Telesis PX24 IP PBX Business Phone System
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Design and manufacture IP PBX Business Phone System, Switching System, VoIP Gateway, and Signaling Converter  

PX24X: Large Capacity Hybrid IP PBX

PX24X in Brief

With a total capacity of 448 TDM ports that allow connections to Basic and Primary Rate ISDN terminal equipment, analog telephone and fax, Telesis digital set, and hundreds of H.323 and  SIP entities such as terminals, gateways, gatekeepers, and registrars for VoIP calls, the Telesis PX24X Hybrid IP PBX is the perfect solution if you are looking into expanding or updating an existing telecommunications network, migrating to the IP Telephony or building a network that is ready for the future. The TDM trunks support a wide range of signaling types, including SS No. 7, V5.2, DSS1 Euro ISDN, QSIG, R1, R2, CL-1B, OCL-1B, TCL-1B, SL/ZSL, SLM, and other CAS signaling. The built-in 10/100 BaseT ethernet interface supports H.323, SIP and Telesis xSIP (eXtended SIP) protocols at the same time. The PX24X is capable of routing TDM calls to VoIP calls and vice versa. 

The Telesis PX24X can be configured as a Hybrid IP PBX, PBX, rural exchange, signaling converter, V5.2 AN/LE or channel bank.

The Telesis PX24X has a modular mechanical structure. The installation of the system or expanding an already operating one by adding subscriber and/or trunk line interfaces is simple and fast.

PX24X Large Capacity Hybrid IP PBX

Telesis PX24X with modular mechanical structure. Easy capacity expansion.

PX24X as a Hybrid IP PBX Business Phone System

The Telesis PX24X Hybrid IP PBX Business Phone System offers significant cost and performance advantages. The obvious advantage of IP Telephony is cost savings. Sending packets of data via an internet network is cheaper than making voice calls over a traditional circuit switched telephone network. The PX24X combines VoIP technology, TDM technology, and Telephony features altogether. Telesis PX24X IP PBX systems are engineered to: 

  • provide quality of voice over internet
  • secure calls
  • provide IP-TDM gateway capability
  • provide softswitch capability
  • include integrated H.323 gatekeeper
  • include integrated SIP registrar
  • include integrated xSIP registrar
  • include integrated FTP client to upload records to an FTP server
  • manage the traffic efficiently
  • provide alternate TDM routes if IP trunk calls fail due to network unavailability
  • support numerous system and user features
  • be scalable for future expansions
  • have open architecture for future implementations 

Big savings can be achieved by using the Telesis PX24X Hybrid IP PBX Business Phone System to:

  • bridge distant corporate offices
  • access long distance call operator rather than paying high traditional call charges

Another advantage of the Telesis PX24X Hybrid IP PBX Business Phone System system is to increase the productivity of employees. Employees would be able to communicate without incurring greater communication costs in the process.

The Telesis PX24X Hybrid IP PBX Business Phone System is equipped with well-known industry standard codecs for high quality communication as well as low bandwidth usage. It is capable of supporting multiple codecs and ready for adding more as technology emerges and popularity changes. Another key technical requirement for the VoIP quality is the echo cancellation. The echo canceler of the Telesis PX24X IP PBX is AT&T certified and exceeding G.168:2002 requirements. It is better than industry standard cancelers under the most important and difficult conditions like double-talk and the presence of background noise.

Considering the security issue, the Telesis PX24X Hybrid IP PBX Business Phone System provides greater protection than traditional phone systems. Furthermore, the ability to encrypt voice via IP call by using 256 bit AES (Advanced Encryption Standard) provides an advantage for security compared to existing telephone systems.

 

 PX24X Hybrid IP PBX Business Phone System and its users schema.

Various local and remote users of the PX24X, which can be IP or TDM.

System Phones

DTS480 is an executive digital telephone set for the PX24X. DTS480 has easy-to-use access menu keys and numerous dedicated keys for various telephony tasks and personnel settings. It has a large color LCD video display.

DTS480 is connected and powered by a single pair of wires. The set handles multiple calls simultaneously, such as receiving a new call while keeping another on hold.

DTS480 Executive Digital Telephone Set image

DTS480 Digital Telephone

A few features of the DTS480 are:

  • Telesis proprietary firmware
  • Automatic firmware upgrade without service interruption
  • 4.3 inch color TFT LCD video display
  • Desktop or wall mount
  • Continuously tiltable stand for comfortable use
  • Full-duplex hands-free operation with advanced echo cancellation
  • Menu-driven easy operation
  • External microphone and headphone jacks
  • 64 keys in total
  • Navigation keys
  • Backlighted 30 keys for quick access (120 quick access with using the Shift key)
  • Busy Display Panel
  • Call park keys
  • Programmable function keys
  • Transfer, Flash, Mute, Conference, Disconnect, Redial etc. function keys
  • Conversation (voice) record key
  • Polyphonic ringer melodies
  • Adjustable volume and ringer levels
  • Missed calls list
  • Dialed numbers list
  • Incoming calls list
  • Directory
  • Call forward unconditional, call forward busy, call forward no-reply, hotline, call waiting, do not disturb, wake-up (reminder) call deflect, call back, call hold, call retrieve,call transfer, tracing transferred calls and many other services
The PX24X provides video video telephony solutions too. Telesis VTS480 is the IP video phone with advanced features for the PX24X.

VTS480 IP Video Phone

VTS480 IP Video Phone

Some of these features are: 

  • Very fast boot up (some seconds only)
  • Telesis proprietary firmware
  • Telesis xSIP (eXtended SIP) protocol
  • Automatic firmware upgrade without service interruption
  • 4.3 inch color TFT LCD video display
  • VGA camera
  • Tiltable (up and down) camera lens direction for best capture
  • Desktop or wall mount
  • Continuously tiltable stand for comfortable use
  • Full-duplex hands-free operation with advanced echo cancellation
  • Menu-driven easy operation
  • Voice call or Video call
  • Start, stop, and restart options for Video during the call
  • Advanced NAT traversal methods for easy operation in LAN and WAN
  • Dual switched 10M/100M auto-sensing Ethernet ports
  • Auto detection of cross-over or straight-through ethernet cables
  • Optional POE
  • Inter-working with other H.264 SIP video phones or video soft phones in the same LAN
  • External microphone and headphone jacks
  • 64 keys in total
  • Navigation keys
  • Backlighted 30 keys for quick access (120 quick access with using the Shift key)
  • Call park keys
  • Programmable function keys
  • Transfer, Flash, Mute, Conference, Disconnect, Redial etc. function keys
  • Conversation (voice) record key
  • Polyphonic ringer melodies
  • Adjustable volume and ringer levels
  • Missed calls list
  • Dialed numbers list
  • Incoming calls list
  • Directory
  • Call forward unconditional, call forward busy, call forward no-reply, hotline, call waiting, do not disturb, wake-up (reminder) call deflect, call back, call hold, call retrieve,call transfer, tracing transferred calls and many other services

Click on Telesis Phones for other digital and IP phones offered for the PX24X.

Some Other Applications with the PX24X

PX24X as a V5.2 Access Network (AN) Equipment

The term AN (access network) refers to the network between the local exchange (or Central Office) and the subscriber. In many countries, this network is still predominantly made up of the copper-cable-based point-to-point connections. However, conventional point-to-point copper cabling has some limitations:

  • It offers limited bandwidth, which is difficult to overcome.
  • Inflexibility: both in time and types of service provided.
  • Due to star topology (from the exchange to the subscriber), reliability is limited.
  • Installation time is long.
  • It is maintenance intensive due to possible cable damage and thus costly.
  • Largely passive, making it difficult to manage.
  • Loop length limitations( ~ 10 km ).
  • Uneconomical in remote, isolated areas with low telephone densities.
  • Prone to electromagnetic interference.

To overcome the above-mentioned issues, several vendors developed AN (access network) technologies, almost all which support the V5.2 protocol in connecting to the voice-switching Local Exchanges.

PX24X as a V5.2 Access Network

Telesis PX24X is an ideal V5.2 Access Network equipment for analog telephone access (with Caller ID support). At its maximum capacity, the PX24X AN can have up to four E1 interfaces with V5.2 protocol. Switching is actively done on the local exchange (LE) side, the number of bearer channels between the LE and the PX24X AN should be planned in accordance with the expected telephone traffic. Erlang loss formula could be used as a service quality target. To illustrate, through a 60-channel (2 x E1) capacity V5.2 interface, total erlang traffic capacity is 44.75, with a lost-call probability of 0.005 at random traffic. Consequently, for a 250-subscriber capacity PX24X AN with a 60-channel V5.2 interface, erlang per subscriber is almost 0.2.

PX24X as a Channel Bank

Telesis PX24X is ideal for highly advanced but cost-effective channel bank solutions. The PX24X is an intelligent channel bank that can interface with E1 ports of any switch, VoIP server, PBX to interconnect FXS, FXO, and E&M. Signaling types and protocols supported by the PX24X`s E1 interfaces are:

Channel-Associated Signaling (CAS)
  • Single-bit E&M emulation
  • Two-bit ITU-T R1
  • Two-bit ITU-T R2
  • Many variations of signaling types (such as CL-1B, OCL-1B, TCL-1B, SL/ZSL, SLM) widely employed in CIS countries
Common-Channel Signaling (CCS)
  • DSS1 (EuroISDN in the TE direction)
  • DSS1 (EuroISDN in the NT direction)
  • ECMA QSIG (in the TE direction)
  • ECMA QSIG (in the NT direction)
  • ITU-T and ETSI Signaling System No.7 ISUP

At its maximum capacity, the PX24X Channel Bank can have up to four E1 interfaces with 120 voice channels to interconnect FXS, FXO, and E&M ports. For various capacity requirements, it may be configured as:

  • four E1s : 120 voice channels
  • three E1s : 90 voice channels
  • two E1s : 60 voice channels
  • one E1 : 30 voice channels

PX24X as an intelligent Channel Bank

Key features of the Telesis PX24X Channel Bank are:
  • Scalability from one to four E1 interfaces
  • Fractional or full E1
  • Clock source could be configured as internal or external
  • Scalability from 30 to 120 voice channels
  • Supports various signaling types and protocols
  • Mixed use of FXS, FXO, and E&M ports
  • Modular architecture allowing custom-configured channel banks to fit the exact requirements
  • Address transmission as Overlap or Enbloc 
  • Internal power supply
Available voice cards are:
  • 16-channel FXS card (with Caller ID)
  • 8-channel FXO card
  • 4-channel 2-wire E&M card
  • 4-channel 4-wire E&M card

PX24X as a Rural Exchange (Rural Switch)

As a rural exchange, the Telesis PX24X serves as a service switching point (SSP). The PX24X is stored-program controlled. SSP and routing functions are integral parts of its operating system. The PX24X rural exchange can be connected to several adjacent SSP and STP switches. 

The PX24X rural exchange also supports many signaling systems of legacy PSTN networks. The PX24X has a call-completion success rate of over 99.999% at 24,000 BHCA (Busy Hour Call Attempts).

Key features of the Telesis PX24X rural exchange are:

  • Scalability from one to four E1 interfaces to connect to adjacent exchanges
  • SSP and STP support
  • Capacity for up to 448 subscribers with long loop drive
  • Caller ID (according to ETSI FSK standards) for analog subscribers
  • Numerous digital and analog signaling protocols
  • Detailed records of originating and terminating calls
  • Traffic measurement
  • Monitoring and analysis of signaling
  • Numerous subscriber services

Technical Specifications

General

 

Operational software

Xymphony PX24X               

Maintenance and administration

Over IP (web and/or SNMP and/or proprietary)

Operating voltage

220 VAC,48 VDC

CPU Type

High speed DSPs             

Switching Matrix

512 x 512

Analog subscriber loop impedance

3,000 ohms

Ethernet interface

10/100 BaseT

Caller ID ETSI FSK modem

Yes

Integrated CMDR buffer

Yes

Integrated DVR (Digital Voice Recorder)

Yes

DVR recording capacity

100 hours
DVR recording channels 4 (up to 44 with licensing)
DVR playing channels 4 (up to 44 with licensing)
Xymphony API server Yes
FTP client Yes

Conference hardware

Yes

DTMF transceivers

Yes

MFR1 transceivers, ITU-T Q.320

Yes

MFCR2 transceivers, ITU-T Q.441

Yes

HDLC transceivers

Yes

ANI transceivers

Yes

Pulse shuttle (R1.5) transceivers

Yes

Real-time charging

Yes
12 or 16kHz charge pulse detectors Yes
Polarity reversal detectors Yes

A-Party analysis

Yes

B-Party analysis

Yes

Subscriber services

Yes

Credited subscribers

Yes

Remote access

Yes

Signaling interworking

Yes

Programmable tones

Yes
Programmable ring melodies Yes
   

Applications

 

Rural switch

Yes
V5.2 Access Network / Local Exchange Yes

Signaling (protocol) converter

Yes

SSP, Service Switching Point for SS7

Yes

STP, Signal Transfer Point for SS7

Yes

PBX

Yes

Hybrid IP PBX

Yes
   

Interfaces (max.)

 

Analog subscribers

448

Digital subscribers

64

Analog DC loop trunks

224

Analog E&M (two- or four-wire)

112

RDTT (Ring Down Tie Trunk, local battery)

224

BRI ISDN S0

4

BRI ISDN Uk0 (Line code 2B1Q)

4

PRI ISDN

4

E1 interfaces (ITU-T G.703)

4

SIP user agents

512

H.323 endpoints

512
Telesis xSIP users 96
   

TDM Signaling

 

Dial-pulse dialing from analog subscribers

Yes

Dial-pulse dialing to analog trunks

Yes

DTMF dialing from analog subscribers

Yes

DTMF dialing to analog trunks

Yes

Caller ID transmission on analog subscribers

Yes

Caller ID detection on analog trunks

Yes
12 or 16kHz charge pulse detection on analog trunks Yes
Polarity reversal detection on analog trunks Yes

Pulsed line/pulsed address on E&M

Yes

Continuous line/DTMF address on E&M

Yes

Single-bit pulsed line signaling types on E1

Yes

Single-bit continuous line signaling on E1

Yes

MFR1 signaling on E&M

Yes

MFCR2 signaling on E&M

Yes

MFR1 signaling on E1

Yes

MFCR2 signaling on E1

Yes

ISDN (Euro ISDN, DSS1), ETSI EN 300 403

Yes

ISDN Supplementary services: 3PTY, AOC, CCBS, CCNR, CFU, CFNR, CLIP, CLIR, COLP, COLR, ECT, DDI, HOLD, MCID, MSN, UUS

Yes

ISDN (QSIG), ECMA-143 PISN

Yes

V5.2 LE protocol, ETSI EN 300 347

Version 2

V5.2 AN protocol, ETSI EN 300 347

Version 2

SS7 ISUP (CCS no.7), ETSI EN 300 356, ITU-T

Yes
   

CIS Countries - Russia

 

Local  trunks SL, Connection line CL

Yes

Toll-connecting trunks ZSL, Ordered connection line OCL

Yes

Toll-switched trunks SLM, Toll connection line TCL

Yes

Two-bit CAS signaling

Yes

Single-bit CAS signaling

Yes

Two-, four-wire analog signaling

Yes

Multifrequency signaling:Pulse packet 1, 2, 3a, 3b

Yes

Multifrequency shuttle signaling:Pulse shuttle, R1.5

Yes

Pulse (decadic) signaling

Yes

ANI request and reception

Yes

ANI response (generation)

Yes

Unilateral call clearing

Yes

Bilateral call clearing

Yes

Calling party category translation

Yes
   

IP Telephony

 

Interface

10/100 BaseT

H.323 protocol, Version 5

Yes
SIP Session Initiation Protocol, RFC 3261 Yes
Telesis xSIP (eXtended SIP) protocol Yes
FTP (File Transfer Protocol) Yes
Adjunct protocol Yes
XDP (Xymphony Discovery Protocol) Yes
Dynamic DNS update client service Yes
G.711 audio codec Yes
G.723.1 (5.3 and 6.4kbps) audio codec Yes
G.729, G.729AB audio codec Yes

G.711 frame length

10 to 90msec

G.723.1 frame length

30 to 90msec

G.729, G.729AB frame length

10 to 90msec

Silence Suppression (VAD)

Yes

Echo Canceler G.168-2002

Yes
QoS (Tos and Diffserv) Yes
T.30 Fax Pass-through for SIP Yes

Integrated H.323 gatekeeper

Yes
Integrated SIP registrar Yes
Integrated xSIP registrar Yes

H.323 endpoints, which can register

512
SIP user agents, which can register 512
xSIP users, which can register 96 (with licensing)
VoIP-TDM gateway channels 48 (with licensing)

Programmable ports / sockets

Yes
MD5 authentication Yes
H.235 Baseline Security Profile Yes
H.235 Baseline Security Profile with integrity Yes
Digest authentication Yes

Audio (voice) encryption

AES-256 (with licensing)

Softswitch capability

Yes

IP to TDM gateway capability

Yes

TDM to IP gateway capability

Yes

H.450 supplementary services

Yes
SIP supplementary services Yes
API (Application Program Interface) Yes
Fidelio Interface Application Specification protocol Yes
Automatic conversation recording for VoIP calls Yes
Hosting Java SIP client applet Yes

Related Readings

We recommend you to visit Telesis Wiki Pages for technical readings.

    
 

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