Introduction
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.
Deliver to the utmost potential with SAPEX-The All-in-One IP-PBX server!
The All-in-One Embedded IP-PBX Server
SAPEX is a family of fully integrated
IP-PBXs, with embedded Registrar, Proxy, Presence and Voice Mail
Servers. The otherwise distributed servers, are embedded within the
single server platform.
Registrar Server
When a user comes online, they get registered
with the embedded registrar server. The registrar authenticates the
users, stores their information and maintains real-time status of
devices used by them for communication.
Proxy Server
SAPEX acts as a central proxy server providing
various call control features. It can also register as a client to a SIP
proxy server. Registration with multiple SIP proxies is supported. This
functionality of the IP-PBX allows services offered by the ITSP to be
shared among the registered users of the IP-PBX. Flexibility to register
with multiple SIP service providers, offers redundancy, allowing
switching to the alternate trunk in case the trunk goes down. Calls from
system users can be routed through a specific SIP trunk as per
predefined dialing plans.
Presence Server
The Integrated presence server maintains and
distributes presence information of users registered with the IP-PBX.
Presence status helps to determine a user’s availability and preferred
mode of communication, even before a communication session is initiated.
Presence notifies that a friend is available to talk and then uses SIP
messages to negotiate the means of communication and establish the
actual communication session. Further, Instant Messaging (IM) is a
popular mode of real-time communication. The capacity to maintain the
status of a user at all instant enables IM sessions to be established
between the SIP extensions, which leverage presence and instant
messaging.
Knowing a user’s status, it is possible to
reach a right contact, in right time and on the right terminal.
Voice Mail Server
The built-in Voice Mail functionality ensures
that a user does not miss upon important calls, by forwarding the calls
to his Voice Mail box as and when required.
- Configurable Voice Mail Size
The Voice Mail size can be
configured individually for all the 500 users. A default 1 GB USB stick
supports up to 18 hours of recording. The size of Voice Mail can be
increased, replacing the USB stick with one having a higher storage.
- Voice Mail Retrieval
Voice Mails can be retrieved by
calling to Voice Mail server using individual access codes. A Voice Mail
can be stamped for its date and time of arrival and can be notified to
the user at the time of retrieval.
- Event Notifications
The server can be programmed to
delete the voice mails dumped in a user’s mail box, at a scheduled time.
A user can also receive a notification/indication for any new mail
arriving in his mailbox or when a mail is either retrieved or deleted
from the mailbox. In this case the SIP terminal should leverage the
same.
- Email Notifications
The integrated SMTP client
functionality, enables notifying the user for any new mail as well as
the capacity status of the mailbox via an Email. The Voice Mail can also
be delivered as an attachment to the user’s Email-ID.
Back-to-Back User Agent (B2BUA)
SAPEX works as a Back-to-Back User Agent (B2BUA).
Basic telephony services like Call Forwarding, Call Transfer
necessitates call management and tracking for entire call duration. A
SIP server with B2BUA becomes an active participant in a SIP call,
enabling many advanced services in addition to these basic telephony
services. A B2BUA-enabled SIP server maintains the call state for the
entire call duration, enabling support for real-time call monitoring,
controlling and accounting services.
DSP Based VoIP Call Processing
SAPEX comes in two variants. SAPEX SDM and SAPEX
DDM with single and dual DSP support. The DSP based VoIP call processing
ensures maximum throughput with standard voice codecs such as G.711
A-Law, µ-Law, G.729AB, G.723.1L, G.723.1H, GSM-FR, GSM-EFR, iLBC-20 and
iLBC-30. Whether it is a high-bandwidth codec such as G.711 or a LBR
(high compression) codec such as G.729, number of simultaneous
transcoded calls vary minimally between the range of 25 to 30 (SAPEX
DDM).
Extended Connectivity
Unlike, a traditional telephony system, SAPEX does
not bind a user to a fixed location. Instead of a phone number, an
IP-PBX user is identified by his SIP URI. In IP telephony, a VoIP call
is established over the IP network. An IP user is free to carry his
extension anywhere in the world. Wherever a user is, provided with an
adequate Internet link, users can establish calls retaining the same
contact credentials (user name, password). NAT and STUN support enables
the VoIP call to be established, when the communicating devices are
hidden behind NAT/firewalls. Dynamic DNS (DDNS) support maintains SAPEX
connectivity with remote users even with a dynamic addressing scheme.
Feature Transparency
SAPEX, with B2BUA functionality and voice
transcoding support, provides DTMF access to standard telephony
features. Features such as Call Forward, Call Hold, Call Toggle, Call
Park, Call Pickup, Call Transfer, 3-Party Conferencing, Do-Not-Disturb
(DND) and others, can be accessed from a SIP terminal, with the same
ease as in case of traditional telephony.
Localization
SAPEX can be configured as per the telecom
standards of the country where it is installed. It comes along with
built-in web server functionality called "Jeeves" which supports
English, French, Spanish, German, Portuguese and Italian language. Time
Zones, Day Light Saving, Call Progress Tones can be set specific to a
country’s telecom requirements.
Open-Standard
SAPEX supports open-standard SIP protocol to
establish calls over the IP network. Devices such as IP phones, ATAs and
Gateways, with support for SIP, can be easily registered as clients to
the IP-PBX. Matrix offers a range of SIP supported VoIP products
including ATAs, VoIP to FXS Gateway, VoIP to GSM, FXO and FXS Gateway
and IP Phones to fulfill varied communication requirements. Following
the route of open-standard, SAPEX is ensured for full interoperability
with an entity with SIP support. Thus, moving towards the next
generation IP telephony with SAPEX, an organization can also integrate
to existing telephony network through the gateway devices and thereby
plan for a smooth migration to a full IP infrastructure.
Simplified Configuration and Management
SAPEX accounts for a highly simplified
communication architecture, as it is possible to route calls also over
the IP network, besides data. With the embedded web server, an
administrator can remotely program, configure the system using the
multi-lingual GUI. A SAPEX extension can be located anywhere on the
global IP network. An administrator can monitor and manage user
registration and feature access in real-time. A new user can be added,
granted calling rights, defined under a user group, granted voice mail
access with a few clicks from the intuitive GUI itself. An Administrator
can easily monitor the network and SIP trunk status from the GUI
itself. SAPEX also supports debugs at various levels, over IP network,
via the syslog client.
Voice Transcoding
With Matrix IP-PBX it is possible to establish
communication between SIP devices with diverse audio specifications.
This is established with the voice transcoding functionality, which
allows for negotiation of audio codecs between the SIP devices engaged
in the SIP call. Being B2BUA-enabled, SAPEX gains the power to negotiate
and bind the communicating devices to common terms for a successful
call establishment, thereby minimizing the ratio of dropped calls. The
administrator can set preference of vocoders and ensure optimal
bandwidth utilization. The transcoding feature can be selectively
enabled for a set of users. If transcoding is not required, turning off
the transcoding feature, system can support up to 44 concurrent calls.
An administrator can thereby make a tradeoff between the number of
transcoded calls and the maximum calls to be supported by the system.
Centralized IP server for Distributed Usersf
SAPEX proves to be a complete solution for
inter-branch office communications. Dispersed branches can be tied
together over the IP network, with SAPEX located at central branch
office.
Low-tariff internet telephony between
geographically spread locations helps to reduce the communication cost
to a great extent. SAPEX supports peer-to-peer calls between locations
without going through a proxy server. The multi-site connectivity over
IP also facilitates usage of common dial plans and numbering across the
geographically distant branches. The seamlessly connected branches can
share a common auto-attendant and voice mail system also.
A remote user’s connectivity is maintained even
when behind a NAT or firewall. A SIP user located over a public/private
network can be registered easily with SAPEX without the need to use VPN
or other tunneling technologies. The embedded Dynamic DNS client ensures
a remote user can register to the IP-PBX configured for a dynamic IP
without much trouble.
In the world of IP, an end user terminal can be
an IP phone, a soft phone, mobile with SIP client, PDA, etc. A user can
have multiple contact points mapped to a common user identity. A user
can be reached on any of the active terminal at a given instant, with
the flexibility to register from any geographic location on the IP
network.
Presence further determines the availability of a
user (such as online, offline), his willingness to participate in a
communication session (busy, available on phone, out of office and
others) and his preferred mode of communication (call or instant
messaging), before an actual conversation begins. A user has more
choices with him now. A user now has a right to alter his presence
status at his will and intimate the same to others, instantly, through a
Presence and IM client.
The Dynamic Contact Centre
Workplaces such as a contact centre, receive heavy
call traffic on a daily basis. The way a contact centre functions
directly mirrors an organization’s professional image. Efficient
handling of calls, routing calls to the right-skilled person, minimizing
the wait-time for customers and round-the-clock availability are direct
measures of the contact centre efficiency and productivity.
SAPEX can be employed to support up to 500
distributed agents. The IP-PBX can register to as many as 10 SIP
accounts, providing least cost routing through a dial plan set up. The
Built-in Auto-Attendant provides users with self-service option and
forwards the calls either to a specific extension, to the operator or
else to the voice mail. With the ingenious call routing an incoming call
can be routed to predefined user groups, Time Tables delineate the way
calls are routed as per the day timing. The IP-PBX also supports call
routing based on the CLI of an incoming call. SAPEX supports various
telephony features such as Call Forward, Call Hold, Call Pickup, Call
Park, Call Transfer and 3-Pary Conferencing.
SAPEX user can be located anywhere on the public
network. Thus an agent is no longer tied to his desktop location. A
User has the option to either use an IP phone, a softphone or his analog
extension connected with an ATA. With Presence and IM leveraging
extensions, knowing the availability of the co-workers and the most
accurate way to contact them, can completely transform the way-of-work
in contact, sales and support centre. An agent can remotely retrieve his
voice mails with his access code. A roaming agent can be notified for
the arrival of new voice mails and mail box status with an Email
notification. The embedded RADIUS client can efficiently store the call
detail logs to a remote server, for post review.
The Matrix SETU VP range of IP phones is
best-suited for such environment. The intuitive IP phone offers rich
user experience and increases productivity. Full-duplex Speaker Phone,
Headset Interface, Programmable Keys, Auto Answer, Speed Dial and such
useful functionality, make SETU VP the ideal fit. The IP phone is very
easy to install and operate. It can be configured using its built-in web
pages served by the internal HTTP server. Auto Configuration is also
supported to control multiple phones from a centralized server.
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
www.abc.com, then SAPEX adds the appropriate access code "*777"
specified by the ITSP and dials out the number "*777223344" instead of
223344.
Auto-Attendant
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.
RADIUS Client
A built-in RADIUS client facilitates efficient
call logging to a remote server. SAPEX logs the details of calls in CDR
(Call Detail Records) files. These CDR files contain the essential
information to monitor and account a user for the services utilized.
These CDR files are therefore requiring a safe and longer storage. Any
storage internal to a system gets overwritten in case the system memory
fills to its maximum capacity (500 calls). If a user needs to refer
older call details, user will have to take CDR print out, very often.
In such cases, the embedded RADIUS (Remote
Authentication Dial-In User Service) client enables the IP-PBX to send
these CDR files to a remote server called RADIUS/Database server.
Further integration with a billing server, can benefit the service
providers in accurate billing of the clients and thereby offer advanced
value-added services to their subscribers.
SMTP Client
System supports SMTP client which enables it to
send the voice mail and call logs as attachment to a predefined Email
-ID.
Time Table
SAPEX offers flexibility to divide a day into
time zones defined as working hours, non-working hours and break-hours.
As per the time zone, incoming calls can be routed to different
extensions. Such efficient call routing delivers a sound and effective
communication setup, boosting the overall productivity.
User Groups
Multiple extensions can be grouped under a User
Group. This facilitates call reception between pre-defined users. On
reception of a call, the extensions will ring according to the assigned
priorities. Thus ensuring no call remain unanswered. Maximum of 16 user
groups can be defined, with 8 members in each group.
VoIP Silence Disconnect Timer
The VoIP Silence Disconnect Timer parameter
defines the time limit after which a call is to be disconnected, if no
voice packets are received. This leads to better utilization of
available bandwidth.
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 |
Management Features
Web Based Configuration/Firmware Management |
Multi-Lingual Web Based Programming Tool (Jeeves) |
Network, User, SIP Trunk and Voice Mail Status Display (Jeeves) |
Status LED indication for SIP trunk status |
Region based default setting of Language, Time Zone, DST, CPTG |
Syslog Client |
PCAP Trace |
Busy Lamp Field (BLF) |
Call Detail Records (CDR) |
RADIUS Client |
User Class of Service (CoS) |
Soft Restart |
Voice Mail
Configurable mailbox size |
Individual voice mail for each user with Access Codes |
Customizable greetings |
Voice Mail to Email |
Auto-Attendant
Call Routing to operator, system user or voice mail server |
Music-on-Hold (MoH) |
Customizable greetings and voice prompts |
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Voice Functionalities
Country specific call progress tones |
SIP QoS and RTP QoS (Diffserv) |
DTMF (Inband, Outband, SIP INFO) |
Voice Transcoding |
Date and Time Settings
Day Light Saving Time adjustment |
SNTP |
Real Time Clock (RTC) |
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VoIP Port Parameters
VoIP Protocols |
SIP v2, SIP over TCP, Symmetric RTP, RTCP, 100rel/PRACK
SIMPLE: Session Initiation Protocol for Instant Messaging and Presence Leveraging Extensions |
Network Protocol |
IPv4,TCP,UDP,SNTP,STUN,ARP,ICMP,PPP,DNS,SMTP, RADIUS |
SIP |
10 SIP Trunks
Outbound Proxy Support |
NAT |
STUN and NAT Keep Alive |
Voice CODECS |
G.711 A-Law, µ-Law, G.729AB, G.723.1L, G.723.1H, GSM-FR, GSM-EFR, iLBC-20, iLBC-30 |
Call Progress Tones |
Dial Tone, Ring Back Tone, Busy Tone, Error Tone |
Fax |
T.38 Relay and Pass-Through |
Quality of Service |
Layer 3 DIFFServ |
Security |
MD5 Authentication for SIP, Password Protected Configuration by Admin and User |
Power Supply
Input |
5VDC@2.0 Amp through External Adaptor |
Power Consumption |
10 W (Maximum) |
Connector |
DC Power Jack |
LED Indications |
LED for Power Status (1)
LED for SIP Trunk Status (1) |
Mechanical
Dimensions (W×H×D) |
230mm×55mm×163mm(9.06"×2.17"×6.42") |
Unit Weight |
0.520 Kg (1.14 lbs) |
Shipping Weight |
1.360 Kg (2.99 lbs) |
Material and Finish |
ABS Plastic |
Installation Mounting |
Wall and Table-Top |
Environmental
Operating Temperature |
-10°C to +50°C (-14°F to +122°F) |
Storage Temperature |
-40°C to +85°C (-40°F to +185°F) |
Operating Humidity |
5-95% RH (Non-Condensing) |
Storage Humidity |
0-95% RH (Non-Condensing) at 40°C |
Matrix SETU Range of VoIP Products
ETERNITY IP-PBX |
The IP-PBX with Universal Connectivity and Seamless Mobility |
SAPEX |
All-in-One Embedded IP-PBX Server |
VYOM CCX |
High-Density SIP Gateway |
ETERNITY |
The Universal Telephony Gateway |
SETU VGFX |
Multi-port SIP based VoIP to GSM-FXO-FXS Gateway |
SETU VFXTH |
Multi-Port VoIP to FXO-FXS Gateway |
SETU VFX |
SIP based VoIP Gateway with 4/8 FXS Ports, 1 FXO (PSTN Pass-Through) and 1 Ethernet Port |
SETU ATA211G |
SIP based Analog Telephone Adaptor with 1 FXS, 1 GSM and 2 Ethernet Ports |
SETU ATA211 |
SIP based Analog Telephone Adaptor with 1 FXO, 1 FXS Port and 2 Ethernet Ports |
SETU ATA2S |
SIP based Analog Telephone Adaptor with 2 FXS Ports and 2 Ethernet Ports |
SETU ATA1S |
SIP based Analog Telephone Adaptor with 1 FXS Port and 2 Ethernet Ports |
SETU VP248PE |
Executive IP-Phone with 6 Lines x 24 Characters LCD Display and PoE |
SETU VP248SE |
Executive IP-Phone with 2 Lines x 24 Characters LCD Display and PoE |
SETU VP248P |
Executive IP-Phone with 6 Lines x 24 Characters LCD Display |
SETU VP248S |
Executive IP-Phone with 2 Lines x 24 Characters LCD Display |
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Contact Person at office : Arun Sharma |
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Addreass :10-11, Dwarka Puri,
Jamnalal Bajaj Marg, C-Scheme,
JAIPUR
302 001
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Contact: |
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2363657, 2369893 |
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Fax: |
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141-2369893 |
Mob.: |
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9414255709, 9214308421 |
e-mail: |
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radhika.electro@gmail.com |
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