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SipX Features
SipX Features
Contents
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1 sipX System Supported Feature List o 1.1 Core Calling Features o 1.2 Voice Quality o 1.3 User Management o 1.4 Dial Plan o 1.5 PSTN Trunking o 1.6 SIP Trunking o 1.7 Analog Lines (FXS) o 1.8 Performance o 1.9 High Availability o 1.10 Call Detail Records collection o 1.11 Security o 1.12 System Administration Features o 1.13 Plug & Play Device Management o 1.14 Voicemail Subsystem o 1.15 Auto Attendant Features o 1.16 Hunt Groups o 1.17 Call Park Server o 1.18 Call Center Server (ACD) (not available yet in open source) o 1.19 sipX Managed Devices o 1.20 Required Hardware
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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
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 (including video) * Support for Polycom HD Voice * Codec negotiation (no transcoding required)
User Management
* * * * * * 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 Custom permissions (release 3.6) 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 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
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Dial Plan
* Easy to use GUI based dial plan manipulation * Rules based least cost routing * Automatic gateway redundancy and failover * Specific E911 routing * Permission based rules * Prefix manipulation * Dialplan templating for international dial plans (release 3.6) * Built-in support for U.S., German, Swiss, and Polish local dial plans (release 3.6) (Any other local dial plan can be added as a plugin) * Specify internal extension length * ISN dialing based in ITAD numbers. See freenum.org (release 3.8) * Redirector plugins - any imaginable dial rule can be added as a plugin (release 3.8)
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 (release 3.6) * DNIS * CLIP Management (release 3.6) o User CLIP o Gateway default CLIP o Prefix stripping / appending * Per gateway CLIR (release 3.6) * Automatic Route Selection (ARS) * Least-cost routing (LCR) * Automatic failover if unavailable * Automatic failover if busy * FAX support
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 (release 3.8) * Route header for flexible call routing through an SBC (rel. 3.8) * B2BUA as a low cost option for NAT traversal (release 3.8) (our aim is to support the SIP Forum SIPConnect standard)
Performance
* Unlimited number of simultaneous calls * 54,000 BHCC, 100,000 BHCC redundant * Up to 10,000 users * 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
Security
* * * * * All outbound calls authenticated through Authentication Proxy Secure user password management DoS attack prevention HTTPS secure Web access TLS bassed signaling for SIP trunks (release 3.8)
Voicemail Subsystem
* * * * * * Integrated voicemail system Number of voicemal boxes only limited by disk size Browser based user portal MWI User configurable distribution lists Manage Notifications: 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 Up to 50 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
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Hunt Groups
* Unlimited number of hunt groups * Serial and parallel forking * Configurable ring time
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Required Hardware
* * * * Intel compatible server (Pentium III, Pentium 4, Core 2 Duo, AMD) Min RAM 256MB, 1GB preferred Linux operating system (RHEL or Fedora preferred) No special HW required