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New
for the download |

ZIP
XymphonyClients, includes
XTools,
XCom, XPhone utilities
and
TAPI driver
10 MB
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SIP
VoIP PRI/E1 gateway and protocol converter
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- Two
PRI/E1 interfaces
- Up
to 512 VoIP users internal/external
- Up
to 45 TDM/VoIP channels
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The
Stillink 200 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.
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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: |
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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. |
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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:
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allow
V5.2 AN systems to be connected to LEs that do not support
V5.2 but support SS7, ISDN, or CAS.
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allow
AN vendors to facilitate the connection of their V5.2 ANs
to existing non-compliant LEs.
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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:
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Integrated gatekeeper
for up to 512 H.323 user
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Integrated
registrar server for up to 512 SIP user
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Proprietary xSIP
protocol for Telesis IP digital phones
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xAPI
and TAPI
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Hounderts
of features and functions
-
IP
(max. 512) and PRI/E1 (2 x 30 channels) external lines
to the public network
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Supports
many different PRI/E1 access protocols
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High-end
Least Cost Routing system
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High-end
CMDR and reporting system
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Can
also be used to expand existing TDM switches with features
like: automated attendant, ACD, voice recording, voice
mail etc.
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Standard
features
- 2
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
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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
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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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