The SIPxchange ECS product line is leading the market in terms of PBX features implemented strictly following the SIP standard. We not only implement fully ratified IETF standards, but also actively participate in the standardization effort and often lead the way by implementing draft IETF standards. In addition to legacy PBX (TDM) features, there are many aspects of a SIP VoIP system that are different and hence feature implenentation has to take that into account and gracefully work in both enironments. That's exactly what we have done building SIPxchange.

Core Calling Features

• Transfer (consultative & blind)
• Call coverage
• Call hold / retrieve
• Consultation hold
• Music on Hold for IETF standards compliant phones
• Uploadable music file
• 3-way conference
• Call pickup (global and directed call pickup)
• Call park & retrieve
• Hunt groups
• SIP URI dialing
• CLID (Calling Line Identification)
• CNIP (Calling party Name Identification Presentation)
• CLIP (Call Line Identification Presentation)
• CLIR (Call Line Identification Restriction)
• Per gateway CLIP manipulation 
• 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 (dialed, received, missed)
• Auto off-hook / ring down
• Incoming only

Voice Quality

• Peer-to-peer media routing for best quality (media not routed through the SIPxchange server)
• Unmatched voice quality with lowest delay and jitter
• Support for any codec supported by the phone (incluidng video)
• Support for Polycom HD Voice
• Codec negotiation (no transcoding required)

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
• 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
• Per user Caller ID (CLID) assignment
• Per user Caller ID blocking

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
• Built-in support for U.S., German, Swiss, and Polish local dial plans (Any other local dial plan can be added as a plugin)
• Specify internal extension length
• ISN dialing based on ITAD numbers (See freenum.org) - (release 3.8)
• Redirector plugins - any imaginable dial rule can be added as a plugin (release 3.8)

Directory, Softkeys, Speed Dial (release 3.8)

• Automated generation of directory information per user or per user group
• Creation and management of many different directories (per user, per user group, per location, etc.)
• Automatic provisioning of directory information into phones
• Allows adding contacts to a directory from a .csv file (Excel)
• User configurable Speed Dial (internal / external numbers, SIP URIs)
• Speed Dial is generated by the SIPxchange server and backed up
• Auto-provisioning of Speed Dial to phones
• User configuration of Busy Lamp Field (BLF) to monitor presence of other user's phones (e.g. attendant console) 

PSTN Trunking

• Unlimited number of PSTN gateways and trunk lines
• Support for any SIP compliant gateway. Refer to the list of certified and supported models.
• Gateways can be in any location
• Gateway selection per dialing rule
• DID
• Local DID per gateway 
• DNIS
• CLIP Management 
  • User CLIP
  • Gateway default CLIP
  • Prefix stripping / appending
• Per gateway CLIR
• 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
• Route header for flexible call routing through and SBC (release 3.8)

Analog Lines (FXS)

• Supports any SIP compliant FXS gateway (refer to the list of certified models) 
• FAX support 
• Analog cordless phone support
• Plug & play management of FXS gateways from Grandstream and Cisco

Performance

• Unlimited number of simultaneous calls
• 54,000 BHCC, 100,000 BHCC redundant
• Up to 10,000 users
• Automatic time distribution of re-registration and subscription events designed to accommodate up to 10,000 users

High Availability

• Optionally fully redundant call control system
• HA based on DNS SRV (no cluster required)
• Load balance under normal operating conditions
• Geographic dispersion of redundant systems
• Real-time synchronization of state information (no calls are dropped if a server fails)
• Reports on load distribution

Call Detail Records collection and reporting

• 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
• Historic call detail record reporting in real-time (release 3.8)
• monitoring of currently active (on-going) calls (release 3.8)
• Export of active and historic CDRs to Excel (.csv file) (release 3.8)
• Direct database access for reporting application (e.g. Crystal Reports)
• SOAP Web Services access to CDR data (release 3.8)

Security

• All outbound calls authenticated through Authentication Proxy
• Secure user password management
• DoS attack prevention
• HTTPS secure Web access

System Administration Features

• Browser based configuration and management
• LDAP integration
• 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
• Support for DNS SRV
• Support for DNS NAPTR based call routing (release 3.8)
• 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 60 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

Presence Server Features (release 3.8)

• Centralized presence server based on SIP/SIMPLE 
• Centralized management of resource lists for dialog events (line state)
• Busy Lamp Field (BLF) feature based on presence
• Support for Attendant Consoles
• ACD call center agent sign in / sign out

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
• Configurable park escape key
• Allow multiple calls on one orbit

Call Center Server (ACD)

• Supports several ACD servers
• ACD server collocated or on a different server hardware
• Several queues per server
• Several lines per queue
• Support trunk lines (many calls per line) or single call per line
• Overflow queues
• Configurable call routing scheme per queue:
  • Circular
  • Linear
  • Longest idle
• Agent barge in
• Agent presence monitor using presence server
• Separate welcome and queue audio
• Call termination tone or audio
• Configurable answer mode
• Configurable maximum ring delay
• Configurable maximum queue length
• Configurable maximum wait time until overflow condition
• Unlimited number of agents per queue
• Statistics:
  • Agent statistics
  • Call statistics
  • Queue statistics
• ACD historic reporting (release 3.8)
• Supervisor authorization for agent monitoring

Required Hardware

• Intel compatible server able to run the Red Hat Enterprise Linux (or compatible) operating system
• Pingtel Appliance SIPxNano, ECS SE, ECS ME, ECS LE
• min 256 MB RAM (1GB preferred)
• No special HW required

Installation and Upgrades

• Standard Linux package management (e.g. up2date, yum)
• Single CD installation including the Linux OS 
• Graphical configuration wizard for system configuration after installation
• Automated installation and configuration of a high-availability redundant system
• Designed so that no Linux admin skills are required for installation and configuration
• Automated upgrades using standard Linux package management

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.