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System Application Services

All the sipXecs sipXcom application services are allocated to specific server roles. Using the centralized cluster management system each role can be instantiated on a dedicated server or several (all) roles can be run on a single server. Configuration of all services and participating servers is fully automatic and Web UI based.

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  • 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 sipXecs 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 sipXecs sipXcom cluster - see XX-4819)

Enterprise Instant Messaging (IM) and Presence

  • XMPP based IM and presence server based on Openfire
  • New role in sipXecs 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 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)

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  • 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 sipXecssipXcom
  • Dynamic conference bridge control

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Note

Since Salesforce.com acquisition Dimdim is no longer an open product. Therefore starting with sipXecs sipXcom 4.4 version Dimdim integration was removed.

  • Integration with DimDim for Web conferencing
  • Automatically launches DimDim with correct voice conferencing info
  • Creates link to conference that can be sent to participants
  • Allows easy setup of DimDim personal account / stores DimDim credentialsCommercial options available through eZuce to link to SeeVogh

User Self-Control (User Web Configuration Portal)

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  • Peer-to-peer media routing for best quality (media not routed through the sipXecs sipXcom server)
  • Unmatched voice quality with lowest delay and jitter
  • Support for any codec supported by the phone or gateway (including video)
  • Support for HD Voice (Polycom and other phones)
  • Codec negotiation (no transcoding required)
  • Conferencing, auto-attendant and voicemail support HD voice w/ transcoding if necessary

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  • 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
  • ISN dialing based in ITAD numbers. See freenum.org
  • ENUM support
  • Specific rule for site-to-site call routing between SIP systems
  • Redirector plugins - any imaginable dial rule can be added as a plugin

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  • 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 with the following ITSPs:
  • BT (UK)
  • AT&TSkype
  • Bandwidth.com
  • CBeyond
  • Bandtel
  • CallWithUs
  • Eutelia (Italy)
  • LES.NET
  • SIPcall (Switzerland)
  • Vitality
  • VOIPUser (UK)
  • VOIP.MS
  • VoxitasAppia
  • Easy configuration templates exist for the above ITSPs
  • Many other ITSPs are compatible, see ITSP interop
  • 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 Server 2000 / 2003 and Exchange 2007/2010/2013

  • 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 sipXecs 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

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  • Call State Events (CSE) collected for all signaling activity
  • Processing of CSEs into CDRs
  • All data stored in a database at all times
  • Flexible report generation using Jasper Reports, built-in
  • Supports redundant call control
  • Determines and records call type information
  • Internal / external calls
  • Calls to specific sipXecs sipXcom services
  • Collates call legs
  • Historic Call Detail Record reporting in real-time
  • Additional reports using call type info
  • Monitoring of currently active (on-going) calls
  • Export of active and historic CDRs to Excel (.csv file)
  • Direct database access for reporting application (e.g. Crystal Reports, Jasper Reports)
  • SOAP Web Services access to CDR data
  • Individual call history per user in the user portal

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  • 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 sipXecs sipXcom application can start all other application processes associated with starting up sipXecssipXcom, 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

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  • Voice conferencing server that can run on the same sipXecs sipXcom server or on dedicated hardware
  • Support for voice conferencing
  • Each user on the sipxecs 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 sipXecs 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

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  • Supports several ACD servers
  • 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 needed

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sipXcom Managed Devices

Any SIP compatible phone works with sipXecs 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 sipXecssipXcom:

  • Polycom SoundPoint all models (IP 301, 320, 330, 430, 450, 501, 550, 560, 601, 650, 670)
  • Polycom SoundStation IP 4000, 6000, 7000 SIP
  • Polycom VXX-1500 video phone (release 4.0.2)
  • 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

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sipXcom Managed Devices (experimental support)

Experimental support 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 tested. Some functionality might not be implemented or supported.

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

Installation and Upgrades

  • Automated installation from CD ISO for OS and sipXecs sipXcom IP PBX application
  • Graphical configuration wizard for system configuration after installation
  • 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

Centrally Managed

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sipXcom Distributed System (cluster)

  • Automated installation and configuration of a distributed system with specific server roles
  • Automated and central configuration of a high-availability redundant sipXecs sipXcom system
  • Allows for dedicated server hardware for conferencing, voicemail, ACD Call Center, and Call Control
  • All configuration for remote servers is centrally generated and distributed securely

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This is probably quite an incomplete list. In any case, sipXecs sipXcom IP PBX is fully SIP standards compliant.

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