.. index:: features Features ======== System Application Services --------------------------- All the 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. * 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) * 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 SOA Architecture / Business Process Integration using Web Services ------------------------------------------------------------------ * Web Services SOAP interfacefor key administrative functions * Web Services REST interface for user portal functions and third party call control * All components centrally managed using XML RPC * Google Web Toolkit (GWT) Core Calling Features (Telephony Features) ------------------------------------------ * Transfer (consultative & blind) * Call coverage * Call hold / retrieve * Consultation hold * Music on Hold for IETF standards compliant phones * User-specific MoH files * MoH music from an external streaming source * Admin or user configurable Busy Lamp Field (BLF) presence and softkeys * Shared Line Appearance / Bridged Line Appearance (Polycom only) * Uploadable music file * 3-way / 5-way video and voice conference on the phone * 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 - Calls from phones to PSTN numbers, or classes of numbers, can be blocked based on: - The destination of the call; for example, when a user or device cannot initiate an international long distance call. - The source of the call; for example, when a lobby phone can only initiate calls to internal numbers. * 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 E911 Emergency Response ----------------------- * Internal notification using email and SMS Remote Branch office support ---------------------------- * 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) Enterprise Instant Messaging (IM) and Presence ---------------------------------------------- * XMPP based IM and presence server based on Openfire * 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) Personal Assistant IM Bot ------------------------- * My Buddy Personal Assistant feature * Dynamic call control using IM * Dynamic conference management using IM * Unified messaging management using IM * Call history / missed calls * Call initiation using corporate dialplan * Corporate directory look-ups Presence and IM Federation -------------------------- * Server side federation with other public XMPP IM systems * 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 Web Conferencing & Collaboration -------------------------------- * Commercial options available through eZuce'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 management * Click-to-call * Individual phone management * Personal auto-attendant * Management of personal IM account * Personal MoH music upload and preferences Superior Voice Quality ---------------------- * Peer-to-peer media routing for best quality (media not routed through the 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 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 * 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 Dialing 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 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 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 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 * Support for complete contact information and user profile, including avatar * Crreation and Management of many different directories (per user, per user group, per location, etc.) * Upload of contacts from GMail and Outlook * User management of directory information * 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 most SIP compliant gateways (e.g. Audiocodes, Mediatrix, Sangoma, 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. Many other ITSP are compatible, see SIP Trunking section - BT (UK) - AT&T - Bandwidth.com - CBeyond - Bandtel - CallWithUs - Eutelia (Italy) - LES.NET - SIPcall (Switzerland) - Vitality - VOIPUser (UK) - VOIP.MS - Appia * 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 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 capabilities Supported Softclients --------------------- Combined SIP and XMPP clients ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ Jitsi and Counterpath Bria clients can be used with the provisioning server for automated mass deployment of SIP and XMPP account setup. * `Counterpath `_ Bria professional * `Jitsi `_ XMPP (IM only) clients ~~~~~~~~~~~~~~~~~~~~~~ * `Pidgin `_ * `Trillian `_ * `Spark `_ Analog Gateways (FXO and FXS) ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ * Supports any SIP compliant FXO or FXS gateway * Analog fax machines FXS gateways * Analog cordless phone support with FXS gateways * Plug & play management of many analog gateway models Performance ----------- * 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 High Availability ----------------- * Optionally fully redundant call control system * Geo-redundant SIP session manager * Based in DNS SRV (no cluster required) * Load balance under normal operating conditions * Geographic dispersion of redundant systems * Real-time synchronization of state information * Automatic recovery after server failure * 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 * Flexible report generation using Jasper Reports, built-in * Supports redundant call control * Determines and records call type information * Internal / external calls * Calls to specific 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 Security -------- * All outbound calls authenticated * Secure user password management * DoS attack prevention * HTTPS secure Web access * TLS based signaling for SIP trunks * HTTPS secures non-SIP communication between sipX components. * HTTPS secures communications between sipX components and admin and user consoles. * Secure channel for retrieving messages from voicemail repository. * HTTP digest authentication for SIP signaling, as specified in RFC 2617, is used for authentication challenges between SIP endpoints and sipX components. * HTTP digest implementation supports MD5. System Administration Features ------------------------------ * 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 * Login history report (successful and unsuccessful) * Automated testing of network services (DHCP, DNS, NTP, TFTP, FTP, HTTP) for proper configuration Plug & Play Device Management ----------------------------- * Auto-discovery of phones & gateways on the LAN * Auto-registration of Polycom phones simplifies installation * 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 Unified Messaging (Voicemail) ----------------------------- * 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 messages 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 * 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 ------------------------ * 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 3rd 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 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 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 Call 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 presence monitor using presence server * Separate welcome and queue audio * Call termination tone or audio * Configurable answer mode * Agent wrap-up time * 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 sipXcom Managed Devices ----------------------- 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 * Counterpath Bria Professional sipXcom Managed Devices (Community supported) --------------------------------------------- Community supported means that the phone plugin for plug & play management is provided as is. These phone plugins are provided and maintained by community members. Some system functionality might not be implemented or supported. * Aastra 53i, 55i, 57i * Snom 300, 320, 360, 370 up to firmware 7.x * 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 * G-Tec AQ10x, HL20x, VT20x Centrally Managed 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 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 SIP Implementation ------------------ This is probably quite an incomplete list. In any case, sipXcom IP PBX is fully SIP standards compliant. * RFC 3261 Session Initiation Protocol using both UDP and TCP transports * Advanced call control using RFCs - RFC 3515 Refer Method - RFC 3891 Referred-By header - RFC 3892 Replaces header * Provide for consultative and blind transfer and third party call controls - Blind transfer (Unannounced) to a different phone without speaking to the other phone prior to transfer. - Consultative transfer (announced) to a different phone without speaking to the other phone prior to transfer. * 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 * RFC 2617 HTTP Authentication: Basic and Digest Access Authentication * RFC 3327 Path header * RFC 3325 P-Asserted identity * RFC 4235 An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP) * RFC 4662 A Session Initiation Protocol (SIP) Event Notification Extension for Resource Lists * RFC 2327 SDP: Session Description Protocol * RFC 3326 The Reason Header Field for the Session Initiation Protocol (SIP) * 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 * BLA support * RFC 3680: A Session Initiation Protocol (SIP) Event Package for Registrations * RFC 3265: Session Initiation Protocol (SIP)-Specific Event Notification * draft-ietf-sipping-dialog-package-06 * draft-anil-sipping-bla-02 * XMPP Compliance * RFC 3920: XMPP Core * RFC 3921: XMPP IM * XEP-0030: Service Discovery * XEP-0077: In-Band Registration * XEP-0078: Non-SASL Authentication * XEP-0086: Error Condition Mappings * XEP-0073: Basic IM Protocol Suite * XEP-0004: Data Forms * XEP-0045: Multi-User Chat * XEP-0047: In-Band Bytestreams * XEP-0065: SOCKS5 Bytestreams * XEP-0071: XHTML-IM * XEP-0096: File Transfer * XEP-0115: Entity Capabilities * XEP-0004: Data Forms * XEP-0012: Last Activity * XEP-0013: Flexible Offline Message Retrieval * XEP-0030: Service Discovery * XEP-0033: Extended Stanza Addressing * XEP-0045: Multi-User Chat * XEP-0049: Private XML Storage * XEP-0050: Ad-Hoc Commands * XEP-0054: vcard-temp * XEP-0055: Jabber Search * XEP-0059: Result Set Management * XEP-0060: Publish-Subscribe * XEP-0065: SOCKS5 Bytestreams * XEP-0077: In-Band Registration * XEP-0078: Non-SASL Authentication * XEP-0082: Jabber Date and Time Profiles * XEP-0086: Error Condition Mappings * XEP-0090: Entity Time * XEP-0091: Delayed Delivery * XEP-0092: Software Version * XEP-0096: File Transfer * XEP-0106: JID Escaping * XEP-0114: Jabber Component Protocol * XEP-0115: Entity Capabilities * XEP-0124: HTTP Binding * XEP-0128: Service Discovery Extensions * XEP-0138: Stream Compression * XEP-0163: Personal Eventing via Pubsub * XEP-0175: Best Practices for Use of SASL ANONYMOUS