Enswitch features

This is a list of major Enswitch features. It is not exhaustive; minor features are too many to list.

You can also view a list of new features in the latest version of Enswitch.

Marketing and billing Customer System Security Optional

Marketing and billing features

Integrated pre-paid, post-paid, and external application billing of calls in and out, numbers, telephone lines, etc.
Customers can be given the full PBX interface, a simplified multi-line residential interface, or a simplified wholesale/trunking interface.
The system owner and resellers can administer prices, numbers, resellers, etc, on the web. Prices can be in any currency, and the system will do conversions automatically. The web interface supports SSL (requires third party certificate). The web interface can be re-branded to your own design on a per rate plan basis, allowing multiple brandings within the same installation.
Customers, including resellers, cannot see which other customers are on the system, or even whether their parent reseller is the system owner or not. This makes the system truly multi-tenant.
The system owner and resellers can set their own prices for outbound calls, numbers, telephone lines, etc. Each reseller can define as many rate plans as they wish, and assign different customers different plans. The system owner creates groups of destination patterns, e.g.. 0044 for the UK, and resellers set how much markup they wish to charge their customers. Call prices can have start and end dates as well as priorities. This allows future prices to be loaded in advance.
Rate plans can inherit pricing from other rate plans, allowing fine-grained control of pricing for different customers without needing to work with large data sets.
Connection to unlimited PSTN carriers with different carriers to different destinations. Each carrier can have prices set (in any currency) for each destination, and automatic least cost routing (LCR) can choose the cheapest carrier for each destination. Prices can imported and exported from/to a spreadsheet. Routing to carriers can be configured per rate plan, allowing higher quality routes for premium customers. Routing can be controlled independently for voice calls, fax calls, and text messages. Automatic failover and load balancing between carriers can be easily configured. Routing to carriers can be done on both called number and callerid. Call completion statistics are gathered for each destination for each carrier. Carriers can be limited to particular customers. Resellers can optionally be allowed to control routing for their customers.
Production of PDF and XML/XSLT invoices for each customer. Invoices can be emailed automatically to customers, or to resellers for forwarding. Invoices can be reviewed, edited, and approved within the web interface before being sent. Customers and resellers can search and download all past invoices on the web interface, and resellers can manage which invoices are paid, due, or overdue. Customers can choose which currency their invoices appear in. PDF invoices can be downloaded as a single file for batch printing. Customers can be warned automatically when invoices become overdue. The appearance of invoices can be controlled using HTML templates, with different templates for different sets of customers.
The maximum number of each feature can be set per customer, and a number of each feature can be included free. For example, a customer could be given 5 free telephone lines, then charged 9.95 per month for any extra up to a maximum of 20, when they may add no more. Prices can be monthly, 3 months, 6 months, or annual.
Customers can be limited to a certain number of concurrent calls. This can be set for each individual customer, and can be set for inbound, outbound, and total calls. When customers reach their limit, additional calls can either be refused or can be allowed at additional cost. Customers can purchase packages of extra concurrent calls. Limits can be ignored for emergency calls.
Rate plans can optionally include a number of minutes per month. Once used up, call prices revert to normal. Minutes call option roll over for one month. Which destinations use the included minutes can be configured on the web interface, and different rate plans can use different sets of destinations.
The system owner and resellers can create products for customers to purchase. Products can be physical (such as SIP phones), services, or blocks of minutes. Billing and invoicing for products is integrated into customers' accounts. Recurring billing is supported.
Other charges can be attached to customers' accounts for extras such as telephone lines, DSL circuits, etc. Charges can be one-off, monthly, quarterly, or annual. Charges are integrated into billing and invoicing. Charges can be automatically pro-rated.
Affiliates can be created, where the system owner and resellers can give a percentage of revenue from one customer (including resellers) to another customer. Affiliates are given a breakdown of where their revenue came from on their invoices.
Customers may choose their telephone numbers from a list, and forward them to telephones, external numbers, hunt groups, IVRs, etc. Different numbers can be priced at different rates, both by area code and by number vanity (e.g. 1-800-555-5555 is more expensive than 1-888-713-2894).
Both inbound and outbound calls can be charged at different rates at different times of the day and different days of the week.
Calls can be re-rated at a later time according to updated pricing.
Customers can create SIP addresses of the form <user>@<domain> for inbound calls.
New customers can sign up using a self service wizard. Resellers can set which rate plans customers may choose from on sign-up.
Calling cards. Resellers can set their own prices and choose features available to users. Users are cut off when credit runs out, and resellers can set whether a warning is played to the user, called party, or both. Both inbound and outbound calls can be billed simultaneously. Users can make multiple outbound calls from one inbound call. Millions of cards can be configured on one system.
Call shops, where a customer can walk in, pay, and make calls from one of the shop's telephones.
PayPal and direct debit integration for customers to top up their accounts. Credit card integration is available at extra cost.
Sales tax, VAT, GST, USF fees, etc, can be automatically calculated and added to invoices, transactions, and CDRs. Multiple taxes can be applied, as either a percentage or a fixed amount. Customers and calling cards can be marked as either liable or not for each tax.
The system owner and resellers can produce pre-paid vouchers to sell through retail channels. Customers can redeem these vouchers automatically on the web interface or via telephone.
The system owner can create an unlimited call IVR which allows callers to call in without authentication. They are then allowed to call any destination which costs the system owner less than the revenue they receive for the incoming call. The system owner can set allowed destinations via a minimum profit margin and percentage, maximum outbound call costs, and fine tune using a class of service.
Wholesale customers can be created and administered purely from the web, with authentication via SIP username and passwords, source IP addresses, or called number prefixes.
The system owner can create new user roles, and define which menus they may see.
The system owner and resellers can send broadcast emails to all their customers, or an uploaded text file of addresses.
A set of customers can be marked as having shared telephony features, while maintaining separate billing. This allows a customer to route calls between PBX features in different departments, while billing each department for their calls.

Customer features

All of the following work seamlessly with multiple levels of customers and resellers. Customers can define their own settings if you choose to allow them to, and they and any resellers are billed automatically.

Per customer centrex numbers forwarded to sip phones, voicemail, etc. Each customer can define their own feature codes, and multiple customers can define the same code with different destinations. Customers can change feature codes for voicemail, etc, as they wish. This reduces training costs when switching from a legacy PBX.
AJAX based receptionist control panel with drag and drop to redirect calls. This can optionally be read-only and in its own pop-up window.
Calls can be authenticated by username and password, source IP address, callerid, account and PIN entered in an IVR, or called number prefix.
Telephone features include telemarketer block, callerid block, selectable callerid for both internal and external calls, call forwarding, variable ring time, do not disturb, call park, distinctive ring based on caller, advice of charge (AOC), etc. Forwarding can also be controlled from the PSTN via a telephone menu.
Voicemail with external access, email notification, web access to messages, and multiple greetings (unavailable, busy, and temporary). Greetings can be uploaded and downloaded on the web interface. Notification of voicemails can be via MWI, email (optionally encrypted using GPG), SMS, and outbound calling, and work with messages left via telephone, web, or JSON API. Users can be warned by email when their mailboxes are nearly full.
Voicemail distribution lists can be created, controlled from the web interface.
Calls can be forwarded to voicemail based on RDNIS.
Multi-level IVRs and auto attendants. Every key on the telephone keypad, as well as time out, can be assigned to any feature on the system, or to an external number. External calls are billed to the IVR owner. The messages played to callers can be recorded from a telephone or uploaded in a .wav or .mp3 file.
Queues/ACD. Unlike Asterisk's queues, calls can be queued across multiple machines, with the machines voting on which call gets forwarded to an agent next. Should one machine crash, calls on other machines are moved up the queue. Destinations can be telephones, external numbers, and SIP URIs. Queues can be assigned priorities, with calls on high priority queues delivered first. Callers can be offered the option to drop out, and be called back when they approach the head of the queue. Reports can be generated per queue and per agent, and queue data can be exported for import into QueueMetrics.
Conferencing. Customers can set when the conferences run, how many people may join, different PINs for administrators, talkers, and listeners, and a set of telephone numbers and email addresses to notify when conferences start. Conferences can be recurring on a daily, weekly, bi-weekly, monthly, or bi-monthly basis. Numbers can route straight into a individual conference, either with or without a PIN.
Fax to email and fax to mailbox with notification via email (optionally encrypted using GPG) and SMS.
Parallel, serial, and circular hunt groups. Hunt groups can call telephones, external numbers, and SIP URIs. Hunt groups include many find me/follow me features.
Group and number pickup.
Per telephone line and shared speed dials.
User settings can be bulk imported from a .csv file.
Virtual telephones, where a telephone line can be logged in on top of a physical telephone and change its settings. Virtual telephones can move between physical telephones for a "hot desk" environment. This is sometimes known as "virtual extensions" or "extension mobility". Virtual telephones can be automatically logged out nightly.
Calls can be routed by date and time of day. Dates and times can be specified up to 20 years in advance, and can be any combination of times, days of the week, days of the month, months, and years. Routing can be changed instantly via web or telephone.
Routing of calls by callerid, by exact number, area code, country, etc. For example, all calls from a region can be routed to the branch in that region.
Routing of calls by regular expressions on called/calling numbers, and digits entered by the caller.
Routing of calls by 3rd party web service. This can be configured by customers, allowing them to control their numbers from their own web server.
When forwarding calls to external numbers, DTMF can be automatically played to the called party to call a particular extension.
Call recording, with comprehensive search and listen on the web. Customers can be billed for both recording and storage. Recording can be done for all calls, a percentage of calls, or on demand at the start of a call. Recordings can be emailed to an external address at the end of each call. Call recording can be started and stopped on demand via DTMF.
Customers can upload their own music. Each music source may contain multiple files.
Speed dials, both shared and specific to one telephone line.
Remote reboot of handsets, if supported by handset model.
Call back to authenticated callerids.
Call screening, where called parties are asked if they wish to accept calls. Callers can be asked to record their names, their callerid can be played, or a message specific to call flow can be played. This is integrated with billing, so answered call legs are billed even if the call is rejected by the called party. Call screening can optionally use Answering Machine Detection (AMD).
Paging, with auto-answer on compatible SIP telephones.
Dial by name, integrated with the main user database and voicemail recorded names.
Busy lamps. Unlike other platforms, these work in a cluster where more than one machine is delivering calls.
Call spy. Unlike other platforms, this works in a cluster where more than one machine is delivering calls.
Customers and resellers can view invoices, transactions, and CDRs (history of calls made) on the web, and download transactions and CDRs to a spreadsheet. CDRs include real time call costs. The flow of complex calls with multiple CDRs can be shown for ease of understanding. This flow can be in tabular format or a plain English description.
Customers can send broadcast emails to everyone in their customer, or an uploaded text file of addresses.
Missed calls to telephones can be viewed on the web, Android, and iOS clients (with notifications). Email notifications of missed calls can be sent.
Customers can configure email and text message alerts for abnormal or potentially fraudulent calls within their customer. The system owner can also create alerts for the entire system. Alerts sent can be stored for later viewing on the web interface.
Customers are automatically notified when their balance drops below a set amount, and they can have the system automatically request a top-up when this happens.
Each user can be in a different time zone, set on the web interface. All dates and times the user sees on the web and invoices are in their local time zone.
If the system owner allows, customers can port in numbers from other providers and configure them on the web interface themselves.
Click to call on the web interface and JSON API. This can be used for both outgoing and incoming calls. Incoming calls can be routed to any feature of the system such as telephones, queues, hunt groups, etc, making creation of "Click here to call us" links on customers' own websites easy. The customer is billed for any chargeable calls.
Bulk dialer for outbound calling campaigns. Customers can manage their own campaigns and sets of numbers to call. Answered calls can be routed to any feature within the same customer. Campaigns can start and stop automatically based on time and/or date. The bulk dialer can optionally use Answering Machine Detection (AMD). Bulk dialer calls can be billed at a different rate then normal calls.
Events for customers' calls can be sent to a remote web server, allowing highly flexible reporting, CRM integration, wallboarding, etc.

System features

Runs on a single machine, or an Asterisk and Kamailio cluster. Supports high availability and failover, allowing any single machine to crash with only a few seconds interruption to service. The cluster architecture allows extra machines to be added at will.
Clusters can be geographically distributed. Failover can be implemented between locations. Customers can register their telephones to their local telephony cluster for low latency calls, with all customers administered from a central location.
Telephones on the same system can be in different countries, and each use their local country's dial plan. Customers can set this on a per telephone basis.
Telephones can be registered to Enswitch, to a third party registrar, or to a mixture.
Telephones can be behind NAT, with Enswitch providing the facilities of a Session Border Controller (SBC). SIP OPTIONS ping of telephones can be controlled per telephone.
Connects to any third party system supported by Asterisk. This includes all common SIP telephones and softphones, all common SIP PBXs, traditional PBXs and PSTN providers supporting Primary Rate ISDN over E1 or T1, analogue telephone lines, and others.
The system owner can switch between carriers on a per route basis on the fly. If a carrier starts dropping calls, switch to a different provider in a few seconds. Routes can be changed in bulk if many routes need changed. Failsafe routing can also be configured so that if a carrier is down, calls are automatically routed to a backup carrier.
The system owner can create AGI plugins to perform arbitrary actions, such as call routing based on database lookups. Customers can then route calls to plugins on the Enswitch web interface.
The system owner can dynamically add new menus to the system for local features, links to external websites, etc. Different sets of customers can see different menus.
A comprehensive JSON API. This is capable of all the same actions as the Enswitch web interface, and handles authentication and data validation.
Events across the cluster can be accessed by websocket, or posted to a remote web server, allowing extensive and highly flexible monitoring and reporting.
Integrates with E911 providers; a legal requirement for ITSP services in the USA.
A plugin can be invoked to do caller name (CNAM) lookups for incoming calls. The results can optionally be cached.
Number portability with millions of entries.
Conferencing works seamlessly across multiple Asterisks without needing a dedicated conference server. The Asterisk machines elect one machine for each conference, and the others forward calls to it. Should it crash, callers can call back in and a different machine will be used.
Statistics of call volumes, with minimum, maximum, and average call volumes for all calls, SIP calls, and E1/T1 calls on all machines in the cluster, or any individual machine, between any two given dates.
System configuration and web interface menu structure can be viewed and edited on the web interface.
All system sound files are provided in high quality G.722, standard quality G.711, and compressed G.729, to reduce transcoding CPU requirements.

Security features

All calls are required to be authenticated using SIP username and password, source IP address, account and PIN, etc, before being allowed to call externally.
SIP registration and calls can be limited to individual source IP addresses, or multiple ranges of source IP address per telephone line. The source IP address can be automatically detected on first registration, and automatically locked to that address.
SIP packets from known scanners are automatically rejected. Optionally, only certain SIP user agents may be allowed.
The system owner and resellers can limit which destinations customers may call using classes of service.
Customer administrators can limit which destinations users may call using classes of service. Classes can be defined down to an individual number. Classes can be controlled by time of day, day of week, etc.
Once a customer's balance reaches their credit limit, they can make no more billable calls. The credit limit can be controlled per customer. A temporary credit limit can be applied to each customer, controlled by time of day, day of week, etc.
Each customer and telephone line can have a daily spending limit set. If they exceed this, all their calls are locked until manually unlocked. They are automatically sent a warning email when approaching their limit.
Each customer can have a maximum number of concurrent calls.
Each peer (carrier) can have a maximum number of concurrent calls.
Alerts can be sent by email or text message for abnormal or potentially fraudulent calls, based on destination, duration, cost, number of concurrent calls, etc.
Web interface supports SSL (requires third party certificate).
Web logins can be limited to multiple ranges of source IP address per account.
All user actions are recorded in audit logs.

Optional components at no extra cost

Android, iOS, and Windows clients.
Click to call plugin for Google Chrome.
Credit card integration for Authorize.net, Barclays, Beanstream, Commweb, and HSBC. When using Authorize.net, credit card numbers are only stored on Authorize.net's servers.
Integrated provisioning for Aastra, Cisco 7940/7960, Digium, Grandstream, Linksys SPA, Snom, Polycom (via either TFTP or HTTP), and Yealink handsets. Provisioning templates can be created, edited, and managed via the web interface. Other manufacturers are available on request.
Outbound faxing from the web interface and Hylafax clients. Outbound calls are billed to the appropriate customer, and the user is notified by email when the fax has been sent. Faxes can include a cover page. Sent faxes can be saved within the system and later viewed on the web interface.
SIP messaging with store and forward. If a telephone is off-line, the system will store the message until it is available. If you have an external SMS gateway, users can send messages to external numbers and be billed accordingly. Stored SMSs can viewed on the web interface.
SMS from the web and JSON API, and for notification of voicemails and faxes. Requires an external gateway.
Plugins to integrate with third party Enswitch servers, and Voxbone, to access their number ordering APIs. This allows Enswitch's pool of spare numbers to be automatically kept topped up.
TR-069 CWMP provisioning.
WebRTC server.

Optional components at extra cost due to custom development requirements

Credit card integration for other vendors.
Number ordering plugins for other vendors.

Optional components from third party vendors

Jabber/XMPP instant messaging server. Authentication is integrated with Enswitch, and telephone status affects IM presence. Different Enswitch customers can have different domains.
Customer premises survivability solution using TelvivaSync and Runway gateway. Optional ViBE support for optimal bandwidth usage, voice and data management and call continuity.
Voice Operator Panel, a SIP softphone for operators and receptionists with Outlook/LDAP/XMPP/MSN/CRM integration.
Click to call plugin for Chrome.
Digium fax driver.
G.729 codec driver.
QueueInsight integration.
QueueMetrics integration.