SipXecs Features
From SIPfoundry sipx, The Open Source SIP PBX for Linux - Calivia
sipXecs System Supported Feature List
Download the sipXecs 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 services * Call Center (ACD) server * Call Park Server * Presence Server (Broadsoft and IETF compliant resource list server) * Group Paging Server * Call Detail Record Collection & Processing Server * Management and Configuration Server * Process Management Server (watchdog)
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-dial (Windows XP) * 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 * ACD presence and supervision capabilities * Individual phone management
Superior Voice Quality
* Peer-to-peer media routing for best quality (media not routed through the sipX server) * Unmatched voice quality with lowest delay and jitter * Support for any codec supported by the phone or gateway (including video) * Support for Polycom HD Voice * Codec negotiation (no transcoding required)
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 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 in ITAD numbers. See freenum.org
* Redirector plugins - any imaginable dial rule can be added as a plugin
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
* 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 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
* Unlimited number of simultaneous calls * 54,000 BHCC, 120,000 BHCC two-way redundant (depends on server HW) * Up to 10,000 users per dual-server HA system * 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
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 configred manually. The following devices are plug & play managed: * Polycom SoundPoint all models (IP 301, 320, 330, 430, 501, 550, 601, 650) * Polycom SoundStation IP 4000 SIP * Snom 300, 320, 360 * Grandstream BudgeTone, HandyTone * Grandstream GXP2000 * Grandstream GXV3000 Video Phone * Hitachi IP3000 and IP5000 WiFi phones * Cisco ATA 186/188, 7960, 7940, 7912, 7905 * ClearOne MaxIP Conference Phone * LG-Nortel LG 6804, 6812, 6830 * Nortel video phone 1535 * Audiocodes gateways (FXO/PRI and FXS) * IPDialog SIPTone V * Linksys
Required Hardware
* Intel compatible server (VIA C3/C7, Pentium III, Pentium 4, Core 2 Duo, AMD, Xeon) * Min RAM 512MB, 2 GB to 4 GB preferred * Linux operating system (RHEL or Fedora preferred) * No special HW required
Installation and Upgrades
* Standard Linux package management (e.g. up2date and yum) * Single install CD that installs the Linux OS and the sipX application * Graphical configuration wizard for system configuration after installation * Automated installation and configuration of a high-availability redundant system * Optional auto-configuration of DNS, DHCP, NTP, FTP, TFTP, HTTP servers * Designed so that no Linux admin skills are required for installation and configuration * Automated upgrades using standard Linux package management (e.g. yum update)
SIP Implementation
* 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
