Callshop is a simple and comprehensive solution for facilities allowing customers to make low cost international calls. It is fully manageable through a comprehensive web interface. The callshop administrator (owner) creates so called cabins which are in fact VoIP accounts associated under the umbrella of the main callshop account.
A customer can make calls using one of the existing accounts (usually they are preconfigured on a PC softphone or on hardware iphones) and when finished he pays at a cash desk where a clerk checks the total sum on the web and bills the customer.
The main callshop account is of a prepaid type. The callshop owner has to make some deposit in advance to their provider or the provider’s reseller. The prepaid system works in real time guaranteeing that callshop accounts will not go negative.
- Modern looking and intuitive web interface in html 5 (support for smartphones and tablets)
- Operator accounts with limited privileges
- Admin accounts with full rights
- Booths/Cabins creation
- Rate plans management
- Profit & Loss reports
- Real-time active calls visual representation with associated cost and duration
- Detailed calls history records
- Auto invoicing
- Multi-language support
- Works with any SIP device or softphone
The Callback module enables various types of callback based services. What is common for all types is that it is the provider’s system that initiates connections to two end points on a user request. The types of callback differ in the way how a user can trigger the service.
DID or missed callback
Initiated by making a phone call to a provider’s service number. The service number can be implemented by a DID from a DID provider, PSTN or GSM gateway.
- By PIN the system calls back to the caller ID that has come with the call; after the call is connected the IVR (Interactive Voice Response system) prompts for a PIN; after successful authorization the system calls the destination number.
- By Caller ID (ANI) the system tries to authorize the Caller ID received with the triggering call; if authorization is successful, the system makes a call back to the caller ID and asks for a destination number.
- By DID in this method each user has a unique DID (virtual phone number) associated with his/her account. After dialing this DID the system initiates a call to a predefined phone number associated with the account (not the caller ID) and then asks for a destination number. This service is mainly for users that request callback using phone services with blocked caller ID.
In this method users trigger the callback by sending an SMS to a provider’s service number.
The system works with SMS numbers that are capable of delivering messages over SIP or HTTP protocols (e.g. Voxbone, Nexmo)
Authorization in this method can be by Caller ID of the phone from which the SMS has been sent or by user/password (or PIN) sent in the SMS content.
Also called Connect Two is a conference between two parties initiated through the User Portal.
This type of callback can also be implemented in softphones along with VoIP and/or Call Through dialing.
- Custom IVR scenarios with additional options like balance, rate, max call duration announcements
- Multi-language support in IVR prompts
- Ready to use sets of prompts in more than 30 languages
- Option to wait until destination leg is connected before calling the initiating party
- Authentication by PIN or Caller ID (PINless)
- Shared accounts among other services like calling cards, mobile VoIP and others
Fax server is an extension enabling faxing over IP services. It is implemented as an application server controlled by the main VoipSwitch system. The fax server role is to process incoming faxes and then deliver them to recipients. Also, in the other direction it converts various file formats to fax over IP formats and transmits to destinations.
The module is equipped with tools for automated conversion from and to different formats.
Customers manage their fax box configuration from the VoipSwitch User Portal or from the VoipSwitch Unified Communication portal (business customers).
- Full t38 protocol support
- Managed through web interface (cover page, banners, retry intervals and times)
- Full set of Web APIs for sending and managing the service
- Separate fax boxes for each user
- Shared fax box for enterprise account
- Download faxes from the web
- Support for doc, pdf, bmp, jpeg and other image formats for sending fax
- File upload and fax send directly through web
- Fax to email delivery
- Email to fax service
Implemented as an application server controlled by the VoipSwitch main platform. It enables VoipSwitch to route and connect calls when the call origination and termination have no common audio codec. The transcoding process is fully transparent for end users.
The server can run as a standalone application or in a cluster configuration with VoipSwitch distributing transcoding tasks among multiple instances.
The transcoding server can decode and then encode any of the codecs supported by VoipSwitch.
The most widely used codecs include:
G729, g711u and a, AMR NB, AMR WB, g722, Speex, Opus, Silk, GSM
The Tunnel technology has been developed by VoipSwitch in order to enable making and receiving VoIP calls for users who are behind firewalls that block VoIP traffic, especially in countries like UAE, Oman and some other which have recently decided to delegalize internet telephony.
The Tunnel is part of the main package thus it’s granted with the purchase of VoipSwitch main platform.
The Tunnel reduces the number of ports needed for VoIP communication to only one. The VoIP Tunnel is compatible with any hardware or software SIP clients. It is also embedded in our softphones, both SIPLink and Video PC Softphone.
The communication between the client and the Tunnel Server can be either in TCP or UDP protocol and can use any port (to be specified by VoipSwitch administrator).