Commend SIP Series
Introduction
2.0/0612
5
Overview of Features
Very high volume
Full Duplex for natural, hands-free communication
Display support NEW!
Full keypad support NEW!
Handset support NEW!
Local directory support NEW!
Chain call support (e.g. automatic processing of call sequences) NEW!
STUN support NEW!
SNMP for surveillance of the station
Using Pre-recorded audio as:
Waiting information at call initiation
Individual call tone for call initiation
Location message
Acoustic indication at line fault
Control of the 2 relays e.g. as door opener via
DTMF post-dial or
Web – or as:
Attendant contacts for various functions, e.g.:
Additional signalisation while ringing, during a call or in case of malfunction
Three inputs for connecting add-on call button modules NEW!
Remote controllable via HTTP; Line-Out (SIP stations)/Line-In (SIP moduls) NEW!
Server redundancy NEW!
Operation without Server possible
Configurable Acoustic Echo Canceller (AEC)
Configurable Background Noise Canceller
Adaptive jitter buffer
Complies with SIP standard for easy integration in every SIP capable PBXes
Integrated webserver for configuration & firmware update
Adjustment of microphone sensitivity and volume
Flexible operation via Power over Ethernet or via external power supply
3.4 kHz speech quality for optimum intelligibility and compatibility
Increased system availability by redundant LAN infrastructure
The integrated data switch function enables the connection of further IP devices, e.g. IP camera
Configurable Auto Answer Function
Instant boot (system boot within seconds)
Communication via IP-data networks – no additional cabling required
Robust construction with protection class IP 65 – vandal resistant versions additionally mechanical
impact resistance up to IK 09
Series F: Dirt-repellent foil surface, resistant to cleaning agents and disinfectants
Configurable backlight
V
O
IP
ACCORDING
SIP S
TANDARD
–
HOW
DOES
IT
WORK
?
Each VoIP subscriber is registered automatically with the respective IP address at a server of the corre-
sponding SIP provider. This provider assigns a new address to the subscriber according to the rules of the
SIP standards, in form of “sip:[email protected]”. This address is allocated to a normal tele-
phone number.
If a subscriber enters this telephone number in order to establish a conversation, it will be translated into
the SIP address at first. In this manner, it is possible to identify the current IP address of the called subscri-
ber. The server is sending this information back to the calling subscriber, whose hardware and software
now is forwarding the audio packets to the IP address of the conversation partner. In order that this con-
versation partner is able to answer, the calling subscriber also forwards his own, current IP address.
