The sipXecs IP PBX Feature List
From SIPfoundry sipXecs IP PBX, The Open Source SIP PBX for Linux - Calivia
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sipXecs IP PBX Release 4.0 Supported Feature List
Download the sipXecs IP PBX datasheet in pdf format
System Application Services
* SIP proxy / registrar based call control server, optionally redundant (load sharing) * Media server for voicemail and auto-attendant (IVR) services * Conferencing server * Call center (ACD) server * Call park / Music on Hold (MoH) server * Presence server (Broadsoft and IETF compliant resource list server) * Group paging server * SIP trunking server (media anchoring and B2BUA for SIP trunking & remote worker support) * Call Detail Record (CDR) collection & processing server * Management and configuration server * Process management server for centralized cluster management
Core Calling Features
* Transfer (consultative & blind) * Call coverage * Call hold / retrieve * Consultation hold * Music on Hold for IETF standards compliant phones * Admin or user configurable Busy Lamp Field (BLF) presence and softkeys * Uploadable music file * 3-way conference * Call pickup (global and directed call pickup) * Call park & retrieve * Hunt groups * Intercom with auto-answer (bi-directional) * 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-call * Redial * Call history (dialed, received, missed) * Auto off-hook / ring down * Incoming only * Configuration of individual Speed Dial softkeys * Auto-generation of directory information
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 voicemail * Configuration of unified messaging preferences * Time based find-me / follow-me * Flexible configuration of call forwarding * Personal call history * Personal phone book, speed dial and presence management * Click-to-call * ACD presence and supervision capabilities * Individual phone management * Personal auto-attendant * Dynamic conference management w/ click-to-conference
Superior Voice Quality
* Peer-to-peer media routing for best quality (media not routed through the sipXecs 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
User Management
* 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
* Granular per user permissions
* Call permissions:
o 900 Dialing
o International Dialing
o Long Distance Dialing
o Mobile Dialing
o Local Dialing
o Toll Free Dialing
o Forward Calls External
* System permissions:
o User has voicemail inbox
o User listed in auto-attendant directory
o User can record system prompts
o User has superuser access
o User allowed to change PIN from TUI
o User can use Microsoft Exchange 2007 VM
* Custom permissions as defined by the admin
* 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)
o To local extension, PSTN number, or SIP address
o Based on user or admin defined time schedules
o Parallel or serial ring
o Allows definition of ring time before trying next number
o Allows several forwarding destinations
o 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
* 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
* Specific rule for site-to-site call routing between SIP systems
* Redirector plugins - any imaginable dial rule can be added as a plugin
Internet Calling
* Ability to configure SIP URI based call routing to other domains * Specific SBC selection for call routing * Configuration of native NAT traversal w/ optionally redundant media anchoring if necessary * Media anchoring supports voice and video for any codec
Directory, Softkeys, Speed Dial
* Automated generation of directory information per user or per user group * Crreation and Management of many different directories (per user, per user group, per location, etc.) * Automated provisioning of directory information into user's phones * Allows adding contacts to the directory from a .csv file (Excel) * User configurable speed dial (internal / external numbers, SIP URIs) * Speed dial generated server side and backed up * Auto-provisioning of speed dial to phones * User configuration of Busy Lamp Field (BLF) to monitor presence of other users or phones (e.g. attendant console)
PSTN Trunking
* Unlimited number of PSTN gateways and trunk lines
* Supports any SIP compliant gateway (e.g. Cisco, Audiocodes, Mediatrix, Vegastream, 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
o User CLIP
o Gateway default CLIP
o Prefix stripping / appending
* Per gateway CLIR
* Automatic Route Selection (ARS)
* Least-cost routing (LCR)
* Automatic failover if unavailable
* Automatic failover if busy
* FAX support
* Mixing of PSTN and SIP trunks with least cost routing
SIP Trunking
* 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 * 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)
Integration with Microsoft Server 2000 / 2003 and Exchange 2007
* Synchronization with Microsoft Active Directory
o Using LDAP interface
o On demand or automatically based on a schedule
o Graphical query design combines ease of use with flexibility
o Allows preview of records to be imported
* Dialplan integration with Microsoft Exchange 2007 voicemail server
o Allows mixed environment with groups of users on Exchange or the sipXecs VM server
o Permission based selection of VM server per user or user group
o Automatic dialplan routing to Exchange VM
o Enables sll speech based Exchange 2007 capabilities
* Desktop integration with Windows and Outlook
o Counterpath softphone as an Outlook plugin allows video and click-to-dial
o Enables call recording on the desktop
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, Grandstream and Cisco
Performance
sipXecs IP PBX SIP trunking performance
* Unlimited number of simultaneous calls (voice, HD voice, video) * 54,000 BHCC, 120,000 BHCC two-way redundant (depends on server HW) * Up to 10,000 users per dual-server HA system * 450 simultaneous calls through the SIP trunking gateway require < 20% CPU on dual core system * >500 simultaneous conferencing ports per server * Automatic time distribution of re-registration and subscription events
High Availability
* Optionally fully redundant call control system * Based in DNS SRV (no cluster required) * Load balance under normal operating conditions * Geographic dispersion of redundant systems * Real-time synchronization of state information * 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 * 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
Security
* All outbound calls authenticated * Secure user password management * DoS attack prevention * HTTPS secure Web access * TLS bassed signaling for SIP trunks
System Administration Features
* Browser based configuration and management
* Several admin accounts
* 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
* Integrated backup & restore
* Scheduled backups
* Diagnostics
o Display active registrations
o Display job status
o Status of services
o Snapshot logs for debugging
o Logging (customizable log levels, message log per service)
o Display active calls
* Domain Aliasing
* Support for DNS SRV
* Support for DNS NAPTR based call routing
* Automatic restart after power failure
* 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
* Auto-discovery of phones & gateways on the LAN * Plug & play management of phones * Plug & play management of PSTN gateways * Auto-generation of phone / gateway config profile * Auto-pickup of profile by phone / gateway * Centralized management of all the parameters * Centralized backup and restore of all the configs * Auto-generation of lines by assigning users to devices * Device group management & properties * Firmware upgrade management
Voicemail Subsystem
* Integrated voicemail system
* Voicemail system can be localized per user by installing language packs
* Number of voicemal boxes only limited by disk size
* Browser based user portal for voicemail management
* Message Waiting Indication (MWI)
* User configurable distribution lists
* Unified Messaging:
o Email notification of new voicemail messages
o Forwarding of message as .wav file
o 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
* Between 60 and 120 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
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
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:
o Operator
o Dial by Name
o Repeat Prompt
o Voicemail login
o Disconnect
o Auto-Attendant
o Goto Extension
o 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 * Support for Attendant Consoles * ACD call center agent sign in / out
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
Call Park Server
* Unlimited number of park orbits * Visual indication on the phone of the state of the park orbit using the presence server (BLF) * Music on park * Uploadable music file * Configurable call retrieve code * Configurable call retrieve timeout * Automatic park timeout with configurable time * Configurable park escape key * Allow multiple calls on one orbit
Group Paging Server
* Integrated group paging server * Unlimited number of paging groups * Supports regular SIP phones using auto-answer * Supports dedicated in-ceiling devices (SIP) * Configurable paging prefix
Conferencing Server
* Voice conferencing server that can run on the same sipXecs server or on dedicated hardware * Support for voice conferencing * Each user on the sipxecs system can have a personal conference bridge * Dynamic conference controls from the user's Web portal (user portal) * Support for HD Audio and transcoding if necessary * Support for > 500 ports of conferencing, dependent on hardware * Configurable DTMF keys for conference controls using the TUI * A sipXecs IP PBX system can have more than one conference server if more capacity is needed * All conferencing servers and services centrally managed and configured
Call Center Server (ACD)
* 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:
o Ring all
o Circular
o Linear
o 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:
o Agent statistics
o Call statistics
o 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
sipXecs Managed Devices
Any SIP compatible phone works with sipXecs 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 sipXecs:
* Polycom SoundPoint all models (IP 301, 320, 330, 430, 501, 550, 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 * Nortel 1210, 1220, 1230, 1235
sipXecs 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.
* Aastra 53i, 55i, 57i * Snom 300, 320, 360 * Grandstream BudgeTone, HandyTone * Grandstream GXP2000, GXP1200, GXP2010, GXP2020 * Grandstream GXV3000 Video Phone * Hitachi IP3000 and IP5000 WiFi phones * Cisco ATA 186/188 * Cisco 7960, 7940, 7912, 7905 * Cisco 7911, 7941, 7945, 7961, 7965, 7970, 7975 * ClearOne MaxIP Conference Phone * LG-Nortel LG 6804, 6812, 6830 * Nortel video phone 1535 * Linksys ATA 2102, ATA 3102 * Linksys SPA8000 * Linksys SPA901, SPA921, SPA922, SPA941, SPA942, SPA962 * Nortel 1120 / 1140 SIP
Required Hardware
* 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 uses external gateways
Installation and Upgrades
* Automated installation from CD ISO for OS and sipXecs 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 sipXecs 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 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
SIP Implementation
This is probably quite an incomplete list. In any case, sipXecs IP PBX is fully SIP standards compliant.
* RFC 3261 Session Initiation Protocol using both UDP and TCP transports * Advanced call control using RFCs o RFC 3515 Refer Method o RFC 3891 Referred-By header o RFC 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
