IceWarp VoIP Server
IceWarp VoIP Telephony Server is based on SIP protocol. It offers SIP Proxy, Registrar and Call Redirect Server in one software product. Connectivity to any external PBX Gateways to route calls across networks, based on custom call-forwarding and number rewriting rulesets.
Easy to use Call Center
Use softphones (Counterpath X-Lite 3.0), wireless Wifi phones (ZyXel Prestige 2000), wired Ethernet phones (Linksys SPA 921) or even dual-mode GSM handsets (Nokia N95).
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Low Telecommunication Costs
You don't need to know anything about SIP protocol to cut down the company's telephone bill. Simplified configuration, one-click deployment, error-free operation, no special settings required on client side make it the ideal solution for company of any size.
- Well-suited for sales departments
- Follows SIP principle
- All calls ring to all users registered for the number
- The call is rejected only when all users are offline
- The one who first picks up wins
- Great flexibility with number rewriting, presence and internal chat support
SIP Registrar, Proxy and Redirect Server in one
Parent SIP Proxy mode is intended for distributed SIP servers, interconnection with existing servers/appliances and to bridge separate local network subnets, so that the call doesn't have to go through an external router/firewall, but instead locally through the proxy. Operation of Registrar and Redirect server is fully automated.
- Proxy serves as a bridge between SIP servers or separate networks
- Forward calls to mobile networks, carrier transatlantic VoIP gateways with QoS,..
- Route calls to other IceWarp VoIP Servers in different location
Works behind firewalls and routers
- SDP NAT Traversal (Port Proxy)
- No configuration required on client side
- Automatically assigned IP address in server configuration
For Your Own Secretary
Dial from any contact list and your phone will ring when the call is placed. Works with any phone with, no proprietary communication protocols between device and application.
- Dial from Outlook contact list
- Set your available/away state by a quick number dial
- Redial last called number
- Dial the last caller number
- Get call history sent to you by email
- Use your favourite HW phone as a handset
Own Calling Plan
Free calls to local and remote VoIP users. Just 1 public VoIP number is necessary for simultaneous calls to carrier networks and conferences.
- Standalone WiFi or Ethernet SIP phones
- Dual mode mobile handsets such as Nokia N95, E61
- VoIP gateways for FXO devices
Free calls between offices regardless of geographic location. International calls to landline or mobile numbers through PSTN gateway in each office for local fee to any destination. |
Own Numbering Plan
Route calls as needed through very flexible, prefix based rules. Assign external numbers to email account IDs. Custom Quick dials to dial Last called/Last incoming, Set/Reset Away status with possibility of call forwarding. Rules can be chained for great flexibility and wide range of usage scenarios unlike HW SIP solutions. No RegEx knowledge is required with GUI interface and intuitive syntax with wildcard convention (%).
- Ultimate call control features
- Extensible call forwarding, call redirection and number rewrite rulesets
- Call forwarding (to another local/external number)
- Separate settings for incoming and outgoing calls
- Rewrite outgoing number to pretend to be a different number
VoIP Security
- Authentication through existing account
- Server access or restricted to internal users only
- User permissions to use specific gateways
Features for ISP's
- Detailed logs per user, can be parsed for billing
- Restrictions for calls going through a gateway
- Unlimited gateways
- API for extensions and system integration
Comprehensive Administration
- Simple setup
- Automatic NAT setting system
- Define/modify rules on fly
Integration Benefits
- One place authentication
- Same authentication box, policy, authentication schema, trusted hosts
- Fast setup, universal, simple for anybody