Key Features
Featured List

Integrating or migrating to new-age IP telephony is a much crucial decision, especially for small and medium enterprises. These organizations need to be more agile and dynamic with limited resources. The right communication solution should not balance features for affordability. Matrix SAPEX is a family of pure IP-PBXs, engineered to bring IP telephony to the SMB and SME premises. The embedded platform integrates a SIP Proxy, Registrar and Presence server in a compact hardware platform. SAPEX users can place voice and video calls over the IP network. Besides centralized call control and media relay services, auto-attendant, voice mail, all PBX telephony features and presence based services are available to SAPEX users. Built-in RADIUS and SMTP client allow advanced services to be integrated.

The multi-functional IP-PBX delivers high performance. Simplified management, reduced communication cost, seamless connectivity with remote users and between geographically dispersed branches, advanced communication means and enhanced productivity are apparent benefits. The system employs open-standard SIP protocol and is hence interoperable with SIP proxies, gateways and IP phones. Communication of small and mid-sized enterprises as well as geographically distributed offices, remote workers and contact centre is much simplified and enhanced with SAPEX.

Key Features

Allowed and Denied Numbers

SAPEX offers flexibility to allow and deny dialing of particular number or a set of numbers. The denied list restricts a user from dialing a number programmed in the denied list.

Automatic Number Translation

SAPEX supports multiple SIP Accounts. These Accounts can be availed from Single or multiple ITSPs. While placing a call, a caller is not conscious of the routing logics defined and the SIP account in use. SAPEX itself modifies the dialed number or part thereof so that it matches with the numbering plan that is understood by the ITSP. For example, if a user has dialed the number 223344 to call, then SAPEX adds the appropriate access code "*777" specified by the ITSP and dials out the number "*777223344" instead of 223344.


The Auto-Attendant informs a caller of the way to reach his ultimate destination. Customized welcome and guiding prompts as per time of the day and Music-on-Hold can be played to a caller. With the help of Automated Attendant, a caller can find-his-way to either reach to a desired extension, retrieve information or to leave back a message for the concerned user in his mail box.

Busy Lamp Field (BLF)

A Busy Lamp field is an array of line status lamps. An extension user can view the status of other extensions e.g. busy, ringing or idle, if a user’s Class of Service (CoS) is provisioned for it. The busy lamp indication forms the umpire’s verdict on an extension status, for the operator to either transfer a call or else pick up the call himself.

Call Forking

IP based communications offer wider terminal options such as an IP phone, a softphone, mobile with SIP client or a PDA. SIP provides a mechanism called Uniform Resource Identifier (URI), mapping a user’s identity to multiple devices. Up to three such terminals can be programmed for a single user. So, when a call is initiated, the same is attempted to all (3) user terminals in parallel, known as call forking. A user now experiences extended connectivity, no matter whether he uses an office IP phone or his cell phone (with SIP client) while on tour or a soft phone to communicate. This also eliminates the need to keep a track of users multiple contact addresses.

Call Progress Tones

SAPEX IP-PBX provides users, the flexibility to match the Call Progress Tones and Ring Cadences to the standard ones used in a country. Country Specific Call Progress Tones like Dial Tone, Ring Back Tone, Busy Tone, Error Tone and others can be programmed.

Caller-ID Based Routing

Based on the Caller-ID details, an incoming call can be routed to a pre-defined extension. For example, calls important to business may be directed to the higher authorities, calls with specific CLI may be directed to specific departments, while calls from anonymous numbers may be directed to the customer support teams.

Direct-Dialing-In (DDI Routing)

A call landing on SIP trunk can be directly routed to an extension as per the DDI numbering. The DDI facility should be activated on the SIP trunk by the SIP service provider. Unlike traditional telephony services, IP telephony does not bind a number to its geographical location. Here, calls are placed over internet and numbers are mapped to IP addresses, which may be anywhere on the internet. An IP extension can always be called irrespective of its current location.

Dial Plan

SAPEX supports multiple SIP trunk registrations. Registration with maximum of 10 SIP servers is supported. Calling rates differ on the basis of area of call, service provider, call time, etc. A Dial Plan allows a user to place a call through the most cost-effective SIP trunk, as per a defined call routing logic. Each user can be assigned multiple Dial Plans, either of which gets activated for a specified timing. The Dial Plans may be same for all users or may differ individually.

Do-Not-Disturb (DND)

This feature is useful when a user does not want to receive any incoming calls without logging off from the IP-PBX or switching off the phone in use.

Daylight Saving Time Adjustment

The Real Time Clock (RTC) of the IP-PBX adjusts automatically to be in tune with the Day Light Saving requirements of the country where it is installed.

Dynamic DNS (DDNS)

Matrix SAPEX Supports Dynamic DNS Client which automates the discovery and registration of its IP addresses on the public network. The remote administrator and the IP clients can thereby connect to the IP-PBX using Domain Name associated with the dynamic IP, preventing reconfiguring of systems, whenever a network infrastructure changes.

FAX Homing

Fax Homing allows a user to utilize a common SIP Trunk for both-voice calls and for receiving a fax. With FAX Homing enabled on a SIP trunk, an auto-attendant can be employed to answer incoming calls. Once a fax tone gets detected, call can be directly routed to an extension where fax machine is connected. This obviates any kind of operator intervention. Such optimal usage of a common SIP trunk for both-voice and fax adds to the cost benefit and saves time.

Instant Messaging (IM)

Instant Messaging is a much popular tool of communication. Ability to communicate via text messages, adds an additional and easy means to communicate with colleagues. Further, with most IM clients, it is possible to alter one’s availability status (Online, Offline, Busy, etc) and intimate the same to others, instantly. SAPEX identifies the users as Presentities (whose status is to be viewed) and Watchers (one who needs to know the status of another user). A Watcher SUBSCRIBES (requests) the presence server for the status of presentity. If the presenter has PUBLISHED (intimated) his status, the watcher can be NOTIFIED (informed) about the status of presentity. An administrator can grant certain users the right to not PUBLISH their status, yet avail the presence and IM functionality.

NAT and STUN Support

NAT allows multiple devices in a LAN to share a single public IP addresses and automatically creates a firewall between the internal network and the internet.

The STUN client allows an IP terminal located behind a NAT to obtain the mapped (public) IP address and port number, allocated for connections to a remote host. The users can thereby easily register to the IP-PBX, hidden behind the NAT router/firewall. The STUN support is critical to establish a VoIP call between SIP users, located behind different type of NATs.

Peer-to-Peer Calling

SAPEX supports VoIP calls between two locations without going through a proxy server. Fixed IP addresses of the locations can be programmed in its Peer-to-Peer table. 500 such entries can be programmed for SAPEX. Short, numeric dialing codes can be defined for calls between these locations. Since the Peer-to-Peer calls are placed over the public IP network, the call cost is minimal.
Featured List

SIP Server

Embedded Registrar, Proxy, Presence Servers Supports SIP v2
Back-to-Back User Agent (B2BUA) Registration of multiple SIP Trunks
Embedded Dynamic DNS (DDNS) Client Support for 500 user registrations
Embedded RADIUS and SMTP Client NAT and STUN support

Calling and Routing Features

Access Codes Conference (3-Party)
Allowed and Denied Numbers Configurable Time Zones for Call Routing
Automatic Number Translation Caller-ID Based Routing
Anonymous Call Rejection Do-Not-Disturb (DND)
Caller Line Identification and Restriction (CLIR) Direct-Dialing-In (DDI Routing)
Call Forward Dial Plan (Multiple)
Call Forking Emergency Number Dialing
Call Hold FAX Homing
Call Pickup (Group and Selective) Peer-to-Peer Calling
Call Park Selective SIP Trunk Access
Call Release Timer Time Table
Call Transfer (Blind and Attended) User Group