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  • SIP Session Router, optionally geo-redundant and load sharing
  • Media server for unified messaging and IVR (auto-attendant) services
  • Conferencing server based on FreeSWITCH
  • XMPP Instant Messaging (IM) and presence server (based on Openfire)
  • Contact center (ACD) server
  • Call park / Music on Hold (MoH) server
  • Presence server (Broadsoft and IETF compliant resource list server for BLF)
  • New: Shared Appearance Agent server to support shared lines (BLA)
  • Group paging server
  • SIP trunking server (media anchoring and B2BUA for SIP trunking & remote worker support)
  • Call Detail Record (CDR) collection & processing server
  • Third party call control (3PCC) server using REST interfaces
  • Management and configuration server
  • Process management server for centralized cluster management


  • Centralized deployment: Branch only provides phones and optionally PSTN gateway for failover, reduced WAN BW consumption or E911 calls
  • Distributed deployment: Branch provides full call server with SIP site-to-site dialing between offices
  • Branch office locations can be defined in the mgmt UI with a postal address
  • Users, phones, gateways, SBC, and servers can be assigned to a branch location
  • A PSTN gateway can be available for calls that originate in a specific branch only or for general use
  • Source routing allows call routing based on location (branch local calls are routed through local gateway preferably)
  • Branch postal address automatically proliferates to user's office address
  • Survivable branch configuration possible with Audiocodes gateways SAS functionality (auto-configured)
  • Certain sipXcom services can be deployed in the branch as part of the cluster (e.g. conferencing) (for a future release we will add full survivable branch capability based on the sipXcom cluster - see XX-4819)

Enterprise Instant Messaging (IM) and Presence

  • XMPP based IM and presence server based on OpenfireNew role in sipXcom cluster management for IM/presence
  • Supports XMPP standards based clients
  • Auto-configuration of user's IM accounts
  • Auto-configuration of IM user groups
  • Personal group chat room for every user auto-configured
  • Federation of phone presence with IM presence
  • Customizable "on the phone" presence status message
  • dynamic Dynamic call routing based on user's presence status
  • Message archiving and search for compliance (pending)
  • Server-to-server XMPP federation
  • Optional secure client connections
  • Client-to-client file transfer
  • Group chat rooms
  • XMPP search
  • Integration of user profile information and avatar (pending)

Personal Assistant IM Bot

  • My Buddy Personal Assistant feature
  • Dynamic call control using IM
  • Dynamic conference management using IM
  • Unified messaging management using IM
  • Enables Fixed Mobile Convergence (FMC) application
  • Call history / missed calls
  • Call initiation using corporate dialplan
  • Corporate directory look-ups


  • Server side federation with other public XMPP IM systems (based on Kraken IM Gateway) including:
  • Yahoo, AIM, MSN
    • ICQ, IRC
    • IBM Sametime
    • Facebook IM
    • MySpace IM
  • Server-to-server federation with Google Talk
  • Allows group chat sessions across systems
  • Allows message archiving (if enabled) across systems
  • User
  • Allows group chat sessions across systems
  • Allows message archiving (if enabled) across systems
  • User self-administration of credentials for other IM systems

Fixed Mobile Convergence (FMC) Application

  • 3rd Party FMC application with the following functionality:
    • Enterprise number dialing
    • System call-back saves on wireless toll charges
    • Corporate directory look-ups
    • Call history
    • Presence sharing
    • IM
  • Server-to-server (XMPP) federation with GTalk allows using GTalk client on smart phones with sipXcom
  • Dynamic conference bridge control

Web Conferencing & Collaboration


  • Commercial options available through eZuce to link to SeeVogh's viewme and viewme Cloud products

User Self-Control (User Web Configuration Portal)

  • Every user on the system gets access to a personal Web user portal for self-management and control
  • Management of unified messaging (voicemail)
  • Configuration of unified messaging preferences
  • Time based find-me / follow-me
  • Flexible configuration of call forwarding
  • Management of personal profile data including avatar
  • Personal call history
  • Personal phone book, speed dial and presence managementAllows contact upload from GMail and Outlook
  • Click-to-callACD presence and supervision capabilities
  • Individual phone management
  • Personal auto-attendant
  • Dynamic conference management w/ click-to-conference
  • Management of personal IM account
  • Setup of DimDim Web conferencing
  • Personal MoH music upload and preferences


  • Create a user, provision a phone and assign a line in only three clicks - easy!
  • Numeric or alpha-numeric User ID
  • User PIN management (UI or TUI)
  • Aliasing facility (numeric and alpha-numeric aliases)
  • Extension and alias uniqueness assurance
  • Management or auto-assignment of user's IM ID and display name
  • Automatic IM buddy list creation based on user groups
  • Granular per user permissions
  • Call permissions:
    • 900 Dialing
    • International Dialing
    • Long Distance Dialing
    • Mobile Dialing
    • Local Dialing
    • Toll Free DialingForward 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
    • User can use Microsoft Exchange 2007 VM
    • User has a personal auto-attendant
  • Custom permissions as defined by the admin
  • Supervisor permission for groups (e.g. Call Center supervisor)
  • Management of user contact record (user profile)
  • Comprehensive profile data
  • Work and home address
  • In-building location information
  • Assistant information
  • Support for avatar including support for gravatar
  • SIP password management for security
  • User groups with group properties
  • Per user call forwarding (follow me)
    • To local extension, PSTN number, or SIP address
    • Based on user or admin defined time schedules
    • 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
  • Time-based dialing rules with different admin defined schedules
  • Rules based least cost routing
  • Dynamic call routing based on user's IM presence status
  • Directly route to voicemail on IM status DND
  • Dynamically add forwarding destination based on phone number in custom presence status
  • Automatic gateway redundancy and fail-over
  • 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
  • Specific rule for site-to-site call routing between SIP systems
  • Redirector plugins - any imaginable dial rule can be added as a plugin


  • Unlimited number of PSTN gateways and trunk lines
  • Supports any most SIP compliant gateway gateways (e.g. Cisco, Audiocodes, Mediatrix, VegastreamSangoma, Patton, etc.)
  • Gateways can be in any location
  • Gateway selection per dialing rule
  • Source routing of calls so that calls can be routed through a local gateway to save WAN bandwidth
  • DID
  • Local DID per gateway
  • DNIS
  • CLIP Management
    • User CLIP
    • Gateway default CLIP
    • Prefix stripping / appending
  • Per gateway CLIR
  • Automatic Route Selection (ARS)
    • Implemented with XML-formatted mapping rules.
    • Mapping values re-write SIP URLs to specify the next hop or destination for a SIP message that has been received by the Communications Server component.
    • Direct messages to different SIP/PSTN trunk gateways, either on premise or at a remote premise location, based on any portion of SIP URL or E.164 number.
    • Route messages to commercial SIP/PSTN service providers, which reduces or eliminates the need for on-premise trunk gateways.
  • Least-cost routing (LCR)
  • Automatic failover if unavailable
  • Automatic failover if busy
  • Inbound FAX support
  • Mixing of PSTN and SIP trunks with least cost routing

SIP Trunking

  • Basic SIP trunking gateway w/ NAT traversal
  • Remote worker support w/ near-end and far-end NAT traversal and auto-detection
  • ITSP templates for simplified configuration
  • Interop (not certified) with the following ITSPs:
    • BT (UK)
    • AT&T
    • CBeyond
    • Bandtel
    • CallWithUs
    • Eutelia (Italy)
    • LES.NET
    • SIPcall (Switzerland)
    • Vitality
    • VOIPUser (UK)
    • VOIP.MS
    • Appia
  • Easy configuration templates exist for the above ITSPs
  • Many other ITSPs are compatible, see ITSP interop in Wiki
  • SIP interop with Nortel CS1000 R6
  • SIP call origination & termination
  • Branch office routing
  • Proxy to proxy interconnect using ACLs
  • Least-cost-routing (LCR)
  • Mixing of PSTN trunks with SIP trunks
  • TLS support for secure signaling
  • Route header for flexible call routing through an SBC
  • Flexible rules for SBC selection (route selection)
  • Support for Skype for Business SIP trunking

Integration with Microsoft


Active Directory and Exchange


  • Synchronization with Microsoft Active Directory
    • Using LDAP interface
    • On demand or automatically based on a schedule
    • Graphical query design combines ease of use with flexibility
    • Allows preview of records to be imported
  • Dialplan integration with Microsoft Exchange 2007 voicemail server
    • Allows mixed environment with groups of users on Exchange or the sipXcom VM server
    • Permission based selection of VM server per user or user group
    • Automatic dialplan routing to Exchange VM
  • Enables sll speech based Exchange 2007 capabilities

Supported Softclients

  • Combined SIP / XMPP clients:
    • Counterpath Bria professional
    • Jitsi
  • Provisioning server for automated mass deployment
  • Automated SIP and XMPP account setup
  • Call recording
  • Supports BLF (workgroups)
  • Scheduled to support BLA
  • Automatic user profile and directory


  • Counterpath X-Litemanagement
  • XMPP clients:
    • Pidgin
    • Google Talk
    • Trillium
    • Spark
  • SIP clients:SIP Communicator
    • 3CX softphone

Analog Lines (FXS)

  • Supports any SIP compliant FXS gateway
  • FAX support
  • Analog cordless phone support
  • Supports analog Polycom speakerphones
  • Plug & play management of FXS gateways from Audiocodes , and Grandstream and Cisco


  • Unlimited number of simultaneous calls (voice, HD voice, video) - only depends on LAN/WAN bandwidth
  • 54,000 BHCC, 120,000 BHCC two-way redundant (depends on server HW)
  • Up to three-way redundant configuration using cluster mgmt Web GUI
  • Up to 10,000 users per dual-server HA system
  • Tested up to 10,000 IM users
  • 450 simultaneous calls through the SIP trunking gateway require < 20% CPU on dual core system
  • Up to 500 simultaneous conferencing ports per server
  • Up to 300 media server ports for unified messaging (supports 15,000 users)
  • Automatic time distribution of re-registration and subscription events


  • Browser based configuration and management
  • Several admin accounts
  • Notification when new version or patches are available
  • GUI based software upgrade
  • GUI based certificate management
  • LDAP integration
  • Integration with Microsoft Exchange 2007 for voicemail and Active Directory
  • SOAP Web Services interface
  • CSV import and export of user and device data
  • Administration of Instant Messaging (IM) and Presence settings
  • 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)
    • Display active calls
  • Domain Aliasing
  • Support for DNS SRV
  • Support for DNS NAPTR based call routing
  • Automatic restart after power failure
    • Single sipXcom application can start all other application processes associated with starting up sipXcom, including dependent processes that must be started in particular order.
    • Configured from browser interface
  • Server statistics (integrated graphs and SNMP)
  • Login history report (successful and unsuccessful)
  • Automated testing of network services (DHCP, DNS, NTP, TFTP, FTP, HTTP) for proper configuration
  • Downloadable test tool to run network services tests from a Windows laptop

Plug & Play Device Management


  • Integrated unified messaging system
  • Localized per user by installing language packs
  • Number of voicemail boxes only limited by disk size (tested up to 10,000)
  • Performance tested up to 300 simultaneous calls (ports) on dual core server
  • IMAP back-end connection
  • Acts as an IMAP client into MSFT Exchange and other compatible email systems
  • User manageable credentials for IMAP federation
  • Properly controls MWI on the phone when message is "read" using the email client
  • Browser based user portal for unified messaging management
  • RSS feed for new messages
  • Message Waiting Indication (MWI)
  • User configurable distribution lists
  • Group and system distribution lists
  • Unified Messaging:
  • Email notification of new voicemail messages
  • Forwarding of message as .wav file
  • Supports several parallel notifications
  • IMAP client into Exchange
  • Per user selectable templates for email format used when forwarding voicemail
  • Manage folders: Folders for message organization
  • Manage greetings: Multiple customizable greetings
  • Operator escape from anywhere
  • Remote voicemail access using a phone
  • SOA Web Services (REST) access to messages and greetings
  • Unlimited number of inboxes
  • Auto-removal of deleted messagesDaily report on disk usage sent to admin

Personal Auto Attendant

  • User configurable personal auto-attendant for every user on the system
  • Up to 10 individual forwarding choices (keys 0 through 9)
  • User can record greeting that corresponds with key configuration
  • Individual zero-out to a personal assistant or receptionist
  • Individual selection of language based on installed language packs
  • Personal greeting


  • Unlimited number of auto-attendantsCustomizable 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

  • Centralized presence server based on SIP/SIMPLE
  • Compatible with Broadsoft or IETF implementations
  • Centralized management of resource lists for dialog events
  • Busy Lamp Field (BLF) feature based on presence
  • Used to support shared lines (BLA)
  • Presence federated with IM presence to show "on the phone" status
  • Support for Attendant ConsolesACD call center agent sign in / out3rd party Attendant Consoles (such as Voice Operator Panel)

Hunt Groups

  • Unlimited number of hunt groups
  • Serial and parallel forking (rings sequentially or at the same time)
  • Configurable ring time per attempt
  • Enable / disable user call forwarding rules while hunting
  • Flexible configuration of destination if no answer


  • Voice conferencing server that can run on the same sipXcom server or on dedicated hardware
  • Support for voice conferencing
  • Each user on the sipXcom system can have a personal conference bridge
  • Recording of conference calls
  • Dynamic conference controls from the user's Web portal (user portal)
  • Dynamic conference control using IM
  • Participant entry / exit messages
  • Roll call
  • Mute, isolate, disconnect, invite
  • Association of personal conference bridge with personal group chat room
  • Automatic migration of group chat to a voice conference using the @conf directive
  • Support for HD Audio and transcoding if necessary
  • Support for up to 500 ports of conferencing, dependent on hardware
  • Configurable DTMF keys for conference controls using the TUI
  • A sipXcom IP PBX system can have more than one conference server if more capacity is needed
  • All conferencing servers and services centrally managed and configured
  • Conferencing based on FreeSWITCH



Queueing (ACD)


  • ACD server collocated or on a different server hardware
  • Several (unlimited) queues per server
  • Several lines per queue
  • Support trunk lines (many calls per line) or single call per line
  • Dedicated overflow queues or overflow to hunt group or voicemail
  • Configurable call routing scheme per queue:
    • Ring all
    • Circular
    • Linear
    • Longest idle
  • Agent barge in (early termination of welcome message if agent becomes available)Agent presence monitor using presence server
  • Separate welcome and queue audio
  • Call termination tone or audio
  • Configurable answer mode
  • Agent wrap-up time configurable per queue
  • Auto sign-out of agents if calls are not answered
  • 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
  • ACD historic reports for agents, calls, queues
  • All reporting stored in database for post-processing if neededagents per queue

sipXcom Managed Devices

Any Almost any SIP compatible phone works with sipXcom if configured manually (i.e. by logging into the phone's Web interface to configure it one phone at a time). The following devices are plug & play managed automatically and centrally by sipXcom:

  • Polycom SoundPoint all models (IP 301, 320, 330, 430, 450, 501, 550, 560, 601, 650, 670)
  • Polycom SoundStation IP 4000, 6000, 7000 SIP
  • Polycom VVX phones (300/310, 400/410, 500, 600, 1500)
  • Audiocodes gateways MP112, MP114, MP118, MP124 FXS
  • Audiocodes gateways FXO and PRI/BRI
  • IPDialog SIPTone V
  • Counterpath Bria Professional
  • Avaya 1210, 1220, 1230, 1235 with SIP firmware

sipXcom Managed Devices (


Community supported)

Experimental support Community supported means that the phone plugin for plug & play management is provided as is. These phone plugins are less frequently updated to the latest firmware and are less testedprovided and maintained by community members. Some system functionality might not be implemented or supported.


  • Intel / AMD x86 compatible server
  • Min RAM 2 4 GB or more
  • Linux operating system (RHEL, CentOS or SuSE)
  • 32 bit and 64 bit versions availablePowerPC (PPC) supported on SuSE (need to compile yourself)
  • No special HW required, sipXcom uses external gateways


  • Automated installation from CD ISO for OS and sipXcom IP PBX application
  • Graphical configuration wizard for system configuration after installation
  • Self-signed Certificate generation (allows installing a signed certificate if desired)
  • GUI based upgrade management from the admin Web interface
  • Standard Linux package management (e.g. up2date and yum)
  • Optional auto-configuration of DNS, DHCP, NTP, FTP, TFTP, HTTP servers
  • Designed so that no Linux admin skills are required for installation and configuration