General questions

VoipSwitch is a complete software platform that can serve different purposes and be used in various applications, for example:

• by Internet Telephony Service Providers as the base system for offering voip services to end-users,
• by wholesale voip termination brokers as the traffic controller,
• by enterprises that want to connect their remote branches,
• as an IP PBX system.
  Thanks to its simplicity, friendly interface and modularity, which allows to minimize the initial investment, VoipSwitch is also an excellent choice for start-ups in this rapidly growing and challenging Voip world.
  VoipSwitch system consists of the main package and additional modules extending its functionality. The main package includes the SIP/h323 softswitch, built-in billing system based on SQL, web interfaces both for administrator and for endusers, SIP softphone with embedded voip tunnel client and some other useful tools. The additional modules are optional and can be added to the system at any moment without interrupting current activity.
  More about the package and the modules can be found here.

Basically all kind of services that are available in the market. We always observe and follow the current trends in technology trying to react timely to changing customers demand by enhancing our system and adding new features.
The services can be divided on retail and wholesale:
Retail :

• PC to phone and PC to PC - using our own sip based softphone with embedded voip tunnel client which allows for communication (both receiving and making calls) from behind voip blockades/firewalls, for example from UAE or other countries where voip ports are closed;
• device to phone i.e. making and receiving calls from SIP or h323 terminals (bidirectional NAT support), full compatibility with all popular brands of soho gateways and IP phones, support for 5 class services, voicemail, waiting message indicator feature, actual balance information on phone's display and more.
• DIDs/Virtual phone numbers services - assigning phone numbers from various geographical regions to SIP or h323 devices or softphones. Customers can pick up a phone number directly from the web interface with immediate delivery. The system can work with leading DIDs provider thru their API or/and using own DIDs numbers stored in the local database.
• Class 5 services - call waiting, call forwarding, follow me - possibility to set forwarding rules conditioned on various events like busy, no answer, not logged with different forwarding destinations/phone numbers (possibility to set multiple numbers with different priorities), hold, hold with conversation
• Voicemail - checking voice messages by calling predefined voicemail number from softphone or SIP/h323 device or from PSTN thru the IVR system (calling cards) or using the callback service. Detailed information about left messages on the web interface. Possibility to record own welcome prompt. Email notification about new voice messages
• Callback services - triggered by SMS, Web or a missed call. Works with DIDs as an access numbers. Authentication by callerID (ANI) or PIN. Also supported DID callback with unique DID associated with each customer and his/her phone number. Manageable thru the web interface, possibility to add multiple ANIs per user. Multilingual Interactive Voice Response system (IVR). More on the callback and its variants here
• Calling cards - phone to phone services in which a customer dials an access number first, then he or she is asked about PIN (or authenticated by callerID/ANI) and then prompted for a destination phone number. The system comes with own IP based IVR system that supports several languages (possibility to add new), balance announcement, max duration for the call announcement, also it allows to recharge account by PIN and register actual ANI during the call and more. There are several scenarios which can be associated with particular access numbers. Moreover new scenarios can be added or the existing ones be modified by voipswitch's administrator as they are programmed in xml format. More on the IP IVR can be found here
• CallShops - by callshop is understood a place like internet café or a shop where are cabins/booths with IP phones from which customers can make calls and then pay for them at a cash desk. The callshop is a windows based application which shows all the cabins and the calls that are taking place. When a customer has finished, the cashier can see the made calls with their details and costs and can charge the customer. The application also allows to print bills. The advantage of this solution is that the callshop application does not require any special hardware as the calls are not sent thru it, instead they go directly to VoipSwitch (thru the internet connection) and the callshop program exchanges with VoipSwitch only little data needed for billing purposes. Also as the clients can be used softphones installed on PCs or FXS gateways with analogue phones connected, not necessarily IP phones.

Wholesale services:

• Wholesale termination - bulk voice traffic termination, in this scenario voipswitch acts as the traffic controller which on one side collects calls from clients gateways or other equipment, authenticate them, and then depending on the routing tables send the traffic further to termination gateways/gatekeepers. The whole traffic is billed and controlled in real time, the billing supports both prepaid and post-paid accounts types. Also the media (voice) flow can be set per destination and be sent in full proxy mode where all packets are going through the switch or in the modes where only signalization is sent through the system while the media goes directly between endpoints. The system allows to hide the source and the destination what is often important especially if you work as a broker buying and selling traffic from various parties. Additionally the system offers high flexibility in modifying calls setup, called numbers, enables you to manipulate prefixes, caller IDs and all other significant data before sending calls to the destination. There is also a web interface both for clients and terminators with CDRs (which can be exported to file) and other reports. More on the softswitch's functionality here.

VoipSwitch software requires a windows based server. The package does not come with any hardware or windows operating system either. These elements have to be arranged on your side and then our support team install and configure the whole system remotely. Recommended hardware (depending on expected traffic volume) and supported windows versions are described here. Such setup can be located at your place or rented from hosting company what in many cases is a good option given their support and high quality internet connection which is often included in the monthly rate. With regards to the required internet bandwidth it depends on the number of concurrent calls, codecs and used proxy method, more info how to calculate the bandwidth you will find here.
After providing the server we will need only access details and then we take care of the rest. It takes about 1-2 working days to get the whole system installed and configured.

In most cases the answer is no as the whole range of scenarios can be deployed without having own equipment. The calls collected from voip clients can be terminated directly through voip carriers and sent to them over Internet without necessity of being connected to them through physical links. The only exception is if you want to offer termination through local PSTN telecom and the only way to connect is through E1/T1 interface (or sometimes analogue lines or GSM). In such implementations you will need a gateway which will provide required interface, however its role will be limited only to converting PSTN to VOIP and vice versa - the rest will be controlled by VoipSwitch. The softswitch is compatible with all h323 or SIP compliant gateways e.g. Cisco, Quintum and others.
Similarly is in origination services like calling cards. The system can work with DIDs, i.e. virtual phone numbers delivered through internet directly from provider to VoipSwitch. For example if you want to establish an access number in UK you can rent a local phone number from desired geographical location (or national or toll free number) and have required number of channels associated with the number, so when your customers dial the number their calls will be instantaneously forwarded through Internet to your VoipSwitch which in turn will carry out the programmed IVR scenario. This IP based approach in the IVR module allows for easy deployment of multiple access numbers in various geographical locations without high capital expenditures in equipment. If still for some reasons you have to connect to local telecom company in order to setup an access number you will need a gateway with proper physical interface (E1/T1, analogue, GSM..). The gateway does not have to support IVR systems, it has only to be configured in the way that all incoming PSTN calls are converted to voip and then forwarded to VoipSwitch. Then it is VoipSwitch which plays voice prompts, asks for PIN (or authorize by ANI) and carry out the whole procedure. More about the IP IVR system can be found here.

Device to phone services calling from softphone?
Required: Main package
To offer these services the main package is sufficient. It includes the SIP/h323 softswitch, built-in billing, invoicing tools, web interface for administrator, web interface for end-users and sip based softphone with embedded voip tunnel client, voicemail and voice announcement of actual account balance. The softswitch acts as SIP proxy/registrar and h323 switch/gatekeeper as well and additionally provide transparent sip to h323 and h323 to SIP conversion. Rich web interface enables customers to see their made calls history, payments history, actual balance, missed calls, also to manage voicemail and set call forwarding rules. It can be also connected to the OnlineShop module which enables customers to purchase accounts online paying by credit cards, Paypal or moneybookers. Existing users can also add funds online. To have a demo of the website please contact us or visit our demo site www.callto.net where you can create an account and then log in to the user's zone. There is also a link to download our softphone. Please note that both the web design and the softphone are fully customizable and www.callto.net is only one of many templates we offer. Also on customer's request we can create a unique design.

Calling cards

Required: main package, calling cards module

Callback

Required: main package, callback module

There are two approaches, one is to use a DID (virtual phone number) rented from a provider and another is to connect to local telecom company through physical interface. The latter requires a gateway with proper interface (e.g. E1/T1, analogue ports, GSM ).
VoipSwitch and its IVR module can work with DIDs, i.e. virtual phone numbers delivered through internet directly from provider to the switch. For example if you want to establish an access number in UK you can rent a local phone number from desired geographical location (or national or toll free number) and have required number of channels associated with the number, so when your customers dial the number their calls will be instantaneously forwarded through Internet to your VoipSwitch which in turn will carry out the programmed IVR scenario. This IP based approach in the IVR module allows for easy deployment of multiple access numbers in various geographical locations without high capital expenditures in equipment. If still for some reasons you have to connect to local telecom company in order to setup an access number you will need a gateway with proper physical interface (E1/T1, analogue, GSM..). The gateway does not have to support IVR systems, it has only to be configured in the way that all incoming PSTN calls are converted to voip and then forwarded to VoipSwitch. Then it is VoipSwitch which plays voice prompts, asks for PIN (or authorize by ANI) and carry out the whole procedure. More about the IP IVR system can be found here.

Similarly like in the calling cards scenario you can use a DID rented from a dids provider or have an own gateway connected to local telecom company. The difference is that in the callback the calls are used only for triggering the callback and therefore they never get connected, the system rejects them as soon as they come. Thus the number of required channels will be much less that in calling cards scenario when the channel stays busy by the whole duration of the call. In the callback the connection from the system to customer is realized independently, through the termination carrier defined in the dialling plan (not through the link used as an access number).

The SMS callback module provides an interface based on http (protocol used in web) for receiving parameters containing body of SMS and the phone number from which the SMS has been sent. The names of the parameters can be freely modified by the administrator. This method allows to connect any equipment which is capable of forwarding SMS messages over Internet via http, such as GSM gateways. Instead of purchasing a GSM gateway you can also rent an SMS number from a provider and have the incoming messages delivered via Internet to specified URL. Such service is often called "SMS on demand".
Other method is to use a mobile phone (like for example Samsung, nokia ) connected to a PC through USB cable. Our company provides special application called SMSreader which reads incoming messages from the phone and then forwards them to defined URL of voipswitch's server. There is no special hardware requirements for this application, it can be a mere PC with any version of Windows OS. Also the internet link does not have to be high speed as the SMS messages need very little bandwidth. Such setup can be located anywhere in the world what is very useful if you want to provide local GSM numbers for this service in various countries. Many access numbers can be connected to one VoipSwitch server.

Yes, definitely. VoipSwitch supports DIDs in many ways. You can associate phone numbers with end-users so that if one dials a DID it will ring on an SIP phone or our softphone (SIPLink) or will be transferred to voicemail or forwarded to any other phone number. Call forwarding and voicemail settings are configurable through the web interface. User can decide on which events, like busy, no available, offline, the call is either transferred to voicemail or sent further.
Users can also purchase DIDs through the web interface. They can select the country/area and place an order which is realized automatically. The setup fee and/or monthly fee is deducted from the customer's account and immediately thereafter the DID can be used.
The system supports DIDs - phone numbers stored in local database, for example if have own stock and want to sell phone numbers online. There is special tool for importing the phone numbers into the database. This way also allows to connect to any wholesale DID supplier. Another way is to connect to a DIDs supplier through API. Such solutions are offered by voxbone.com and didx.net The main benefit is that you do not have to keep own stock of DIDs from various regions what is costly, instead your system will connect to the provider and place an order automatically with immediate delivery at the exact moment when your customer click order button on your website.
More on DIDs scenarios can be found here.

Yes, responsible for this part is the OnlinShop module. It consists of two parts - the web interface integrated with the VoipSwitch's end-users website and the standalone application installed on the server. The module can work with several credit cards processors and also other electronic payment systems like paypal, moneybookers or others (the full list of currently supported systems ).
Customers can sign up online funding their accounts with chosen amount. In a few seconds after the transaction is approved the system sends an email with the account's settings. From this moment the customer can start using the service. Optionally the system can be configured to enable customers to create free accounts (with 0 balance, for example for test).
The accounts can be funded at any moment from the recharge menu on the website.

Yes, the users can recharge their accounts by entering PIN either on the web or through the IVR (interactive voice response system). The recharge PINs can be either a special PINs that are generated only for this purpose or they can be just normal accounts. In the latter case the new account which funds have been used to recharge the existing account is then emptied and blocked.

Yes, there are two interfaces for managing the system: one is windows based standalone application and the other is web based. Both offer the same functionality. More information about the web version can be found here.

Yes, it is a complete website fully integrated with VoipSwitch. It comes with several ready to use templates which can be easily modified by the administrator without having any special knowledge of web programming or designing. The web interface includes the Admin part with special tools for editing the web pages, menus and other objects.
Under this link you can learn more about the Web and see some exemplary designs.
The users after logging in to the interface can see their actual balance, profile, detailed calls history (with possibility to export to a file), payment history, rates, rates calculator for callback, make a call from the webphone or web callback, add funds through the recharge menu, manage their virtual numbers, check for new messages left in voicemail, see missed calls history with details, set call forwarding rules and more.
To see the demo please visit www.callto.net - sign up for a test account and then log in to browse through the users zone.

When we started working on the software in 2001, the VoipSwitch was initially designed as the system for voip carriers/wholesalers. Then, throughout the years, we have gradually been extending its functionality of some typically retail services but always keeping in mind the wholesale part. As of today, VoipSwitch is advanced softswitch with sophisticated features required in these kind of applications. One of the unique characteristic of VoipSwitch, distinguishing it from other softswitches, is own, built-in billing system. This approach allows for full control of internal billing and authentication processes and significantly increases the robustness of the whole platform.
The system is a traffic controller that enables intelligent interconnectivity among different voip service operators, providing a unified way to authenticate, monitor, control and bill for the voice sessions. You can for example buy termination from one carrier, mark up the rate and then offer to other carriers. Thus, you can act as a broker exchanging traffic and earning on differences in the prices. Low ratio between invested capital and the volume of the traffic the system can handle makes it a perfect solution for this kind of business activity.
VoipSwitch supports several proxy modes which can be selected per each destination/route, provides transparent conversion from sip to h323 and from h323 to sip, enables advanced manipulation of call setups parameters, supports load-sharing, balancing, re-routing and many other, very useful functions.
VoipSwitch can also secure your internal network from unauthorized access from Internet, for example if you have own terminating gateways, instead of exposing them in the open network, you can give to your clients the IP address of VoipSwitch as the central element of your system and let them send the whole traffic through it. Thus, you can control the calls flow and see the actual system status.
More about the softswitch functionality

Yes, it is included in the package. Each user can have a voicemail associated with his/her account. The configuration is through the web interface. User can set the events on which the incoming call will be transferred to voicemail. Also on the web are displayed information about left messages (with details like caller ID, duration, time). To hear the messages user dials voicemail access number (for example 950), it can be called from a sip/h323 device, softphone or even through a PSTN access number or callback (for example when a user has no access to Internet and wish to check his/her voicemail). From the voicemail's IVR level user can navigate using keypad in order to hear new messages, browse through old messages or delete them. Also users can record own welcome message.
Additionally there is email notification about new messages.

Yes, the system supports call forwarding, network based call forwarding (follow me), call waiting, caller ID, voice mail, call hold, hold with conversation, speed dialing and in addition to regulatory requirements, services such as E911.

About 1-2 working days for installation/configuration. In the next step a training is provided, after which you should be ready to go live with the services.

Yes, the softphone (named SIPLink) which is delivered with the main package, is always customized to reflect your VoipSwitch's IP address or domain name, additional links to the web and also the required name. Additionally you can have your logo on the softphone's interface

Yes, however it is extra charged. Some examples you can see in our gallery.

Yes, the switch in connection with the IVR can act as an IP PBX system. It can be configured to answer incoming calls (through DID or a gateway) and then carry out auto-attendant scenario. It is also possible to record own voice prompts. The caller can then dial extensions which are assigned to given users/departments. In addition, for each extension, call forwarding rules can be defined for different events like busy, no answer, offline. Also hunt groups can be established so that if the first phone is unreachable the call will be sent to the next destination and so on. Besides, each extension can have own voice mail with own welcome message.

By callshop is understood a place like internet café or a shop where are cabins/booths with IP phones from which customers can make calls and then pay for them at a cash desk. The callshop is a windows based application which shows all the cabins and the calls that are taking place. When a customer has finished, the cashier can see the made calls with their details and costs and can charge the customer. The application also allows to print bills. The advantage of this solution is that the callshop application does not require any special hardware as the calls are not sent thru it, instead they go directly to VoipSwitch (thru the internet connection) and the callshop program exchanges with VoipSwitch only little data needed for billing purposes. Also as the clients can be used softphones installed on PCs or FXS gateways with analogue phones connected, not necessarily IP phones.

Yes, users within voipswitch can call to each other using a phone numbers associated with their accounts or by logins (usernames). In the latter approach however, if you allow non-numeric characters in logins then they can have problems with calling such a destination from a numeric phone's keypad. The other way is to let users select the numbers from the web interface the same way as they can choose DIDs (PSTN numbers delivered through internet). You can have own numbering plan or use internet phone numbers pool (more on e164.org ) and allow users to select a number themselves after they sign up (or you can do that for them). Then they can call these numbers the same way as they do when calling outside the network. On voipswitch's side you can set for example that internal calls are free or at special rate, it is up to you. Moreover, the users can be called from other networks if there is interconnection with voipswitch, for example from PSTN through an access number (DID) hooked to the switch. In this case you can be away from Internet and make a call from a regular phone to the access number and then through the IVR dial the internal number and get connected to the user.

Yes, thanks to our voip tunnel technology that enables making and receiving calls from behind any blockade. The client-server technology encapsulates voip packets and use only one UDP port for communication. It can be any port - what is also configurable in the software, for example DNS port or any other that usually is allowed.
In the SIPLink the voip tunnel client is already built-in and does not require any special action from the user. The SIPLink is widely used in countries like for example UAE. Apart from making calls, users can also receive incoming connections, for example through DID numbers associated with the accounts.

Yes, it is possible.