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Fernprogrammiersoftware für EC II  FLEX Geräte
ZIP XymphonyClients, includes
XTools, XCom, XPhone utilities
a
nd TAPI driver
10 MB

SIP VoIP PRI/E1 gateway and protocol converter

Stillink 800
  • Eight PRI/E1 interfaces
  • Up to 512 VoIP users internal/external
  • Up to 45 TDM/VoIP channels

The Stillink 800 provides cost effective solutions for:

Digital VoIP gateway
Stillink VoIP access gateways are unique in many respects. One is that they support dozens of TDM signaling / protocol types such as SS No. 7, DSS1 Euro ISDN, QSIG, R1, R2, CAS, CIS, as well as VoIP protocols such as H.323 and SIP. Stillink VoIP access gateways may connect to any legacy equipment with any TDM signaling or protocol and provide VoIP connectivity.

Beispiel 1:  VoIP/ISDN Gateway Beispiel 2: VoIP/SS7 Gateway Beispiel 3: VoIP/CAS Gateway
E1 Signaling Converter
Stillink systems give organizations and telecom operators the advantage of being able to add newer generation TDM signaling support to their old TDM network infrastructure more quickly and at lower cost than would be the case if they upgraded their existing core network switches. Stillink systems have fully non-blocking circuit switching technology and can handle very high call volumes without congestion. The all-protocol switching capability contained in Stillink systems provide a bridge to today`s heterogeneous TDM networks. Furthermore, advanced routing capabilities provide easy and flexible E1 signaling conversion management.

Stillink systems provide solutions for:
Beispiel 1: SS7 zu ISDN Konvertierung Beispiel 2: ISDN zu CAS Konvertierung Beispiel 3: SS7 zu CAS Konvertierung

Combined Digital VoIP Gateway and E1 Signaling Converter
The Stillink is a unique all-in-one solution combining both digital VoIP gateway and E1 signaling conversion functions in the same system at the same time.

E1 Switch
One of the specialized functions of the Stillink 800 device is te capability to use it as a E1 switch. This is usefull if:

  • E1 trunks have to be alloctaed on different telephony terminals with E1 interfaces;
  • More telecom E1 provider are to be supported on one PBX without expanding its own resources.
  Kombinierte Funktion als VoIP Gateway und Protokollkonverter  

V5.2 access gateway and converter
Stillink V5.2 - VoIP Access Gateways provide V5.2 Access Network solutions for telecom operators having next generation core networks. With the interworking capability, Stillink systems enable operators to connect legacy V5.2 AN systems to their next generation networks cost-effectively and immediately.
With the interworking capability, Stillink V5.2 Protocol Converters enable an exchange to adapt its existing equipment to the V5.2 protocol. This means Stillink V5.2 Protocol Converters:

  • allow V5.2 AN systems to be connected to LEs that do not support V5.2 but support SS7, ISDN, or CAS.
  • allow AN vendors to facilitate the connection of their V5.2 ANs to existing non-compliant LEs

 

  V5.2 VoIP Gateway  

VoIP PBX and migration solution
In many large installations a large number of analogue and digital telephone terminals are operative. Now many companies have the need to expand or substitude with VoIP based terminals the existing PBX keeping the already installed PRI digital lines to the public network. The Stillink 200 is here an optimal solution and alternative. This system is not only a VoIP gateway but also a complete VoIP telephone switch with up to 512 IP extensions or lines.

The Stillink 200 VoIP PBX includes the paket and the TDM switching capability in one system. Some of the features provided with the Stillink 200 PBX are:

  • Integrated gatekeeper for up to 512 H.323 user
  • Integrated registrar server for up to 512 SIP user
  • Proprietary xSIP protocol for Telesis IP digital phones
  • xAPI and TAPI
  • Hounderts of features and functions
  • IP (max. 512) and PRI/E1 (2 x 30 channels) external lines to the public network
  • Supports many different PRI/E1 access protocols
  • High-end Least Cost Routing system
  • High-end CMDR and reporting system
  • Can also be used to expand existing TDM switches with features like: automated attendant, ACD, voice recording, voice mail etc.
  VoIP TK-Anlage und sanfte Migration  

Standard features

  • 8 PRI/E1 (ITU-T G.703) interfaces
  • 512 SIP user
  • 512 H.323 user
  • 768 V5.2 user
  • Up to max. 45 VoIP-TDM gateway channels (15,30,45 as required)
  • Rotary dial for V5.2 PSTN user
  • DTMF dial for V5.2 PSTN user
  • CLIP for V5.2 user
  • Single-bit pulsed line signalling on E1
  • Single-bit continuous line signalling on E1
  • MFR1 signalling on E1
  • MFCR2 signalling on E1
  • ISDN (Euro ISDN, DSS1), ETSI EN300403
  • ISDN added features: 3PTY, AOC, CCBS, CCNR, CFU, CFNR, CLIP, CLIR, COLP, COLR, ECT, DDI, HOLD, MCID, MSN, UUS
  • ISDN (QSIG), ECMA-143 PISN
  • V5.2 LE protocol, ETSI EN300347 version 2
  • SS7 ISUP (CCS no.7), ETSI EN300356, ITU-T
  • Signalling for CIS countries - Russia
  • Local trunks SL, connection line CL
  • Toll-connecting trunks ZSL, ordered connection line OCL
  • Toll-switched trunks SLM, toll connection line TCL
  • Two-bit CAS signalling
  • Single-bit CAS signalling
  • Single-frequency signalling (1VF)
  • Multifrequency signalling: pulse packet 1, 2, 3a, 3b
  • Multifrequency shuttle signalling: pulse shuttle, R1.5
  • Pulse (IWV) signalling
  • ANI request and receipt
  • ANI answer (generator)
  • Unilateral call clearing
  • Bilateral call clearing
  • Calling party category translation
  • IP telephony
  • Supports H.323 protocol, version 5
  • Supports SIP Session Initiation Protocol, RFC 3261
  • Supports Telesis xSIP (eXtended SIP) protocol
  • Supports FTP (File Transfer Protocol)
  • Supports adjunct protokol (CTI)
  • Supports XDP (Xymphony Discovery Protokol)
  • Supports Dynamic DNS update client service
  • G.711 audio codec
  • G.723.1 (5.3 und 6.4kbps) audio codec
  • G.729, G.729AB audio codec
  • G.711 frame length: 10 to 90ms
  • G.723.1 frame length: 30 to 90ms
  • G.729, G.729AB frame length: 10 to 90ms
  • Silence suppression (VAD)
  • Echo canceler G.168-2002
  • QoS (Tos and Diffserv)
  • T.30 fax pass-through over SIP
  • Integrated H.323 gatekeeper
  • Integrated SIP registrar
  • Programmable ports/sockets
  • MD5 authentification
  • H.235 Baseline security profile
  • H.235 Baseline security profile with intergrit
  • Digest authentification
  • Supports H.450 added features
  • Suppports SIP added features
  • Telesis XApi
  • TAPI 2.1
  • Hosting Java SIP clientapplet

Optional features

  • CAS - VoIP (SIP, H.323) access gateway
  • ISDN - VoIP (SIP, H.323) access gateway
  • SS7 - VoIP (SIP, H.323) access gateway
  • CAS - ISDN protocolconverter
  • ISDN - SS7 protocolconverter
  • CAS - SS7 protocolconverter
  • V5.2 - CAS protocolconverter
  • V5.2 - ISDN protocolconverter
  • V5.2 - SS7 protocolconverter
  • V5.2 - VoIP (SIP, H.323) access gateway
  • AES-256 cryptography
  • Automatic voice recording for VoIP connections

Technical data

  • Operative software: XymphonyX
  • Service and administration over IP: Web and/or SNMP and/or Telesis client
  • CPU: High MIPS Digital Signal Processors
  • Power supply: 90-264 Vac
  • Nominal cosumption: 0,25A at 240VAC
  • TDM switching matrix: 512 x 512
  • Ethernet interface: 10/100 BaseT
  • CLIP ETSI FSK modem
  • Integrated CMDR buffer
  • Integrated DVR (Digital Voice Recorder): 100 hour
  • Conference hardware
  • DTMF transceivers
  • MFR1 transceivers, ITU-T Q.320
  • MFCR2 transceivers, ITU-T Q.441
  • HDLC transceivers
  • ANI transceivers
  • Rotary dial (R1.5) transceivers
  • Programmable tone progress level, fequency and duration
  • Dimensions: 255 x 220 x 77 mm (b x t x h) without optional 19" adapter
  • Weight: 3.25 Kg
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