• Large enterprise features, small enterprise price:  Already running VoIP communications for Sterling National Bank, Swarthmore College, and other Fortune 500 companies, SIPxchange ECS packages VoIP features typically found in only large enterprise systems—advanced call control features, integrated voicemail and email, auto attendants, interactive voice response, and web-based configuration and management—into the affordable SIPxNano.
• Better voice quality: SIPxchange's voice quality surpasses the competition due to its distinctive peer-to-peer voice connectivity. This direct connectivity leads to better voice quality because no additional delays are introduced and there is no single point of failure or transcoding–just a clear, direct voice connection.
• Redundant high-availability (HA) configuration: Unique in its affordable high-availability configuration capability, the SIPxNano allows you to synchronize separate SIPxNanos in different locations for redundant high-availability configuration.  
• Unlimited number of trunk ports:
  SIPxNano allows an unlimited number of trunk ports supporting any number of FXS/FXO or digital external gateways.  SIP trunking permits the convergence of data and voice on the same network while reducing costs and eliminating the need to purchase local PSTN gateways and costly ISDN interfaces.

SIPxNano IP PBX provides award winning SIPxchange ECS software that has the same ease-of-use, reliability, scalability, and feature richness as much larger systems. This system will truly grow with your business.

 
Note: SIPxNano does not include the ECS Call Center Server component (ACD).

Hardware:

• 40GB Hard Drive
• 800-MHz VIA processor equal to 1.3GHz Pentium
• FAn-less- No moving parts; nothing to break; longer life
• Standard external I/O interfaces (serial, keyboard, mouse, USB, Line/out, Ethernet)
• Dimensions 6.7x4.9x1.5 in (17x12.4x5.8 cm)
• 30 Users, Voice Mail, Auto Attendant

Core Calling Features

• Transfer (consultative & blind)
• Call coverage
• Call hold / retrieve
• Consultation hold
• 3-way conference
• Call pickup (global and directed call pickup)
• Call park & retrieve
• SIP URI dialing
• CLID (Calling Line Identification)
• CNIP (Calling party Name Identification Presentation)
• CLIP (Call Line Identification Presentation)
• CLIR (Call Line Identification Restriction) (release 3.6)
• Per gateway CLIP manipulation  (release 3.6)
• Call waiting / retrieve
• Do not Disturb (DnD)
• Forward on busy, no answer, do not disturb
• Multiple line appearances
• Multiple calls per line
• Multiple station appearance
• Outbound call blocking
• Click-to-dial (Windows XP)
• Redial
• Call history
• Auto off-hook / ring down
• Incoming only

User Management

• Numeric or alpha-numeric User ID
• User PIN management (UI or TUI)
• Aliasing facility (numeric and alpha-numeric aliases)
• Extension and alias uniqueness assurance
• Granular per user permissions
• Call permissions:
  • 900 Dialing
  • International Dialing
  • Long Distance Dialing
  • Mobile Dialing
  • Local Dialing
  • Toll Free Dialing
  • Forward Calls External
• System permissions:
  • User has voicemail inbox
  • User listed in auto-attendant directory
  • User can record system prompts
  • User has superuser access
  • User allowed to change PIN from TUI
• Custom permissions (release 3.6)
• Supervisor permission for groups (e.g. Call Center supervisor)
• SIP password management for security
• User groups with group properties
• Per user call forwarding (follow me)
  •  To local extension, PSTN number, or SIP address
  • Parallel or serial ring
  • Allows definition of ring time before trying next number
  • Allows several forwarding destinations
  • Follow-me configuration using user portal
• Extension pool with automatic assignment
• Dial Plan
• Easy to use GUI based dial plan manipulation
• Rules based least cost routing
• Automatic gateway redundancy and failover
• Specific E911 routing
• Permission based rules
• Prefix manipulation
• Dialplan templating for international dial plans (release 3.6)
• Specify internal extension length

PSTN Trunking

• Unlimited number of PSTN gateways and trunk lines
• DID
• Local DID per gateway (release 3.6)
• DNIS
• CLIP Management (release 3.6)
• User CLIP
• Gateway default CLIP
• Prefix stripping / appending
• Per gateway CLIR (release 3.6)
• Automatic Route Selection (ARS)
• Least-cost routing (LCR)
• Automatic failover if unavailable
• Automatic failover if busy
• FAX support

SIP Trunking

• SIP call origination & termination
• Branch office routing
• Proxy to proxy interconnect using ACLs
• Least-cost-routing (LCR)
• Mixing of PSTN trunks with SIP trunks

Performance

• Unlimited number of simultaneous calls
  54,000 BHCC, 100,000 BHCC redundant
• Up to 10,000 users

High Availability

• Optionally fully redundant call control system
• Load balance under normal operating conditions
• Geographic dispersion of redundant systems
• Call Detail Records collection
• Call State Events (CSE) collected for all signaling activity
• Processing of CSEs into CDRs
• All data stored in a database at all times
• Supports redundant call control

Security

• All outbound calls authenticated through
• Authentication Proxy
• DoS attack prevention
• HTTPS secure Web access

System Administration Features

• Browser based configuration and management
• LDAP integration (release 3.6)
• SOAP Web Services interface
• CSV import of user and device data
• Integrated backup & restore
• Scheduled backups
• Diagnostics
  • Display active registrations
  • Display job status
  • Status of services
  • Snapshot logs for debugging
  • Logging (customizable log levels, message log per service)
  • Domain Aliasing (release 3.6)
  • Support for DNS SRV
  • Automatic restart after power failure

Plug & Play Device Management

• Plug & play management of phones
• Auto-generation of phone config profile
• Auto-pickup of profile by phone
• Centralized management of all phone parameters
• Centralized backup and restore of all phone config
• Auto-generation of lines by assigning users to devices
• Device group management & properties
  Firmware upgrade management

Voicemail Subsystem

• Integrated voicemail system
• Browser based user portal
  MWI
• User configurable distribution lists
• Manage Notifications:
  • Email notification of new voicemail messages
  • Forwarding of message as .wav file
  • Supports several parallel notifications
• Manage folders: Folders for message organization
• Manage greetings: Multiple customizable greetings
• Operator escape from anywhere
• Remote voicemail access
• Unlimited number of inboxes
• Up to 30 virtual media server ports per server
• Message store only limited by disk size
• Auto-removal of deleted messages
• Daily report on disk usage sent to admin

Auto Attendant Features

• Unlimited number of auto-attendants
• Customizable IVR menus with VXML
• Dial by extension and name
• Night and holiday service
• Special auto-attendant
• Transfer on invalid response
• Nested auto-attendants (multi-level)
• Fully customizable actions:
  • Operator
  • Dial by Name
  • Repeat Prompt
  • Voicemail login
  • Disconnect
  • Auto-Attendant
  • Goto Extension
  • Deposit Voicemail
• Uploadable custom prompts
• Configurable DTMF handling

Hunt Groups

• Unlimited number of hunt groups
• Serial and parallel forking
• Configurable ring time

Call Park Server

• Unlimited number of park orbits
• Music on park
• Configurable call retrieve code
• Configurable call retrieve timeout
• Automatic park timeout (release 3.6)
• Configurable park escape key (release 3.6)
• Allow multiple calls on one orbit

Pingtel Managed Devices

• Polycom SoundPoint IP 301, 430, 501, 601
• Polycom SoundStation IP 4000 SIP
• Snom 300, 320, 360
• AudioCodes TP-260/SIP, MP-114, MP-118
• Cisco ATA 186/188, 7960, 7940, 7912, 7905

SIP Implementation

• RFC 3261 Session Initiation Protocol using both UDP and TCP transports
• Advanced call control using RFCs
  • 3515 Refer Method
  • 3891 Referred-By header
  • 3892 Replaces header
• Provide for consultative and blind transfer and third party call controls
• RFC 3263 Locating SIP Servers - use of DNS SRV records for call routing control and server redundancy.
• RFC 3581 Symmetric Response Routing (rport)
• RFC 3265 SIP Event Notification - for phone configuration and
  RFC 3842 Voice mail message waiting indication (MWI)
• RFC 3262 Reliable Provisional Responses
• RFC 2833 Out-of-band DTMF tones
• RFC 3264 Offer/Answer model for SDP for Codec Negotiation
• Early media (SDP in 180/183)
• Delayed SDP (SDP in ACK)
• Re-INVITE: Codec change, hold, off-hold
• Route/Record-Route header fields
• Configurable RTP/RTCP ports
• Configurable SIP ports
• Note – Some features are dependant upon other SIP components such as Phones and Gateways.