional features (FAX, SMS for example).

These features will then be integrated to the application in its own dedicated "Online" window.

To display this online web page you need to enter its URL in: Settings > Web > Online web settings URL.

We also offer a method to add new menu items to our existing right-click menus.

When the user select a new menu item, our application issue a web request to an URL with parameters containing the context (selected call, user, contact).

The web request can be silent or the response can be displayed in the "Online" window.

You can embed our application in any hardware PC with a touch screen interface.

Through the configuration file (VOP.xml) the application can be automatically launched at windows startup, the window location and size can be set up, the window features can be selected/limited (move, resize, minimize, close), the window can also always stay on top.

For touch screen interface, the application font type and size can be adjusted, the context menus can be switched to show directly on item selection.

You can watch a video from one of our integrators here: VOP Touch Screen integrated by Teo