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 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:
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provide quality of voice over internet
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secure calls
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provide IP-TDM gateway capability
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provide softswitch capability
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include integrated H.323 gatekeeper
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include integrated SIP registrar
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include integrated xSIP registrar
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include integrated FTP client to upload records to an FTP server
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manage the traffic efficiently
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provide alternate TDM routes if IP trunk calls fail due to network unavailability
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support numerous system and user features
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be scalable for future expansions
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have open architecture for future implementations
Big savings can be achieved by using the Telesis PX24X Hybrid IP PBX Business Phone System to:
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bridge distant corporate offices
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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.

Various local and remote users of the PX24X, which can be IP or TDM.
System Phones, System and User Services
DTS821 is an executive digital telephone set for the PX24X IP PBX for the comfort of the user. The DTS821 has easy-to-use access menu keys and numerous dedicated keys for various telephony tasks and personnel settings. It has a large backlighted graphic LCD. DTS821 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. Remarkable features are:
- easy-recording (single key press) the conversation bi-directionally (both calling and called party voices)
- access to missed calls list
- access to dialed numbers list
- access to incoming calls list . access to directory
- playing, deleting voice messages and recorded conversations
- setting volume
- setting ringer volume
- selecting ringer melody
- programming function keys
- programming quick access keys
- setting call forward unconditional
- setting call forward busy
- setting call forward no-reply
- setting hotline
- activating call waiting
- activating do not disturb
- activating wake-up (reminder) service
- observing firmware version
- deflecting calls
- activating call back
- holding calls on
- retrieving calls
- transferring calls
- tracing transferred calls (up to 4 calls)
- activating conference

DTS821 Digital Phone in gray color
Two other digital telephones are DTS821S and DTS200S. DTS821S is the digital telephone set offering advanced telephony tasks similar to these of the DTS821 executive set. Users without need of many quick access keys may choose the DTS821S and get the benefits of the Telesis PX24X Hybrid IP PBX Business Phone Systems. DTS200S is a good match for economy and utility. DTS200S offers the comfort of digital voice quality as well as numerous value added services like one-key-touch personal conversation recording.

DTS821S Digital Phone in white color

DTS200S Digital Phone in white color
The Telesis PX24X Hybrid IP PBX Business Phone System provides hundreds system features, system parameters, and user services. If it is desired, these may be activated or deactivated. With these programmable features and services, the PX24X IP PBX make traffic management highly efficient and versatile. The convenience of dealing with all-in-one system software Xymphony allows the user to support as many features as desired, up to the maximum capacity. Another aspect of the PX24X IP PBX system`s software is an advanced algorithm for creating routing tables.
Many of the user services are applicable for SIP calls with using the basic SIP supplementary services such as Invite (call hold), call forward and call transfer (with refer method). Similarly, these are also applicable for H.323 calls with using the basic H.450 supplementary services.
Furthermore, Telesis proprietary xSIP (eXtended SIP) protocol allows value added services of the DTS821 digital telephone sets also to be functional over IP.

ITS821 IP Phone and XPhone IP Softphone (PC edition) have easy-to-use access menu keys and numerous dedicated keys for various telephony tasks within the PX24X Hybrid IP PBX System.

XPhone IP Softphone (Mobile edition) for Pocket PC Phone and Smartphone.
For ensuring the easy and fast programming of the PX24X IP PBX, each programmable parameter is set to a default value. The standard default values are carefully selected and optimized for a wide range of customers` requirements. Thus, the flexibility does not bring a complexity.

19inch subrack option allows PX24X Hybrid IP PBX Systems to be also mounted into industry-standard racks.
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.

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:
- 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
- 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

- 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
- 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 |
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Operational software |
Xymphony PX24X |
|
Maintenance and administration |
Over IP |
|
Operating voltage |
220 VAC,48 VDC |
|
CPU Type |
High speed DSPs |
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Switching Matrix |
512 x 512 |
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Analog subscriber loop impedance |
3,000 ohms |
|
Ethernet interface |
10/100 BaseT |
|
Caller ID ETSI FSK modem |
Yes |
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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.729A audio codec | Yes |
|
G.711 frame length |
10 to 90msec |
|
G.723.1 frame length |
30 to 90msec |
|
G.729, G.729A 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 |
Related Readings
For further readings, see the Technical Documentation section.