Sip Series P

PRODUCT MANUAL
ENGLISH
Commend SIP Series

VERSION 2.0/0612
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Commend SIP Series
Manufacturer’s Reference:
This equipment fulfils the requirements of the EU standard 89/336/EEC
(EN 55022, EN 55024).
Therefore this equipment is CE-labelled.

Please keep this description in safe custody!
Attention:
Mounting and installation of the SIP devices and of the equipment may be carried out by authorised ser-
vice personnel only.
Modules may be exchanged only with voltage switched off.
Before exchanging the modules, ESD precautions have to be observed.
Commend SIP Series

Version: 2.0/0612
Number of pages: 69
Errors and omissions excepted.
2 2.0/0612
Commend SIP Series
Content
Introduction 4 Configuration via Web Interface 34
Commend Security for the World of SIP 4 1st Connection 34
Technical Data 6 Configuration of the SIP Station 35
Extent of Supply 9 SNMP 58
System Requirements / Compatibility 10
SIP Series Versions 10
Appendix 62
Basic Knowledge about Audio Configuration 62
Mounting 14 Serverless Operation 64
SIP Series P, SIP Series F 14 Display Menu WS 800V / WS 800F 66
SIP Series V, SIP Series VE 19 Call Initiation at SIP-WS 500F 67
SIP Series M 28
Technical Support 69
Connection 29
SIP Series P, SIP Series V, SIP Series VE, SIP Series F 29
SIP Series M 31
Vers. 2.0/0612 3
Introduction Commend SIP Series
Introduction
Commend Security for the World of SIP

SIP Stations with Intercom Power
The full duplex capable stations of the SIP Series links the world of SIP technology with the reliability
and quality of solutions by Commend,
The stations are connected directly to the Ethernet (LAN/WAN) and in this manner are connected to a
compatible SIP Server via the IP-network. The built-in switch with downlink function allows direct con-
nection of an additional IP-device (e.g. an IP-camera).
Besides high volume, the SIP stations provide a numerous amount of further features: Pre-recorded
audio can be applied in a multipurpose manner, e.g. as acoustic indication at line fault or as waiting
information at call initiation, a configurable background noise canceller provides a crystal clear
communication in challenging situations. Furthermore, the stations are perfectly suited for use as door
stations at entry- and gateways, due to integrated relay outputs.
The robust construction of the SIP Series provides full protection against water, dirt and dust – protec-
tion class IP 65. Each button can be allocated to a call number and the relevant label area can be filled
in individually.
SIP Series P SIP Series V SIP Series VE SIP Series F SIP Series M
Note:
The several versions of the SIP Series are described on page 10.
Speech Connection according SIP Standard

The speech connection is established via Voice over IP (VoIP) according the SIP standard over the con-
nected Ethernet LAN, whether with assistance of a SIP capable PBX, of a SIP provider or via dialling an
IP address directly.
WHAT IS SIP?
The network protocol SIP is only one among many protocols which are used for VoIP; the “Session Ini-
tiation Protocol” establishes the conversation.
This means, SIP is only signalling the conversation. After that, the Session Description Protocol (SDP) ne-
gotiates the conversation modalities: audio codec and transmission protocol. The latter is responsible for
the actual data exchange.
The actual data stream, i.e. the coded speech, is transmitted via the Realtime Transport Protocol (RTP).
This protocol dismantles the audio data into packets and is sending them over UDP – i.e. the User Data-
gram Protocol (UDP) is responsible for the transmission of the data packets.
4 2.0/0612
Commend SIP Series Introduction
VOIP ACCORDING SIP STANDARD – 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.
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
2.0/0612 5
Technical Data
Note:
In the following tables the technical data of all „SIP Series Versions” are listed.
Network, Codecs
DTMF Decoding: RFC 2833
SIP User Agent (UDP): RFC 3261
IP Protocol: IPv6 ready
IPv4, TCP, UDP, HTTP, RTP, RTCP, DHCP, SNMPv2c, STUN
Ports: Web interface for configuration (http): TCP Port 80
SIP: UDP Port 5060
RTP: UDP Port 7078 (incoming)
Optional DNS: UPD and TCP Port 53
Optional SNMP: UDP Ports 160 and 161
Ethernet: 2 x 10/100 MBit/s (Full/Half Duplex)
Codecs: G.711 a-Law
G.711 µ-Law
prepared for G.722
Frequency range 300 – 3,400 Hz
Power Supply
Power consumption: 1,6 W idle
approx. 2 W at conversation (depending on volume)
PoE (Power over Ethernet): Standard IEEE 802.3af
Power consumption of the terminal device:
Class 0 (0.44 W to 12.95 W)
SIP Series P, SIP Series V, SIP Series VE only:

Power supply: 24 VDC ± 2 V, 500 mA or PoE
Attention:
It is mandatory to ensure a correct power supply of the SIP station: min. 22 VDC, max. 26 VDC
SIP Series M only:

Power supply: 12 – 24 VAC or 15 – 35 VDC, 500 mA or PoE
Hardware
RAM Memory: 32 MByte
Flash Memory: 8 MByte
Handset, Headset: EM sensitivity: 14 mVeff
EM impedance: 3.3 kΩ / EM supply: 2.5 V
EP level: 850 mVeff at 0 dBm0 / EP impedance: 200 Ω
Amplifier: Built-in amplifier class “D” with 2.5 W
Outputs: 2 SPDT relay outputs
30 V / 1 A: 100,000 make-and-break cycles
Inputs: 3 inputs for floaing contacts
Relative Humidity: up to 95% not condensing
Connection: pluggable screw terminals
IP Uplink/Downlink:
shielded RJ 45 modular jacks
Cabling: min. Cat. 5
6 2.0/0612
SIP Series P, SIP Series V, SIP Series VE, SIP Series F only:
Operating temperature range: –20° C to +60° C (–4° F to 140° F)
Storage temperature range: –20° C to +60° C (–4° F to 140° F)
Microphone: Omnidirectional electret microphone for
max. 7 m (23 ft) speaking distance
Loudspeaker: Special membrane type for optimal sound quality,
sound pressure: 85 dB/1 W/1 m (3.28 ft), 2 x 8 Ω
Line output: for connection of loudspeaker module
Status indication: red LED
Keypad, call button: SIP Series F, SIP-WS 800x: alphanumeric full keypad, white
backlight activation force: 3 N, 1 x 106 cycles,
SIP WS 20xP, SIP-WS 20xV, SIP Series VE: 1 – 3 direct dialling
buttons
SIP-Serie VE only: large red emergency call button
Display: SIP-WS 800x: Mono-LCD display,
128 x 64 pixel, white backlight
SIP Series M only:

Operating temperature range:: –40° C to +70° C (–40° F to 158° F)
Storage temperature range:: –40° C to +70° C (–40° F to 158° F)
Microphone input: for electret microphone or
dynamic microphone
Mic Input: nominal level 2.8 mV on 3.3 kΩ
(Mic-feeding voltage 2.5 V)
Connection for loudspeaker: 2.5 W at 4 Ω / 1.5 W at 8 Ω
LS-Output: max. 3.5 Veff (LS-Pot max)
for feed-in of audio
(e.g. music, radio conference)
Line-Input: nominal level 0 dBu 0.775 V at 10 kΩ
Status indication: Possibility for connection of a LED 6 mA
Call button: Possibility for connection of 3 single buttons
Housing, Mounting
SIP Series P, SIP Series F:
IP rating: IP 65
Front panel: SIP Series P: Polycarbonate
SIP Series F: Polycarbonate with protective foil
Additional mounting material: Flush mount kit WSFB 50P
Surface mount kit WSSH 50P
Measurements: Mounting with flush mount kit, surface mount kit:
see page 14 to page 18
Weight incl. package: approx. 750 g (1.65 lbs)
Colour: front panel: light-grey (like RAL 7035)
front panel frame: graphite-grey (like RAL 7024)
2.0/0612 7
SIP Series V, SIP Series VE:

IP rating: IP 65
Mechanical impact resistance acc. EN 62262 SIP Series V: IK 09
SIP Series VE: IK 08
SIP-WS 800V: IK 07
Front panel: Stainless steel, 3 mm (0.12 in)
Additional mounting material: Flush mount kit WSFB 50V
Flush mount kit WSFB 50V FL
Surface mount kit WSSH 50V
Rain protection roof WSRR 50V
Measurements: Mounting with flush mount kit, surface mount kit:
see see page 19 to page 27
SIP Series M:
Measurements: SIP-ET 908A: 65 x 130 x 18 mm (2.56 x 5.12 x 0.71 in)
SIP-ET 908A-1: 65 x 130 x 21 mm (2.56 x 5.12 x 0.83 in)
Weight incl. package: ca. 220 g (0.5 lbs)
8 2.0/0612
Extent of Supply
SIP Series
”SIP Series P” ”SIP Series V”: ”SIP Series VE”: ”SIP Series M”:
”SIP Series F”:  SIP station  SIP station  SIP module
 SIP station  Screws for mounting  Screws for mounting  4 fixing spacers with M3
 SIP-WS 800F MD:  Short reference  3 x paste-on label (“SOS”/ thread
Locking block with “EMERGENCY”/“HELP”)  Short reference
screw for emergency call button
 Short reference  Short reference
Additional Mounting Material

The following listed mounting material must be ordered separately:
WSFB 50P: WSSH 50P:
 Flush mount box plastic  Surface mount box (incl. gaskets)
(incl. cavity wall claws & cover lid)  4 x IP 65 cable feed through
 Mounting frame plastic  Wall mount kit
(incl. gaskets and mounting screws) (screws, dowels, gasket rings and washers)
 Cable fixing brackets (incl. screws)  Cable fixing brackets (incl. screws)
 Short reference  Short reference
Weight incl. package: approx. 380 g (0.84 lbs) Weight incl. package: approx. 300 g (0.66 lbs)
WSFB 50V: WSFB 50V FL:

 Flush mount box plastic  Flush mount box plastic with mount frame metal
(incl. cavity wall claws & cover lid) (incl. cover lid)
 Mounting frame plastic  Cable fixing brackets (incl. screws)
(incl. mounting screws)  Short reference
 Flush mount frame metal & gasket
 Cable fixing brackets (incl. screws)
 Short reference
WSSH 50V: WSRR 50V:

 Surface mount box metal  Rain Protection Roof
(incl. 6 x dummy plugs)  Fixing screws
 Plastic box  Short reference
 4 x IP 65 cable feed through
 2 x sabotage plates
(currently not applicable for SIP stations)
 Wall mount kit
(screws, dowels, gasket rings and washers)
 Cable fixing brackets (incl. screws)
 Short reference
Note:
For mounting of the several surface mount kits and surface mount kits see page 14.
2.0/0612 9
System Requirements / Compatibility

 Compatible SIP Server (see ”Compatibility SIP PBX”) or
 G8-VOIPSERV (for additional features) incl. Asterisk Server
Compatibility SIP PBX

Basically the SIP modules can be used with any SIP server. The following server types have been tested
explicitly by Commend and therefore a proper functionality can be confirmed:
Manufacturer* Type Version
Digium Asterisk Version 1.2, 1.4, 1.6
Cisco Call Manager
Cisco Version 5 or higher
Cisco Unified Communication Manager
Nortel CS1000 Version 6
Innovaphone Virtual Appliance IPVA Version 9 final
Alcatel OmniPCX Enterprise OXE
Siemens Hipath 4000 Version 5
3CX 3CX for Windows 3CX PhoneSystem 9
Starface Starface free Version 4.6.5.1
Aastra MX-ONE Version 4.1 SP 1
*The listed products and company names are brand names or registered trademarks of their respective owners.
VoIP Provider Type Version

Sipgate Sipgate.at, Sipgate.de May 2012
Vodafone Arcor vodafone.de January 2011
blueSIP bluesip.net May 2011
dus.net dus.net May 2012
Attention:
Not compatible: Microsoft Lync
SIP Series Versions

The stations of the SIP Series are available in 5 different versions:
 ”SIP Series P”
 ”SIP Series V”
 ”SIP Series VE”
 ”SIP Series F”
 ”SIP Series M”
10 2.0/0612
SIP Series P
SIP-WS 20xP
 SIP station with up to 3 programmable direct dialling buttons (white backlight) in polycarbonate
construction for interior and outdoor areas.
 Each button can be allocated to a call number and the relevant label area can be filled in individually.
 The robust construction provides full protection against water, dirt and dust – protection class IP 65.
SIP-WS 800P
 SIP station with an alphanumeric keypad and function keys with white backlight, in polycarbonate
construction for interior and outdoor areas.
 The station provides a LCD graphic display with white backlight.
SIP-WS 201P: SIP-WS 202P: SIP-WS 203P: SIP-WS 800P:

(1 direct dialling button) (2 direct dialling buttons) (3 direct dialling buttons) (keypad, display)
SIP Series V
SIP-WS 20xV
 Vandal resistant SIP stations with up to 3 programmable direct dialling buttons (white backlight),
stainless steel front panels with 3 mm (0.21 in) thickness, poke protection and special screws.
 Each button can be allocated to a call number and the relevant label area can be filled in individually.
 The robust construction provides full protection against water, dirt and dust – IP rating IP 65 and
mechanical impact resistance IK 09.
SIP-WS 800V
 Vandal resistant SIP station with an alphanumeric keypad and function keys with white backlight,
stainless steel front panels with 3 mm (0.21 in) thickness, poke protection and special screws.
 The station provides a LCD graphic display with white backlight.
2.0/0612 11
SIP-WS 201V: SIP-WS 202V: SIP-WS 203V: SIP-WS 800V:

(1 direct dialling button) (2 direct dialling buttons) (3 direct dialling buttons) (keypad, display)
SIP Series VE
 Vandal resistant SIP emergency stations with up to 2 programmable direct dialling buttons, stainless
steel front panels with 3 mm (0.21 in) thickness, poke protection and special screws.
 The red emergency call button is easily visible from a considerable distance and can be activated
quickly in emergency situations. Each button can be allocated to a call number and the label area of
the direct dialling button (white backlight) of the SIP-WS 212V can be filled in individually.
SIP-WS 211V: SIP-WS 212V:

(1 direct dialling button) (2 direct dialling buttons)
SIP Series F
 SIP station with an alphanumeric keypad and function keys with white backlight, in polycarbonate
construction with a dirt-repellent foil surface (resistant to cleaning agents and disinfectants), for in-
terior and outdoor areas.
 The station SIP-WS 800F additionally provides a LCD graphic display with white backlight. The sta-
tion SIP-WS 800F MD additionally provides an anti-bacterial foil surface.
12 2.0/0612
SIP-WS 500F: SIP-WS 800F: SIP-WS 800F MD:

(without LCD-Display) (with LCD-Display) (with LCD-Display)
SIP Series M
 SIP modules for integration in existing housings and panels or building of customer specific stations.
 Available in 2 different versions: with RJ 45 sockets mounted horizontally or vertically.
 Application examples are for e.g. emergency stations at highways, park ticket machines or also for
smaller systems with door functions.
SIP-ET 908: SIP-ET 908-1:

(RJ 45 sockets mounted (RJ 45 sockets mounted
horizontally) vertically)
2.0/0612 13
Mounting Commend SIP Series
Mounting
SIP Series P, SIP Series F

For installation additional mounting material is required (must be ordered separately).
Following additional mounting material is available for the SIP Series P, SIP Series F:
 Flush mount kit WSFB 50P
 Surface mount kit WSSH 50P
The mounting instruction is split into following sections:

 „Measurements of the Front Panel” of the stations of SIP Series P and SIP Series F
 „Measurements Flush Mount Kit WSFB 50P”
 „Mounting Flush Mount Kit WSFB 50P”
 „Measurements Surface Mount Kit WSSH 50P”
 „Mounting Surface Mount Kit WSSH 50P”
Measurements of the Front Panel

Measurements in mm (inch), not to scale!
Depth: 13 (0.51)
cavity wall mounting: 15 (0.59) (Ö shadow gap between front panel & wall)
165 (6.5)
INNER CONTOURS
VARY DEPENDING ON
VERSION OF STATION
280 (11)
14 2.0/0612
Commend SIP Series Mounting
Measurements Flush Mount Kit WSFB 50P

*Mounting distance
between flush
30 (1.18)*
MR ...... Mounting frame mount boxes
150 (5.91)
front flush with plaster 135.6 (5.34) 30 (1.18)*

MR MR
.3 .5
7
Ø
)
4x
(0
274.5 (10,82)
250.6 (9.87)
265 (10.44)
197 (7.76)
15
(0.59)
1 (0.04) 0.5
(0.02) 82 (3.23)
45.5
(1.79) 159.5 (6.28)
12x Expansion openings (can be broken out) Ø 21

(0.59)
15
(0.32)
8
MR
2.0/0612 15
Mounting Flush Mount Kit WSFB 50P

To corresponding expansion opening of
flush mount box of e.g. WS expansion module
gasket
1
T R MOUNTING
R DIRECTION
gasket
1
* Illustration
front panel
4
4a
5
1 ..... Connection spacer (with gaskets) for connecting two flush mount boxes and
for cable feed through
If possible, then 2 connection spacers shall be used per expansion
(at deppened openings, see above)
2 ..... Flush mount box (plastic)
For cavity wall mounting or installation in a desk, the cavity wall claws have to be used.
(wall thickness: 5 mm to 30 mm / 0.19 in to 1.18 in)
3 ..... Gasket
4 ..... Mounting frame (plastic) 4a ..... Top gasket for mounting frame (mounted ex works)
Screw mounting frame onto the flush mount box - screws in extent of supply
5 ..... Front panel (plastic) with station electronics
Mounting: Press the front panel onto the mounting frame.
Dismounting: Pull off the front panel.
16 2.0/0612
Measurements Surface Mount Kit WSSH 50P

4x Mounting openings Ø 7.5 (0.3) 160 (6.3)

(can be broken out)
275 (10.84)
202.5 (7.97)
197 (7.76)
2 x rear openings
8.5 2 (0.08)
82 (3.23)
(0.33) 46.5 (1.83)
2.0/0612 17
Mounting Surface Mount Kit WSSH 50P

To corresponding expansion opening of 3
surface mount box of e.g. WS expansion module Snap in
gasket
2
R
Insert cable
T
R
Pull back tight
gasket
MOUNTING
DIRECTION
2 * Illustration
front panel
4 4a
5
1 ..... Gasket for expansion opening at the rear of the housing
The expansion openings have to be broken out with a blunt object at the outer notch!
If the rear opening is used, attach the gasket at the backside of the flush mount box (see Measurements).
2 ..... Connection spacer (with gaskets) for connecting two surface mount
boxes and for cable feed through
The expansion openings have to be broken out with a blunt object at the outer notch!
If possible, 2 connection spacers shall be used per expansion.
Make sure the connection spacer is snaped-in correctly in the expansion opening!
3 ..... IP 65 cable feed through
The cable feed through can be put in after breaking out the expansion openings
(installation cable feed through see above).
4 ..... Surface mount box (plastic) 4a .....Top gasket for surface mount box (mounted ex works)
Mounting via wall mounting kit (screws, dowels, gasket rings and washers) – in extent of supply.
Washer and gasket ring have to be mounted inside the surface mount box.
5 ..... Front panel (plastic) with station electronics
Mounting: Press the front panel onto the surface mount box
Dismounting: Pull off the front panel.
18 2.0/0612
SIP Series V, SIP Series VE

For installation additional mounting material is required (must be ordered separately).
Following additional mounting material is available for the SIP Series V and SIP Series VE:
 Flush mount kit WSFB 50V
 Flush mount kit WSFB 50V FL
 Surface mount kit WSSH 50V
 Rain protection roof WSRR 50V
The mounting instruction is split into following sections:

 „Measurements of the Front Panel” of the stations of SIP Series V and SIP Series VE
 „Measurements Flush Mount Kit WSFB 50V”
 „Mounting Flush Mount Kit WSFB 50V”
 „Measurements Flush Mount Kit WSFB 50V FL”
 „Mounting Flush Mount Kit WSFB 50V FL”
 „Measurements Surface Mount Kit WSSH 50V”
 „Mounting Surface Mount Kit WSSH 50V”
 „Measurements Rain Protection Roof WSRR 50V”
 „Mounting Rain Protection Roof WSRR 50V”
Measurements of the Front Panel

160 (6.3)
140 (5.5)
.2 7
Ø
8)
4x
(0
INNER CONTOURS
VARY DEPENDING ON
VERSION OF STATION
255 (10.05)
275 (10.84)
2.0/0612 19
Measurements Flush Mount Kit WSFB 50V

*Mounting distance
MR .... Flush mount frame (metal) between flush
30 (1.18)*
with mounting frame (plastic) mount boxes
150 (5.91)
135.6 (5.34) 30 (1.18)*
front flush with plaster
MR MR
.3 .5
7
Ø
)
4x
(0
250.6 (9.87)
197 (7.76)
265 (10.44)
279 (10.99)
15
(0.59)
1 (0.04) 0.5
(0.02) 82 (3.23)
45.5
(1.79) 164 (6.46)

(0.59)
15
(0.55)
14
MR
20 2.0/0612
Mounting Flush Mount Kit WSFB 50V

To corresponding expansion opening of
flush mount box of e.g. WS expansion module
gasket
1
T R
R MOUNTING
DIRECTION
gasket
1 * Illustration
front panel
3
4
5
1 ..... Connection spacer (with gaskets) for connecting two flush mount boxes and
for cable feed through
If possible, then 2 connection spacers shall be used per expansion (at deppened openings, see above)
2 ..... Flush mount box (plastic)
For cavity wall mounting or installation in a desk, the cavity wall claws have to be used.
(wall thickness: 5 mm to 30 mm / 0.19 in to 1.18 in)
3 ..... Gasket
4 ..... Flush mount frame (metal)
5 ..... Mounting frame (plastic)

Screw mounting frame together with the flush mount frame (metal) and the gasket onto the flush mount box
(screws in extent of supply)
6 ..... Front panel (metal) with station electronics
Screw front panel onto flush mount frame
2.0/0612 21
Measurements Flush Mount Kit WSFB 50V FL

front flush with plaster
279 (10.99)
251.6 (9.88)
22
(8.87)
53 (2.09) 164 (6.46)
136 (5.35)

(0.59)
15
22 2.0/0612
Mounting Flush Mount Kit WSFB 50V FL

...this
...oben,side
Montagerichtung
up, mounting direction
beachten
* Illustration
front panel
1
2
1 ..... Flush mount box (plastic) with Flush mount frame (metal)
Install the flush mount box by moulding into plaster or concrete.
Do not remove plaster cover until the flush mount box is mounted.
2 ..... Front panel (metal) with electronics and gasket
Screw front panel onto flush mount frame
2.0/0612 23
Measurements Surface Mount Kit WSSH 50V

164 (6.46)
148 (5.83)
218 (8.58)
229 (9.02)
279 (10.99)
200 (7.88)
4x Mounting openings Ø 6 (0.24)
4x Ø 5 (0.2)
50 (1.97) 114 (4.49)
60 (2.36)
.8 2
(0 Ø 2
7)
6x
24 2.0/0612
Mounting Surface Mount Kit WSSH 50V

Attention:
 The dummy plugs for the surface mount box (in extent of supply) can not be removed without
opening the housing!
* Illustration
front panel
1 3
2
4 Snap in
1 ..... Surface mount box (Metal)

Mounting via wall mounting kit Insert cable
(screws and dowels) – in extent of supply
2 ..... IP 65 cable feed through
The cable feed through can be put into the plastic box
after breaking out the expansion openings
(installation cable feed through see picture to the right).
3 ..... Plastic box
Pull back tight
Press the plastic box into the surface mount box
4 ..... Front panel (metal) with station electronics
Screw the front panel onto the surface mount box
2.0/0612 25
Measurements Rain Protection Roof WSRR 50V

foam strip, approx. 2 mm (0.08 in)
164 (6.45) 80 (3.15)

277 (10.9)
26 2.0/0612
Mounting Rain Protection Roof WSRR 50V

Note:
The rain protection roof can be used for vandal resistant SIP stations in surface mount version and in
flush mount version.
Following illustration shows the installation of the rain protection roof WSRR 50V to the surface mount
kit WSSH 50V.
Example:
Surface mount kit WSSH 50V
Front panel (metal) with

station electronics
Rain protection roof Fixing screws required for

WSRR 50V WSRR 50V (in extent of supply)
 Remove the screws with which the station front panel is mounted
 Afterwards, screw the rain protection roof with the station front panel onto the surface mount box
or the flush mount frame by using the fixing screws in the extent of supply (see above).
2.0/0612 27
SIP Series M
The modules of the SIP Series M are build-in kits for integration in existing housings and panels or
building of customer specific stations.
Measurements SIP-ET 908 / SIP-ET 908-1

SIP-ET 908A:
18 (0.71)
SIP-ET 908A-1:
21 (0.83)
6.5
(0.26)
15 (0.59)
25 (0.99)
ø5.5 (2x)
(0.22)
35 (1.38)
Potentiometer LS
130 (5.12)
80 (3.15)
60 (2.36)
ø3,2 (4x)
(0.13)
ø4 (4x)
(0.16)
55 (2.17) 5
65 (2.56) (0.2)
28 2.0/0612
Commend SIP Series Connection
Connection
SIP Series P, SIP Series V, SIP Series VE, SIP Series F

Alternative * relates to:
DC– - SIP-WS20xP
power supply
24 V DC ± 2 V DC+ - SIP-WS20xV
- SIP Series VE
OUT 2
OUT 2
OUT 1 Slots for inserting

OUT 1 button labels*
External Loudspeaker
IN 1 MUTE
IN 2 GND
IN 3 LINE– Factory
Reset
GND LINE+ button
EM+
EM–
EP+
EP–
IP Uplink IP Downlink Handset:

+ PoE (e.g. IP-camera) Microphone
Loudspeaker
LAN Connection
“Ethernet LAN” is connected to the RJ 45 socket “IP Uplink” (see connection diagram above).
Power Supply
For connection of power supply, 2 different possibilities are available:
PoE
The SIP station is supplied via “Power over Ethernet” (Standard IEEE 802.3af) via the RJ 45 socket “IP
Uplink” (see connection diagram above).
DC–, DC+
If ”PoE” (“Power over Ethernet”) is not available, the SIP station alternatively can be supplied with
“24 VDC  2 V” via the screw terminals “DC–” and “DC+”.
Attention:
For connection of the power supply, the notes and technical data on page 6 have to be observed!
Note:
If both power supply types (i.e. PoE and supply via “DC–,DC+”) are active at the same time, then the
following has to be considered:
 In case of a breakdown of the power supply via “DC–,DC+”, and the consequentially take over of PoE,
a reboot of the station is initiated.
 In case of a breakdown of PoE, and the consequentially switch-over to power supply via “DC–,DC+”,
the operation of the station will not be interrupted – provided that the 2nd ethernet link (“IP downlink”
see connection diagram above) is connected to the respective LAN network.
2.0/0612 29
Connection Commend SIP Series
Relay Outputs
The terminals OUT1 / OUT1 as well as OUT2 / OUT2 are operating as relay outputs (changeover
contacts, OUT as make-contact, OUT as break-contact).
For the available functions of the relay see ”Relay Configuration” on page 50.
Inputs
GND It is possible to connect floating contacts to the terminals „IN 1 , IN 2, IN 3,
IN 3
IN 3 IN 2 GND“.
IN 1 The type of the contacts to be connected (standard input or direct dialling
IN 2
IN 1 module) can be defined by means of configuration via the web interface
see page 49.
Connection of Handset /Headset

It is possible to connect a handset or headset to the terminals “EP+, EP–, EM+, EM–” (see connection
diagram page 29).
The way the audio output (internal loudspeaker and/or loudspeaker handset/headset) and the audio
feed-in (internal microphone or microphone handset/headset) is carried out at the SIP station, is
defined by means of configuration via the web interface see page 46.
Connection of an additional Loudspeaker

It is possible to connect an additional loudspeaker (e.g. WSLM 56/WSLM 52) or an amplifier e.g. series
AF to the terminals “LINE+” and “LINE–”. When using loudspeakers of series WSLM 56 or WSLM 52 it
is possible to use also the MUTE function (terminals “MUTE, GND”).
The way audio is put out at simultaneous use of an internal and external loudspeaker (and a handset/
headset), is defined by means of configuration via the web interface see page 46.
Connection of additional IP Devices

Via the RJ 45 socket “IP Downlink” (see connection diagram on page 29), additional IP devices can be
connected (e.g. IP camera, PC).
Attention:
Power supply via PoE is not available via this RJ 45 connection!
Reset Button
Via the reset button a factory reset of the SIP station can be carried out (while rebooting) see page 55.
Slots for Button Labels

Depending on the button version of the station (see page 10), the button labels can be inserted into the
three slots of the PCB.
30 2.0/0612
SIP Series M
Note:
The illustration shows a SIP-ET908A; but the connection diagram also applies to the SIP-ET908A-1.
Connection of Direct Dialling Buttons:

Up to 3 buttons at the terminals X,0,T
Loudspeaker ุ 8 or 4 Ohm
Line In
Handset: Microphone
Loudspeaker
LINE +
LINE–
MIC+
MIC–
GND
GND
EM+
EM–
EP+
LS2
LS1
EP–
IN3
IN2
IN1
X0T
Factory
Reset Button
OUT2
OUT2
LED+
PWR1
PWR2
IP Uplink IP Downlink
LED
OUT1
OUT1
+ PoE (e.g. IP-camera)
12 24 VAC
500 mA
15 35 VDC
LAN Connection
“Ethernet LAN” is connected to the RJ 45 socket “IP Uplink” (see connection diagram above).
Power Supply
For connection of power supply, 2 different possibilities are available:
PoE
The SIP module is supplied via “Power over Ethernet” (Standard IEEE 802.3af) via the RJ 45 socket “IP
Uplink” (see connection diagram above).
DC–, DC+
If ”PoE” (“Power over Ethernet”) is not available, the SIP module alternatively can be supplied with
“12 – 24 VAC or 15 – 35 VDC” via the screw terminals “PWR1” and “PWR2”.
2.0/0612 31
Connection Commend SIP Series
Attention:
For connection of the power supply, the notes and technical data on page 6 have to be observed!
Note:
If both power supply types (i.e. PoE and supply via “PWR1,PWR2”) are active at the same time, then the
following has to be considered:
 In case of a breakdown of the power supply via “PWR1,PWR2”, and the consequentially take over of
PoE, a reboot of the station is initiated.
 In case of a breakdown of PoE, and the consequentially switch-over to power supply via
“PWR1,PWR2”, the operation of the station will not be interrupted – provided that the 2nd ethernet
link (“IP downlink” see connection diagram on page 31) is connected to the respective LAN network.
Relay Outputs
The terminals OUT1 / OUT1 as well as OUT2 / OUT2 are operating as relay outputs (changeover
contacts, OUT as make-contact, OUT as break-contact).
For the available functions of the relay see ”Relay Configuration” on page 50.
Inputs
IN 1
It is possible to connect floating contacts to the terminals „IN 1 , IN 2, IN 3, GND“.
IN 2 The type of the contacts to be connected (standard input or direct dialling module)
IN 3
can be defined by means of configuration via the web interface
see page 49.
GND
IN 3
IN 2
IN 1
Connection of Handset /Headset

It is possible to connect a handset or headset to the terminals “EP+, EP–, EM+, EM–” (see connection
diagram page 29).
The way the audio output (internal loudspeaker and/or loudspeaker handset/headset) and the audio
feed-in (internal microphone or microphone handset/headset) is carried out at the SIP station, is
defined by means of configuration via the web interface see page 46.
Connection Direct Dialling Buttons

T
It is possible to connect up to 3 buttons to the screw terminals “GND, 0, X, T”
0 (position of the screw terminals see page 31).
X The respective call destination of the single direct dialling buttons 0, X, T is
defined via the web interface see page 39.
X0T
GND
32 2.0/0612
Connection Loudspeaker, Microphone

A loudspeaker (4  or 8 ) can be connected to the screw terminals “LS1” and “LS2” (see page 31).
Attention:
Please note that the power output at a loudspeaker with 4  is higher than at a loudspeaker with 8 ;
for technical data see page 6.
A respective electret- or dynamic microphone can be connected to the screw terminals “MIC–” and
“MIC+” (see page 31).
Note:
For the exact technical data of the microphone to be connected see page 6.
Connection of additional IP Devices

Via the RJ 45 socket “IP Downlink” (see connection diagram on page 31), additional IP devices can be
connected (e.g. IP camera, PC).
Attention:
Power supply via PoE is not available via this RJ 45 connection!
Reset Button
Via the reset button a factory reset of the SIP module can be carried out (while rebooting) see page 55.
2.0/0612 33
Configuration via Web Interface Commend SIP Series
Configuration via Web Interface

The configuration of the SIP station has to be carried out via the integrated web interface.
1st Connection
Establish Connection
The SIP stations are delivered ex works with a standard IP address, via which the web interface of the
station can be accessed:
 IP address 192.168.1.200
 Subnet mask 255.255.255.0
Attention:
When connecting the SIP station with this IP-address to the local network (LAN), it is essential to make
sure that this IP-address does not already exist in the network!
If the station can not be used in the local network (LAN) with this IP address, then the following pro-
cedure is recommended:
 Establish connection between PC and SIP station via a hub (or switch) or via a direct connection
cable.
 The PC must be in the same subnet as the SIP station.
 This means, an appropriate IP address of that subnet range (e.g. 192.168.1.199) has to be allocated
to the PC temporarily.
SIP station: PC / Notebook:
192.168.1.200 / 24 e.g.: 192.168.1.199 / 24
Reception
Office
Garage Hub / Switch
Note:
If more stations are configured consecutively, the ARP cache has to be deleted at the PC/notebook.
Login
Enter the IP address of the SIP station in the address bar of the respective web browser
It is recommended to use the following web browsers:
 Mozilla Firefox min. Version 3.5
 Internet Explorer min. Version 8
After entering the IP address, a login dialogue appears where following data has to be entered:
 User name
 factory default: admin
 Password
 factory default: commend
Note:
It is recommended to change the user name and password see ”System Settings” on page 53.
34 2.0/0612
Commend SIP Series Configuration via Web Interface
Note:
The appearance of the login dialogue depends on the used web browser.
After successful login the “home” page of the web interface appears (see page 36).
Note – Factory Reset:

It is possible, to reset all settings carried out via the web interface, to factory default (see page 34) –
therefore see page 55.
Configuration of the SIP Station

The web interface of the SIP station provide different tabs, via which the following settings are
possible:
 Home:
Overview of configuration settings see page 36.
 Network:
Configuration of the appropriate settings for the network into which the SIP station is integrated, as
well as the required settings for the use of the SNMP and NAT function see page 37.
 SIP:
Configuration of the required settings for the respective SIP provider and/or SIP PBX. Furthermore,
call settings and settings for the LED are configured in this tab see page 39.
 Phonebook:
Configuration of the desired call destinations. It is also possible to configure call chains complete
phonebooks can be created and saved. see page 45.
 Audio:
Configuration of the microphone sensitivity, loudspeaker, echo and noise suppression, pre-recorded
audio etc. see page 45.
 Input:
Configuration of input contacts see page 45.
 Output:
Configuration of relay outputs, e.g. as attendant contacts see page 50.
 System:
 System Settings: Modification of user accounts, firmware updates and configuration of the
backlight see page 53.
 SIP Trace: Current log data is displayed.
2.0/0612 35
Overview Configuration Settings

The home page provides an overview of the current software and hardware versions and configuration
settings of the SIP station – the data is only informative, i.e. no configuration can be carried out via this
page.
Device Info
In the first section the following device information is indicated:
 The type designation of the SIP station.
 The software and hardware version of the SIP station.
 The time period the SIP station is already in operation.
Network Info
 In this section the network settings configured at tab Network (see page 37) are indicated.
SIP Info
 In this section the relevant SIP settings configured at tab SIP (see page 39) are indicated.
36 2.0/0612
Network Settings
Select tab Network – following dialogue is indicated:
 Any modifications made in the fields are taken over in the running configuration as soon as the but-
ton (at the bottom right) is clicked and a reboot has been made.
 The reboot has to be carried out manually at tab System see page 53.
 DHCP enabled: If the checkbox is activated, the required IP settings (i.e. the described settings
below) will be requested from a DHCP server automatically after reboot.
 This also means, that the settings made in the fields IP, Subnet Mask, Gateway, DNS have no
effect on the running configuration.
 IP: The IP address is assigned manually –
required only, if DCHP is deactivated.
 Subnet Mask: The appropriate subnet mask is entered manually –
 Gateway: The IP address of the router or standard gateway is entered manually –
 DNS: The IP address of the DNS server is entered manually –
Attention:
Changes of these settings may not be carried out without the permission of the system administrator!
Incorrect IP settings may lead to network instabilities!
2.0/0612 37
SNMP Settings
In the section SNMP Settings the settings required for the use of the monitoring function via the
“Simple Network Management Protocol” are entered.
Note:
The definition and use of the function is explained at ”SNMP” on page 58.
 SNMP enabled: With activation of the checkbox, the SNMP function is activated.
Allowed IP: The IP address of the PC to be authorised for monitoring the SIP station, has to be
entered (so called “management station”).
 Trap Destination: The IP address of the PC to which “traps” (station messages) shall be sent, has
to be entered.
NAT Settings
In the section NAT Settings the settings required for the use of the SIP devices behind a NAT, are en-
tered.
WHAT IS STUN?
Session Traversal Utilities for NAT is a network protocol, i.e. a client-server protocol based on UDP, for
detecting firewalls and NAT routers and to traverse NAT routers.
STUN detects the current IP address of the connection. Thus, e.g. a SIP telephone can detect and commu-
nicate its own current IP address. This is important, in order the conversation partner is able to address its
conversation data correctly.
Therefore, a request to a server (“STUN server”) is sent. The answer of the server includes a public “IP
address”, i.e. the IP address and the sender UDP port of the SIP device (“client”). This address is coded
with a XOR mask, to avoid a translation of the address via a NAT router.
 STUN enabled: With activation of the checkbox, the STUN function is activated.
 STUN Server: Entering the IP address incl. port of the “STUN server”: <IP>:<Port> (e.g. stun.sip-
gate.net:10000).
 Keep Alive Messages Enabled: With activation of the checkbox, “dummy” data packages will be
sent regularly to the SIP Server to keep the connection over NAT open.
Note:
These dummy packages are sent constantly, i.e. no matter if actions are carried out at the station.
38 2.0/0612
SIP-, Call- and LED Settings

Select tab SIP – following dialogue is indicated:
Any modifications made in the fields are taken over in the running configuration as soon as the button
(at the bottom right) is clicked. This may take a couple of seconds.
Note:
The information put in parentheses (next to the appropriate fields) represent standard values or support
for configuration.
SIP Settings
In this section, the general settings enabling a communication via SIP, have to be configured.
 Display Name: An optional text (“Caller-ID”) for the SIP station can be entered.
Note:
The display name may not contain any special characters and not more than 200 digits.
 Local SIP Port: Here the port for the SIP protocol can be changed.
Standard: 5060
 Local RTP Port: Here the UDP port for the RTP protocol can be changed.
Standard: 16384
2.0/0612 39
 Registration max. Expires: Here the timer is set, which is responsible for the update of the regis-
tration of the station at the SIP server. I.e. the entered time in seconds is sent by the SIP station to
the SIP server (Proxy see below). The server is sending an acknowledge or alternate time interval
back, which will then be set and used immediately. I.e. within this time interval the registration of the
SIP station at the SIP server will be updated regularly.
Standard: 3600 sec.
Note:
The registration is updated 60 seconds before the timer expires (but not less than 40 seconds).
Server Settings
In the section Primary Server, settings required for the connection to a SIP server, have to be confi-
gured.
 Registration Enabled: If this checkbox is activated, the SIP station tries to register (with the appro-
priate User and Password) at the SIP server specified at Proxy (see below).
Note:
In case of operation without server, this checkbox has to be deactivated.
 User: The call number of the SIP station for the SIP registration has to be entered, i.e. the call
number of the corresponding SIP account on the SIP server.
Note:
For some SIP servers it may be required to enter additionally the domain  “user name@domain”.
 Password: The corresponding password (defined at the SIP server) for the SIP registration has to
be entered (required for authentication).
 Proxy: The IP address or URL of the appropriate SIP server to which a connection shall be establis-
hed (i.e. at which the SIP station shall be registered).
Additional Server
In the section Secondary Server, a connection to an additional SIP server can be configured. Thus a
better reliability of the appropriate SIP station is achieved (e.g. if the connection to the primary server
is lost, the SIP station can still be reached via the secondary server).
 Secondary Server Enabled: With activation of this checkbox, the connection to a second server is
enabled. I.e. the SIP station tries to register (with the respective User and Password) at the SIP ser-
ver, which is entered at Proxy.
 User / Password / Proxy: see above
Call Settings
In this section, the settings required for establishing a call, have to be configured.
 Answer Call: Combo-box for configuration of the call acceptance.
Menu items:
 Autoanswer: An incoming call is accepted without any ringing and without pressing any button.
 Button <x>: The call is not accepted until the corresponding button of the SIP station is pressed.
Note:
The number and the labelling of the buttons in the drop-down menu depends on the used station ver-
sion (see page 10) and the number of configured direct dialling modules in use (see page 49).
 Any Button: The call can be accepted via pressing any button of the SIP station (however it has
to be pressed one button).
 Cancel Call: Combo-box for configuration of how a running conversation (between two SIP de-
vices) shall be cancelled.
40 2.0/0612
Menu items:
 Disabled: A running conversation can not be cancelled at the SIP station (i.e. only at the respec-
tive remote station).
Note:
The feature is very useful for emergency situations, where e.g. only the emergency center shall be
able to cancel the call – so it is ensured that the initiator of the emergency call can not cancel the call
unintentionally.
If a “Call Time Limit” is configured (see below), then the conversation is cancelled automa-
tically after this defined time.
 Button <x>: Besides the “Call Time Limit” the conversation can only be cancelled with the cor-
responding button at the SIP station.
Note:
 Any Button: Besides the “Call Time Limit” the call can be cancelled via pressing any button of
the SIP station.
 Call Time Limit: Time in seconds, after which a running conversation (between two SIP devices) is
cancelled automatically.
 If the “call time limit” shall be deactivated, then “0” has to be entered.
 Allow Arbitrary Dialing: If the checkbox is activated, it is possible to dial any desired number. I.e.
it is possible to dial a number which is not saved in the ”Phonebook” (if the checkbox is deactivated,
only call destinations saved in the phonebook can be dialled).
As soon as the number is entered, the Enter button has to be pressed, in order to set up the call.
Note:
This feature only refers to stations with full keypad and registration at a SIP server. At serverless
operation no arbitrary dailing is possible.
LED Configuration
In this section, settings for the LED behaviour for the following states can be configured:
 Outgoing Call – Outgoing call from the SIP station
 Incoming Call – Incoming call from the SIP station
 In Call – Running conversation between two SIP devices
 Standby – Idle state (inactivity) of the SIP station
Note:
Idle state = registered at SIP server or in case of operation without server: “ready for operation”.
 Error – Error in connection to SIP server, signalling or other problems

Following settings regarding LED behaviour (for the states listed above) are possible:
 On: LED is permanently on
 Off: LED is permanently off
 Flashing:Fast: LED blinks in a fast rhythm
 Flashing:Slow: LED blinks in a slow rhythm
Note:
The LED (currently) can indicate the colour RED only.
2.0/0612 41
Phonebook Settings
Select tab Phonebook – following dialogue is indicated:
Note:
The information put in parentheses (next to the appropriate fields) represent standard values or support
for configuration.
Configuration Call Destinations

In the section Add new Contact the desired call destinations are added.
 Name: Here an optional name for the call destination can be entered.
 Call Destination: In this field the respective call destination is entered.
user...call number of the respective call destination
user@server address
server address...IP address or host name of the SIP Server
When the respective button of the SIP station is pressed,
E.g.: [email protected] the station dials number “307” and sends the number to
the SIP server “192.168.170.8”.
When the respective button of the SIP station is pressed,
E.g.: [email protected] the station dials number “1234” and sends the number to
the SIP server “sipgate.at”.
user@proxy:port - E.g.:
It is also possible to enter the port additionally (optional)
[email protected]:5061
Note:
 If a SIP station is called, and the call is not acknowledged within 60 seconds, then this non-acceptance
is indicated to the caller and the call potentially has to be initiated again.
 Entering a port is only required for serverless operation <user.domain:port>
42 2.0/0612
 Speed Dial Number: When using a SIP station with full keypad, a direct dialling number can be
entered here for dialling the respective call destination.
Note:
 For 1-,2-,3-button versions this field has no function.
 Besides the buttons 0–9 it is also possible to use the button T for direct diallings (e.g. T1, T3...)
 Direct Dialling Button: Selection of the direct dialling button for the call destination.
Note:
 The number and the labelling of the buttons in the drop-down menu depends on the used station
version (see page 10) and the number of configured direct dialling modules in use (see page 49).
 When using a handset, it is possible to select “Handset offhook”. Via this feature it is enabled, to dial
the number of the respective call destination by simply going off-hook.
 With click on the button Add, the entered call destination (i.e. the entry) is added to the ”Phonebook”
Configuration Call Chain

With click on the button Dialing sequence the input mask for configuration of a call chain is opened.
I.e. the call is taken over within a configurable time period, the call is forwarded to the appropriate call
destinations.
 Name: Here a name for the call chain can be entered.

 Destination x: In the fields up to 10 call destinations can be entered, which are dialled auto-
matically, if the call is not taken over within a definable time (see field next).
The call destinations are entered as described on page 42.
 next: Here the time in seconds is entered, after which the next call destination of the sequence is
dialled (if the call is not accepted).
 Reiterate: If the checkbox is activated, the call chain will be started once again from Destination
1, after the last call destination has been reached. This process will be repeated until the call is taken
over ( infinite loop).
 Speed Dial Number: When using a SIP station with full keypad, then here a direct dialling number
for dialling the respective call destination (and therewith the call chain) can be entered.
Note:
For 1-,2-,3-button versions the field has no function.
 Direct Call Button: Here the button for direct dialling the respective call destination (and thus the
call chain) can be selected.
Note:
 With click on the button Add, the entered call chain is added to the ”Phonebook”.
2.0/0612 43
 With click on the button Contact the input mask for entering single call destinations is indicated
again see page 42.
Import/Export Phonebooks
In the section Import/Export it is possible to save and import phonebooks.
 Import: With click on the button to the right of the field, a dialogue for loading a file is opened. Here
a already created phonebook (.csv) can be selected.
With click on the button Import the selected file is imported, i.e. loaded.
Attention:
If a previously saved phonebook (e.g. phonebook.csv) is loaded, then this .csv file overwrites the current
entries in section Phonebook!
 Export: With click on the button Export the current Phonebook is exported, i.e. saved as .csv file in
the local network.
Phonebook
In the section Phonebook the entered call destinations and call chains are listed.
 Name / Call Destination / Speed Dial Number / Direct Call Button: Here the configured data
in the input masks Add new contact (see page 42) and Add new dialing sequence (see page 43)
are displayed.
 Edit: With click on the symbol the respective entry is indicated in the appropriate input mask and
thus can be edited.
 Delete: With click on the respective entry is deleted.
44 2.0/0612
Audio Settings
Select tab Audio – following dialogue is indicated:
Any modifications made in the “Audio Settings” fields are taken over in the running configuration as
soon as the button (at the bottom right of the section “Audio Settings”) is clicked.
 Modified values in the fields Mic Sensitivity and Speaker Volume become active immediately du-
ring a running conversation.
 Modified values in the fields Noise Suppression and Echo Suppression become active not before
the next conversation.
 Mic Sensitivity: Configuration of the desired sensitivity of the (connected) microphone of the SIP
station in 7 levels (–12 dB to +6 dB).
Default: 0 dB
 Speaker Volume: Configuration of the desired volume level of the (connected) loudspeaker of the
SIP station in 13 levels (–24 dB to +12 dB)
Default: 0 dB
Note:
The configured level defines the volume of the ring tone (at an incoming call) as well as the volume of
the different signal tones.
 Noise Suppression: Configuration of the background noise suppression in 5 levels:

 Minimum level “1 (low)” to maximum level “4 (maximum)”.
 If no background noise suppression shall be used, then “0 (off)” has to be selected.
Default: 1 (low)
 Echo Suppression: Configuration of the acoustic echo canceller (AEC) in 5 levels:

 Minimum “Echo Cancelling” level “0 (minimum)” to maximum level “4 (maximum)”
Default: 2 (medium)
2.0/0612 45
Attention:
For changing the default values of the microphone sensitivity / volume level (in particular for conver-
sations between 2 SIP stations), in combination with changes of the values for the noise and echo
suppression, the information regarding audio configuration on page 62 shall be read previously.
Audio In/Out
In the section Audio In/Out the configuration of the audio output and audio feed-in is carried out:
 Internal Loudspeaker: If the checkbox is activated, the audio signal is put out via the internal loud-
speaker.
 Line out: If the checkbox is activated, the audio signal is put out via the terminals “Line+” and
“Line–” (connection possibility for an external loudspeaker see page 29).
Attention:
This function is not available for the SIP Series M.
The terminals “Line +” and “Line –” function for devices of the SIP Series M as “Line In”. I.e. the ter-
minals can be used for feed-in of audio (e.g. music, radio conference).
 Handset Loudspeaker(EP): If this checkbox is activated, then the audio signal is put out via the
terminals “EP+” und “EP–”, i.e. via the loudspeaker of the handset/headset (see page 29).
 Handset Loudspeaker(EP) overrides Audio Out: If this checkbox is activated, then the audio si-
gnal is put out only via the terminals “EP+” and “EP–” as soon as a handset/headset is connected
(even if Handset Loudspeaker(EP) is not activated). This also means, even if Internal Loud-
speaker and Line out are activated, the audio signal will only be put out at the connected handset/
headset.
Attention:
 This function is not available for the handset WSSH 50P. If the audio signal shall be put out via the
loudspeaker of the handset WSSH 50P – as soon as the handset is connected – then the checkbox
Handset Microphone (EM) overrides Audio In and Out has to be activated see below.
 This also applies to handsets/headsets. which can only be detected via the microphone input.
 Internal Microphone: If the option field is activated, audio is fed-in only via the internal micro-
phone (i.e. terminals “EM+” and “EM–” for microphone of the handset/headset are deactivated).
 Handset Microphone (EM): If the option field is activated, audio is fed-in only via the terminals
“EM+” and “EM–”, i.e. via the microphone of the handset/headset (the internal microphone is deac-
tivated).
 Handset Microphone (EM) overrides Audio in: If the checkbox is activated, then the audio signal
is fed-in only via the terminals “EM+” and “EM–” as soon as a handset/headset is connected (even
if Handset Microphone(EM) is not activated). This also means, even if Internal Microphone and
is activated, the audio signal will only be fed-in via the microphone of the connected handset/head-
set.
 Handset Microphone (EM) overrides Audio In and Out: This checkbox is required for the hand-
set/headset, which can be detected (by the respective SIP device) via the microphone input only, like
the handset WSSH 50P.
I.e. if the audio signal shall be put out only via the terminals “EP+” and “EP–” when using a
WSSH 50P (as described at Handset Loudspeaker(EP) overrides Audio Out above), then this
checkbox has to be activated. Additionally to that, audio will be fed-in via the terminals “EM+” and
“EM–” as described at Handset Microphone (EM) overrides Audio In above.
Pre-Recorded Audio
In this section, audio files (pre-recorded audio) can be loaded into the memory of the SIP station:
Memory: max. 60 seconds
Accepted: Recommended:
Audio format: WAV / 8– 48 kHz WAV / min. 16 kHz
8, 16 bit / mono,stereo 16 bit / mono
Attention:
One single WAV file may not exceed the file size of 5 MB!
46 2.0/0612
Pre-recorded audio can be used for the following states:

 Outgoing Call Destination <x> – Outgoing call from the SIP station, initiated via the respective
button (e.g. 0, T or X).
 The audio file can be configured e.g. as calm down message at call initiation or as individual call
tone for the call initiator.
Note:
The labelling and the number of the “call destination” fields depends on the used station version
(1-,2-,3-button version see page 10).
 Incoming Call – Incoming call to SIP station.

 The audio file can be configured as individual calling signal (and/or ring tone).
Note:
In order that the calling signal (ring tone) is put out at the station, “Autoanswer” may not be confi-
gured for call acceptance see page 40.
 Error – Error in establishing a call (e.g. call destination can not be reached or registration not carried
out etc.).
 Pre-recorded audio can be used e.g. as acoustic signal for line fault
 Location Message – The called SIP device (e.g. mobile phone) receives a location message as infor-
mation where (i.e. from which SIP device) the call comes from.
Loading Pre-recorded Audio:
 In the desired line click on the 1st button to the right of the blank field (“Browse...”)
Note:
The labelling of the button depends on the used browser and operating system.
 In the appearing dialogue select the desired audio file

 Via the button Upload the selected audio file is loaded into the station.
Note:
 Uploading, recoding and saving the audio file can take some seconds:
 Loading time = 2 x file length;
 E.g. file with 10 seconds: loading time = 2 x 10 = 20 seconds
 All audio files together may not exceed the max. length of 60 seconds see page 46.
2.0/0612 47
Indication of the file length:

If the desired audio files are loaded into the memory of the SIP station, the length of the respective
audio file is displayed in the appropriate line ( 2 ) and the total length of all loaded files ( 3 ) are dis-
played in the bar below the fields in the section “Prerecorded Audio”.
1 2
Remove Loaded Audio Files from Memory:

 In case of a loaded audio file, the button Restore Default ( 1 ) is indicated in the respective line.
 Click on this button to remove the previously loaded audio file from the memory; the respective state
will then be reset to the default settings.
Note:
 Default setting for “Incoming Call” and “Outgoing Call (Button <x>)” means, that the factory default
tones become active again.
 Default setting for “Error” and “Location Message” means, that no signal tones are put out for these
states.
48 2.0/0612
Input Configuration
Select tab Input – following dialogue is indicated:
Configuring Inputs
In the sections Input 1 to Input 3 separately for all 3 inputs of the SIP station, the following option
fields are available:
 None: The respective input is deactivated (connected contacts can not be used).
 Standard Input: The input is activated for potential-free contacts (but no input level detection).
 3-Button Module: The input is activated for use of the direct dialling module WSDD 53V.
 9-Button Module: The input is activated for use of the direct dialling module WSDD 59V.
Note:
 If a button of the direct dialling module WSDD 53V/WSDD 59V is configured for dialling a call des-
tination in the ”Phonebook”, then the respective direct dialling module can not be deactivated.
Therefore, another button has to be configured for the respective call destination or for the respective
call chain in the phonebook (or the entry in the phonebook has to be deleted), see page 42 and
page 43.
 Input contacts can only be used for call initiation (i.e. no call acknowledge, delete call...)
Resistance Values for customer specific input circuits

Instead of using the direct dialling modules WSDD 53V and WSDD 59V, it is possible to create indi-
vidual resistance circuits for input contacts, to which the desired buttons can be connected.
Following table shows the appropriate resistance values for the respective buttons 1 – 9:
Button Range [] Value []
1 0 – 260 100
2 270 – 760 560
3 770 – 1k2 1k
4 1k3 – 2k1 1k5
5 2k2 – 3k7 2k7
6 3k8 – 5k6 4k7
7 5k7 – 7k4 6k0 (3k3 + 2k7)
8 7k5 – 11k3 8k2
9 11k4 – 20k 12k4 (6k8 + 5k6)
2.0/0612 49
Relay Configuration
Select tab Output – following dialogue is indicated:
50 2.0/0612
Configuring Relays
In the sections Relay 1 and Relay 2 the following settings can be configured separately for both relays
of the SIP station:
 Status: Indication of the current relay status.
 Manual Actions: Clicking on the buttons in this row triggers the following relay actions directly
from the web interface:
 On: Close relay output (activate)
 Off: Open relay output (deactivate)
 Flashing: Switch relay output to blinking (switch on and off in rhythm of one second)
 Door opener – door opener function: relay output is closed for a definable time (see below “Door
opener Timer”).
 Toggle: Invert current state of the relay output (switch on/off)
 DTMF Password Enabled: If the checkbox is activated, a password has to be entered at the SIP
telephone, (see DTMF Password), in order to be able to trigger the respective relay of the SIP sta-
tion via DTMF after dialling at the SIP telephone.
 DTMF Password: The appropriate DTMF password for authorization of DTMF after dialling, for
triggering the relay of the SIP station, at the SIP telephone has to be entered (configuration of but-
tons for DTMF after dialling see below).
PROCESS OF ENTERING THE PASSWORD:
 After pressing a button at the telephone, an appropriate prerecorded audio file (“Password”) is si-
gnalling that the password has to be entered now (“password mode”).
 With it a timer of 10 seconds is started.
 If no button is pressed within these 10 seconds, then the “password mode” is cancelled.
 If the correct button for the respective digit of the defined DTMF Password has been pressed within
the 10 seconds, then the 10 seconds timer is restarted for entering the next digit of the password.
 If the entered password is not correct, then a double tone is put out and the “password mode” is
cancelled.
 If the entered password is correct, then a single tone is put out and the appropriate relay of the SIP
station is triggered.
 Button for On / Off / Flashing / Toggle / Door opener:

Selection of the desired DTMF-button (0–9, *, #) of a SIP telephone for control of the relay outputs
(see above). Control of the relays is carried out via DTMF after dialling during the conversation.
 Door opener Timer: Enter the time in seconds how long the relay shall stay active after pressing
the door opener button.
Default: 2 seconds
Configure Relay as Attendant Contact

In the sections Attendant Contacts Relay 1 and Attendant Contacts Relay 2 it is possible to
separately configure the relays as attendant contact for the following states:
Note:
The “attendant contacts” (i.e. the relay outputs) will stay in the triggered mode, as long as the respective
state is active (for relay modes see below.
 Outgoing Call – Outgoing call from the SIP station

 Incoming Call – Outgoing call to the SIP station
 In Call – Active conversation (between two SIP stations)
 Standby – Idle state of the SIP station
Note:
Standby = registered at SIP Server or at server less operation: ready for operation.
 Error – Registration of station not possible, due to e.g. fault in connection to the SIP server, si-
gnalling or other problems.
2.0/0612 51
The following relay modes are possible for the above mentioned states:
 Nothing: Deactivate attendant contact (relay not used)
 On: Close relay output (activate)
 Off: Open relay output (deactivate)
 Flashing: Switch relay output to blinking (switch on and off in rhythm of one second)
 Door opener – Door opener function: relay output is closed for a definable time (“Door opener
Timer” see above).
Note:
If the respective state (for which “Door opener” has been selected) changes while the “Door opener
Timer” is active, the relay will be triggered immediately – i.e. before the configured door opener timer
has expired.
52 2.0/0612
System Settings
Select tab System.
A dropdown menu is opened with following options for selection:
 ”System Settings” – Configuration of remote control and the background light, as well as firmware
updates and reboots are carried out.
 ”SIP Trace” – Indication of SIP messages.
System Settings
All changes in the fields at User Accounts and Background Light are taken over as soon as the but-
ton (at the bottom right) is clicked and a ”Reboot the SIP station” is carried out.
This may take a couple of seconds.
Firmware Update and Reboot

In the section System Settings a firmware update as well as a reboot of the SIP station can be carried
out.
Update of station firmware
 In the row Firmware update click on the left button (Browse...)
Note:
The description of this button depends on the browser and operating system used.
 Select the desired firmware in the dialogue

 Click Update to download the selected firmware in the station
2.0/0612 53
Attention:
 The dialogue for acknowledgement of the start of the download process is indicated 30 seconds after
the update button has been pressed.
The update button may not be pressed (again) during the active download!
 Update from firmware 1.x to 2.0: Settings like e.g. configuration of buttons (”Phonebook Set-
tings” are not saved!
Therefore it is recommended to carry out a Factory Reset after updating to firmware 2.0!
Reboot the SIP station

 To reboot the station, click on the button Reboot.
!! ATTENTION !!
During a firmware update the power supply of the SIP station may not be disconnected under
any circumstances – THIS CAN DAMAGE THE STATION!
The firmware update is completed with the reboot of the SIP station.
User Accounts
In the section User Accounts the user settings can be administered:
 Webuser Login: Here the user name for the web interface login can be changed.
 Webuser Password: Here the password for the web interface login can be changed.
 Remote control User Login: Definition of a user name for remote access to the SIP station via
HTTP.
 Remote control User Password: Definition of a password for remote access to the SIP station via
HTTP.
Note:
How to access the relay outputs by means of a “remote control user account” is described at ”Remote
Control”.
Background Light
In the section Background Light the configuration of the backlight for the call buttons and label areas
of the SIP station can be carried out:
 Active Level: Configuration of the intensity of the backlight in active mode (e.g. while dialling a call
number) in 10 levels:
 Minimum level “1 (dimmest)” to maximum level “9 (brightest)”.
 If no modification of the backlight at activity shall take place, then “0 (off)” has to be selected.
Default: 8
 Idle Level: Configuration of the intensity of the backlight for idle state of the SIP station in 10 levels:
 Minimum level “1 (dimmest)” to maximum level “9 (brightest)”.
 If no modification of the backlight in idle state shall take place, then “0 (off)” has to be selected.
Default: 0 (off)
 Idle Delay: The time (in seconds) of inactivity is defined, after which the backlight illumination is
switched to idle mode (see above).
Default: 30 seconds
 Link with Multifunction LED: If the checkbox is activated, the behaviour of the backlight illumina-
tion is linked with the LED behaviour (see page 41). This means, as soon as the LED is on, also the
backlight for the call buttons and label areas is switched on; if the LED is blinking, also the backlight
will be blinking etc. (if e.g. an distinctive optical call indication is required, this function may be the
ideal solution).
Default: deactivated
54 2.0/0612
SIP Trace
If SIP Trace is selected from the drop-down menu, then a list with the current SIP messages are indica-
ted. This makes it easier to reconstruct actions and enables a more effective troubleshooting.
 The latest messages are indicated at the top of the list.
 Only the last 20 messages will be indicated.
 With click on the button Clear all SIP messages are deleted in the list.
 With click on the button Reload the list will be updated.
Factory Reset
It is possible to reset all changes made in the web interface settings to factory default:
 To reset to factory settings press the “Reset button” on the PCB (see page 29 or page 31) for about
20 seconds during reboot of the SIP station (or at startup after powering on) until the status LED
starts to blink.
2.0/0612 55
Remote Control
To be able to control different operating states for automated processes (or without the authorisation
of a “full access” onto the web interface), the possibility for establishing a “remote control account”
for following actions of the respective station is available:
 Switching Relay outputs
 Establish Call
 Simulation of keypress
 Take over / Cancel / Refuse Call
 The remote control account for the “remote control user” is established via the web interface
see ”User Accounts” on page 54.
The remote control to the SIP station is enabled by sending HTTP strings.
The HTTP string is made up of 2 parts:
The 1st part of the HTTP string is universally valid and required for all actions to be carried out
http://<User>:<PW>@<IP-Adr.>/cgi-bin/remotecontrol/...
 <User>: User name for remote access (see “Remote control User Login” on page 54)
 <PW>: Password for remote access (see “Remote control User Password” on page 54)
Note:
It is not necessarily required to enter the user name and password in the HTTP string. If the user and
password is not included in the HTTP string, then this data has to be entered after copying (and acknow-
ledging) the string into the browser.
 <IP address>: IP-Address of SIP station
The 2nd part of the HTTP string is made up of action-specific parameters:
Parameters for Switching Relay Outputs

http://<User>:<PW>@<IP-Adr.>/cgi-bin/remotecontrol/relais.cgi?relais=<1od.2>&action=<Trigger>
 <1 or 2>: value 1 for relay 1, value 2 for relay 2

 <Trigger>: desired trigger mode of relay (see also page 51):
 on: close relay output (activate)
 off: open relay output (deactivate)
 flashing: switch relay output to blinking
 dooropener&dooropenertimer=<time in sec> – door opener function: relay output is closed
for a definable time (“dooropenertimer” e.g. 3 seconds).
 toggle: invert current state of the relay output (switch on/off)
Example:
Closing (activating) relay 1 of SIP station 192.168.123.234 (User: Hudson, PW: Saul):
 http://Hudson:[email protected]/cgi-bin/remotecontrol/relais.cgi?relais=1&action=on
Switching relay 2 of SIP station 192.168.11.22 to blinking (User: Ian, PW: Fraiser):
 http://Ian:[email protected]/cgi-bin/remotecontrol/relais.cgi?relais=2&action=flashing
Closing (activating) relay 1 of SIP station 172.16.9.222 for a (door opener) time of 4 seconds:
 http://172.16.9.222/cgi-bin/remotecontrol/relais.cgi?relais=1&action=dooropener&dooropenertimer=4
Parameter for Simulation Keypress

http://<User>:<PW>@<IP-Adr.>/cgi-bin/remotecontrol/button.cgi?button=<button>
56 2.0/0612
 <button>: desired button to be simulated; possible actions:

 1-,2,-3-button version: b1 (button 1), b2 (button 2), b3 (button 3)
 Stations with full keypad: 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, T, X, menu, up, down, enter
 SIP-modules (SIP-ET 908A): 0, T, X
Note:
For the different station versions see ”SIP Series Versions” on page 10.
Examples:
Simulation of button 2 of SIP-WS 203V (IP: 192.168.123.234 – User: Hudson, PW: Saul):
 http://Hudson:[email protected]/cgi-bin/remotecontrol/button.cgi?button=b2
Simulation of button 6 of SIP-WS 500F (IP: 192.168.11.22 – User: Ian, PW: Fraiser):
 http://Ian:[email protected]/cgi-bin/remotecontrol/button.cgi?button=6
Simulation of Enter button of SIP-WS 800V (IP: 172.16.9.222 – User: SD, PW: Chargers):
 http://SD:[email protected]/cgi-bin/remotecontrol/button.cgi?button=enter
Parameter for Take over / Cancel / Refuse Call

http://<User>:<PW>@<IP-Adr.>/cgi-bin/remotecontrol/hook.cgi?hook=<Call, Conversation>
 <Call, Conversation>: desired action at incoming call or active conversation:

 take over incoming call: answer
 refuse incoming call or cancel active conversation: cancel
Examples:
Take over incoming call at SIP station 192.168.123.234 (User: Hudson, PW: Saul):
 http://Hudson:[email protected]/cgi-bin/remotecontrol/hook.cgi?hook=answer
Cancel active conversation at SIP station 192.168.11.22 (User: Ian, PW: Fraiser):
 http://Ian:[email protected]/cgi-bin/remotecontrol/hook.cgi?hook=cancel
Refuse incoming call at SIP station 172.16.9.222 (User: SD, PW: Chargers):
 http://SD:[email protected]/cgi-bin/remotecontrol/hook.cgi?hook=cancel
Parameter for Establishing a Call  only with full keypad (SIP-WS 800x/500x)
http://<User>:<PW>@<IP-Adr.>/cgi-bin/remotecontrol/call.cgi?destination=<Call destination>
 <Call destination>: desired call destination; possible settings:

 call number only, e.g. 5000
Note:
This setting (Call no. without IP address) is not possible for serverless operation!
 Call Number + Hostname or IP address of the SIP Server, e.g. [email protected]
Examples:
Establish call from station 192.168.123.234 to call destination 5000 (User: Hudson, PW: Saul):
 http://Hudson:[email protected]/cgi-bin/remotecontrol/call.cgi?destination=5000
Establish call from station 192.168.11.22 to call destination [email protected] (User: Ian, PW: Fraiser):
 http://Ian:[email protected]/cgi-bin/remotecontrol/[email protected]
2.0/0612 57
SNMP
General Definition
The “Simple Network Management Protocol“ (SNMP) is a network protocol for controlling and moni-
toring network elements (e.g. router, server, switches, SIP stations etc.) via one central management
station/PC. In this process the protocol is controlling the communication between the monitored de-
vice and the monitoring station. SNMP describes the structure of the data packets as well as the
communication process.
The state of the respective device is detected by programs (so called “agents”) which are running di-
rectly on the monitored device (i.e. SIP station). By means of the SNMP, the “management station” is
able to communicate with this “agent” via network.
“Agent 1”
e.g. switch
SNMP
“Agent 2”
Management station Reception
Office
Garage
SIP station
Management Information Base

The respective values (of a SIP station) read out by the management station, are described in a
“Management Information Base” (MIB). MIBs are description files in which the single values are listed
in table form. A MIB contain values specifically from one single SIP station only.
Using SNMP Function for SIP Stations

For use of the monitoring function via SNMP for SIP stations, following requirements have to be con-
sidered:
 SNMPv2c
 MIB according RFC 1213
 Support for IP-MIB and IF-MIB
 Commend-MIB
 Private Enterprise Number (PEN): 37568
(for download of the MIB see web link on page 69)
MIB Browser
By means of a MIB browser the values can be displayed. There are different providers for this type of
browser; some MIB browsers can be downloaded from the WWW as freeware.
 In order to be able to use the SNMP function for the SIP station, the feature has to be activated as
well as the IP address of the management station has to be entered via the web interface see
page 37.
Attention:
For reading out the MIB values, the “SNMP Version 2” has to be selected in the MIB browser.
58 2.0/0612
MIB Values
In the following tables the most essential values for use of SIP stations are listed:
Note:
In order that the “Name” of the respective value (2nd column see below) is indicated, the file “com-
mend.txt” (Commend-MIB) has to be read out via the MIB browser (for download of the file see web
link on page 69).
Device data Name OID Parameter (“Value”)

<Station series>
Station series sysDescr.0 .1.4.6.1.2.1.1.1.0
e.g. Commend SIP Series
commendStationCommon- <Station series>
Station series .1.3.6.1.4.1.37568.2.1.1.1
StationType.0 e.g. Commend SIP Series
<Station type>
Station type sysName.0 .1.3.6.1.2.1.1.5.0
e.g. SIP-ET908A
commendStationCommon- <Station type>
Station type .1.3.6.1.4.1.37568.2.1.1.1
StationSubType.0 e.g. SIP-ET908A
commendStationCommon-
Software version .1.3.6.1.4.1.37568.2.1.1.3 <Software version>
StationSoftwareVersion.0
Uptime sys.Uptime.0 .1.3.6.1.2.1.1.3.0 <Uptime of SIP station>
Hardware version .1.3.6.1.4.1.37568.2.1.1.4 <Hardware version>
StationHardwareVersion.0
Call number .1.3.6.1.4.1.37568.2.1.1.10 <Call No. of SIP station>
StationCallNumber.0
commendStationCommon- <Display Name of the SIP
Station description .1.3.6.1.4.1.37568.2.1.1.11
StationStationName.0 station>
Station location .1.3.6.1.4.1.37568.2.1.1.12 <Location of the SIP station>
StationLocation.0
Network data Name OID Parameter (“Value”)

Station IP ipAdEntAddr. .1.3.6.1.2.1.4.20.1.1 <IP Address SIP station>
Station MAC ifPhysAddress.2 .1.3.6.1.2.1.2.2.1.6.2 <MAC address SIP station>
Subnet mask ipAdEntNetMask .1.3.6.1.2.1.4.20.1.3 <Subnet mask>
Router IP ipNettoMediaNetAddress.2 .1.3.6.1.2.1.4.22.1.3.2 <IP Address Gateway>
Online status .1.3.6.1.4.1.37568.2.1.1.20 online, offline
StationOnlineState.0
offhook, onhook, ringing,
Conversation status* .1.3.6.1.4.1.37568.2.1.1.80 dialling, user@domain (con-
StationCallState.0
versation partner)
commendStationConnec-
DHCP .1.3.6.1.4.1.37568.2.1.2.1 enabled, disabled
tivityType.0
commendStationConnec-
Registration status* .1.3.6.1.4.1.37568.2.1.2.5.1 registered, not registered
tivitySipRegistration.0
commendStationConnec- <Time>, > 60 min, never,
Last registration .1.3.6.1.4.1.37568.2.1.2.5.2
tivityLastRegistration.0 unknown
*In case of a change of the state, additionally ”Traps” are sent (to the configured host).
2.0/0612 59
Audio data Name OID Parameter (“Value”)

commendStationAudio- <Microphone type>
Microphone type .1.3.6.1.4.1.37568.2.1.3.1.1
Mic1Type.0 e.g. Commend SIP
Microphone commendStationAudio-
.1.3.6.1.4.1.37568.2.1.3.1.2 4 – 10
sensitivity Mic1Sensitivity.0
commendStationAudio- <Loudspeaker type>
Loudspeaker type .1.3.6.1.4.1.37568.2.1.3.10.1
Amp1Type.0 e.g. Commend SIP
commendStationAudio-
Speaker volume level .1.3.6.1.4.1.37568.2.1.3.10.2 0 – 12
Amp1Sensitivity.0
Echo Suppression .1.3.6.1.4.1.37568.2.1.3.50.1 0–4
AECMode.0
Noise Suppression .1.3.6.1.4.1.37568.2.1.3.51.1 0–4
NCMode.0
Relays Name OID Parameter (“Value”)

commendStationOutput-
Description Relay 1 .1.3.6.1.4.1.37568.2.1.5.1.1 Relais 1
Name1.0
Status Relay 1* .1.3.6.1.4.1.37568.2.1.5.1.2 on, off, flashing
State1.0
Description Relay 2 .1.3.6.1.4.1.37568.2.1.5.2.1 Relais 2
Name2.0
Status Relay 2* .1.3.6.1.4.1.37568.2.1.5.2.2 on, off, flashing
State2.0
Inputs Name OID Parameter („Value“)

3 Button Module, 9 Button
commendStationInputType
Type IN1 .1.3.6.1.4.1.37568.2.1.4.1.1 Module, Standard
1.0
Module, None
commendStationInput- No Button pressed, Button
Status IN1* .1.3.6.1.4.1.37568.2.1.4.1.2
State1.0 <Button> pressed
2.0
Module, None
Status IN2* .1.3.6.1.4.1.37568.2.1.4.2.2
3.0
Module, None
Status IN3* .1.3.6.1.4.1.37568.2.1.4.3.2
*In case of a change of the state, additionally ”Traps” are sent (to the configured host).
Traps
Independently from the values of the MIB, so called “traps” are sent.
A “trap” is an unrequested message of a SIP station to the management station, reporting that an event
has occurred, e.g. a conversation between two SIP stations has been established.
Note:
It is possible to define a separate destination IP to where these “traps” are sent, via the web interface
see ”SNMP Settings” on page 38.
The traps can be displayed by means of a MIB browser
60 2.0/0612
For the following events and/or at change of the following states, traps are sent from the SIP station to
the management station:
 Registration status (registered, not registered)
 Registration data (who is registered at which domain – primary and secondary server)
 Conversation status (outgoing/incoming call, idle state, conversation)
 Status Call destination (which button is used for which call destination)
 Status Ringing (incoming call no.)
 Refuse call
 Station busy
 Status active Conversation (who is talking to whom – from...to)
 Status Conversation end (who cancelled the conversation and how has it been cancelled)
Name OID Parameter („Trap“)
 standby
 hung up  buttonpressed
 remote hung up
 dialling  <button no.> <user@proxy>,
e.g. button 1, [email protected]
 <module no. button no,> <user@proxy>
e.g. module 2 button 7, [email protected]
commend-
 <short dial no.>, user@proxy
StationOb-
.1.3.6.1.4.1.37568.2.10 e.g. short dial 5, [email protected]
jectStatus
 <arbitrary dialing>
Notification
e.g. [email protected]
 <http request user>, <user@proxy>
e.g. http request 556, [email protected]
 call in  TO: <user> e.g. TO: 556
progress
 ringing  sip:<user>@<domain>
e.g. sip:[email protected]
 register  secondary registration disabled
 registered primary, <user>@<domain>
.1.3.6.1.4.1.37568.2.1.1 e.g. registered primary, [email protected]
 registered secondary, <user>@<domain>
e.g. registered secondary, [email protected]
 Status Relay 1 (on, off, blinking)

 Status Relay 2 (on, off, blinking)
2.0/0612 61
Appendix Commend SIP Series
Appendix
Basic Knowledge about Audio Configuration

In order to carry out a proper audio configuration with the tab Audio (see page 45), basic knowledge
is recommended.
In the following section the various settings and the resulting consequences are explained.
Echo Cancelling according to LS / Mic / Room

“Echo Cancelling” or “Echo Suppression” are methods used to suppress or remove echo or hall effects
at simultaneous sending and receiving of signals (full duplex).
For example, the voice out of the loudspeaker of a hands-free speaking system (e.g. SIP station) during
hands-free speech is picked up by the microphone and transmitted back. Therefore the subscriber on
the other end hears his own voice as an echo.
3
4
The following factors determine which “Echo Suppression”-level (see page 45) is required, to sup-
press Echo on the opposite side:
 Configured volume level of the loudspeaker ( 1 )
 Configured microphone sensitivity ( 2 )
 Properties of the Room (size, angles, materials, furnishing 3 )
 Distance of the speaker to the station (and therefore the microphone of the station 4 )
Effect of the Configuration always at the other Station/Device

Important information for echo cancelling configuration:
 Settings for microphone sensitivity, echo- and noise suppression at the station have direct effects to
the acoustic behaviour at the called station.
 Therefore if echo is audible at a device (Intercom, telephone) the configuration needs to be carried
out at the conversation partner!
62 2.0/0612
Commend SIP Series Appendix
Example 1:
Echo
Configuration
is carried out Effects of
at this station configuration
SIP audible at
conversation
Echo
partner
Reception
* 0 #
Office
Garage
Example 2:
Echo
Echo audible Configuration
at this station has to be
during carried out at
conversation conversation
partner
Echo
Reception
Reception
Office
Office
Garage
Garage
Note:
The standard audio settings (see page 45) are well-proven values and are suitable for a large part of all
applications.
Possible Consequences of to High Volume

In this section the assumption “louder ≠ better” is described.
If the volume level (“speaker volume”) is increased by several dB, counteractive measures may be
required to ensure a echo-free conversation at the station:
 It is possible to decrease the microphone sensitivity (“Mic Sensitivity”) and /or
 the Echo Canceller, i.e. the “Echo Suppression”-level is increased
Too much Echo
Counteractive
Measures
lower higher
“Mic Sensitivity” “Echo Suppression”
Consequence: Consequence:
Speaker must The conversation
be closer to might feel more
microphone or like with switched
speak louder duplex/half duplex
2.0/0612 63
Microphone sensitivity is reduced – consequence:

 In order for the called station to hear the speaker properly, the distance to the microphone must be
decreased or the speaker must speak louder.
Echo Suppression is increased – consequence:

 The more the “Echo Suppression”-level is increased, the more the conversation feels as if switched
with a “speech balance” (like switched duplex/half duplex).
 This “speech balance” takes control of the conversation direction and switches through the louder
side of the conversation and reduces the volume of the quieter side.
WHAT IS A SPEECH BALANCE?

A “speech balance” is used in a “half duplex” conversation.
Because the stations of the SIP Series are “full duplex” capable, the speech balance is not active (ex-
works). Only when a special configuration (see above) is carried out, the conversation is controlled by a
similar behaviour.
For a clearer understanding please read the description of the two conversation modes:
 HALF DUPLEX (SPEECH BALANCED):
The conversation direction is controlled by a “speech balance”. Therefore it is possible to talk and listen
without pressing a button. However only the louder side is switched through.
 FULL DUPLEX:
The conversation is permanently switched through in both directions. Therefore both conversation part-
ners can simultaneously speak and listen (=natural communication).
Unique Feature of the SIP Series

The great advantage of the SIP Series compared to other products is, that the above-mentioned
behaviour can be adapted to the requirements on site and therefore provide the greatest possible
conversation quality.
Serverless Operation
The following section gives an overview of the required basic configuration of a SIP station as well as
a SIP telephone, at serverless operation.
As example a Snom®telephone is used.
A detailed description of the Snom®web interface won´t be found in this document, as on the basis of
the following example solely the basic know-how for the configuration of a serverless operation bet-
ween two SIP devices, shall be communicated.
Connection Commend SIP Station – Snom®Phone

IP: 10.10.1.201 IP: 10.10.1.200
Dummy Account: 124
Snom®
Reception
* 0 #
Office
Garage
64 2.0/0612
Note:
 The following Snom®web interface description relates to version 8.
 The indicators “a1”, “b1”, “c1” in the screenshots of the web interface of the SIP station correlate
with the settings “a2”,“b2”,“c2” in the web interface of the Snom®telephone.
Required Configuration of the SIP Station

 Deactivate checkbox “Registration Enabled” (see also page 40).
a1
WHY THIS SETTING?

The deactivation of the checkbox enables a serverless connection between 2 SIP devices. I.e. there will
be no attempt of the SIP station to register at a SIP server.
 Configuration of the appropriate Call destination (Snom®telephone), i.e.:

 the call number (“dummy account”) of the Snom®telephone
 the IP address of the Snom®telephone
 desired direct dialling number or direct dialling button
(see also page 42).
b1 c1
Required Configuration for Snom®Phone

 Activate “Support broken Registrar” to enable an operation without server.
a2
WHY THIS SETTING?

The activation of the option field “on” enables a serverless connection of the Snom®telephone to a SIP
terminal (e.g. Commend SIP station).
 Create Dummy-Account (e.g. “124” + Password).
b2
WHY THIS SETTING?

As no server is in use, no SIP account for the registration on a server can be created. In order a connec-
tion between the SIP station and Snom®Phone can be established (i.e. the Snom®Phone can be ad-
dressed correctly), a so called “dummy” account (call no. and password) for the Snom®Phone via web
interface is required.
 At “Registrar” the IP address of the conversation partner, i.e. of the SIP station is entered.
c2
2.0/0612 65
Display Menu WS 800V / WS 800F

Following sections provide display menu operating instructions for the SIP stations WS 800V and
WS 800F.
LCD graphic display
Alpha numeric full keypad

Menu button
Arrow buttons Up / Down

Enter button
Help Functions
Possible inputs are indicated at the bottom of the display as help.
Idle Mode
STANDBY
In idle mode, the screen to the right is indicated.
MENU
Main Menu
MENU
 Open the menu in idle mode with menu button . 0 Subscriber ›
 With the arrow buttons an option is selected and initiated with . 1 Volume
 With  the Main Menu can be left without selecting an option. 2 Status
Subscriber: Show the subscriber list X-CANCEL
Volume: Show menu “Volume” UP/DOWN ›OK
Status: Show information about the station (call no., name, versions)
Volume
VOLUME
 Select Volume from the main menu with .
 With the arrow buttons the desired volume is selected.
 Save with  or leave with  without saving any changes.
›SAVE
+/- X-CANCEL
66 2.0/0612
Note:
 The volume can also be controlled during an active conversation.
 With  the respective volume is saved also for the next conversation (as described above).
Status
Here the station data is indicated, like e.g. IP address, registration status, version.
 Select Status from the main menu with .
 Leave the menu with .
General Note:
According to the selected option at Answer Call (tab SIP see ”Call Settings” on page 40) different dis-
play screens at incoming calls are possible.
Call Initiation at SIP-WS 500F

As the SIP-WS 500F do not contain any display, the operation is facilitated with the output of audio si-
gnals.
The way a call is initiated depends on whether the checkbox Allow Arbitrary Dailing (see page 41)
is activated or deactivated:
Allow Arbitrary Dailing NOT activated

 After a button is pressed, it is checked whether the dialled call number is stored in the
”Phonebook”.
 If a match is found, then the call number will be dialled immediately.
 If the call number could not be found in the ”Phonebook”,
 then the complete number is deleted after a timeout of 10 seconds and
 a double tone is put out and the “input mode” is restarted, i.e. a new number can be
dialled.
 If the complete “line” or last digit is deleted, then
 a double tone is put out and input mode is restarted, i.e. a new number can be dialled
 The conversation can be cancelled according to the respective configuration see page 40.
Buttons
Menu button: no function
Volume controllable by arrow buttons
Enter button: only saving volume
press short: delete last digit

press for at least 1 second: delete complete line (call number)
no function
2.0/0612 67
Allow Arbitrary Dailing activated

 After a the Enter button is pressed, the entered call number is dialled.
 If no button is pressed within a timeout of 10 seconds, then
 If the complete “line” is deleted, then
 The conversation can be cancelled according to the respective configuration see page 40.
Buttons
Menu button: no function
Volume controllable by arrow buttons
Enter button: saving volume, setting up the call
press short: delete last digit

press for at least 1 second: delete complete line (call number)
no function
68 2.0/0612
Commend SIP Series Technical Support
Technical Support
For more information about the Commend SIP Series visit:
www.commend.com/sip
2.0/0612 69

Sip Series P

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PRODUCT MANUAL

Commend SIP Series

Downloaded from www.Manualslib.com manuals search engine

Therefore this equipment is CE-labelled.

Before exchanging the modules, ESD precautions have to be observed.

Commend SIP Series

Errors and omissions excepted.

Commend Security for the World of SIP

Speech Connection according SIP Standard

VOIP ACCORDING SIP STANDARD – HOW DOES IT WORK?

SIP Series P, SIP Series V, SIP Series VE only:

SIP Series M only:

SIP Series M only:

SIP Series V, SIP Series VE:

Additional Mounting Material

WSFB 50V: WSFB 50V FL:

WSSH 50V: WSRR 50V:

System Requirements / Compatibility

Compatibility SIP PBX

VoIP Provider Type Version

SIP Series Versions

SIP-WS 201P: SIP-WS 202P: SIP-WS 203P: SIP-WS 800P:

SIP-WS 201V: SIP-WS 202V: SIP-WS 203V: SIP-WS 800V:

SIP-WS 211V: SIP-WS 212V:

SIP-WS 500F: SIP-WS 800F: SIP-WS 800F MD:

SIP-ET 908: SIP-ET 908-1:

SIP Series P, SIP Series F

The mounting instruction is split into following sections:

Measurements of the Front Panel

Measurements Flush Mount Kit WSFB 50P

front ﬂush with plaster 135.6 (5.34) 30 (1.18)*

12x Expansion openings (can be broken out) Ø 21

Mounting Flush Mount Kit WSFB 50P

Measurements Surface Mount Kit WSSH 50P

4x Mounting openings Ø 7.5 (0.3) 160 (6.3)

12x Expansion openings (can be broken out) Ø 21

Mounting Surface Mount Kit WSSH 50P

Pull back tight

SIP Series V, SIP Series VE

The mounting instruction is split into following sections:

Measurements of the Front Panel

Measurements Flush Mount Kit WSFB 50V

12x Expansion openings (can be broken out) Ø 21

Mounting Flush Mount Kit WSFB 50V

4 ..... Flush mount frame (metal)

5 ..... Mounting frame (plastic)

Measurements Flush Mount Kit WSFB 50V FL

front ﬂush with plaster

12x Expansion openings (can be broken out) Ø 21

Mounting Flush Mount Kit WSFB 50V FL

Measurements Surface Mount Kit WSSH 50V

50 (1.97) 114 (4.49)

Mounting Surface Mount Kit WSSH 50V

1 ..... Surface mount box (Metal)

Measurements Rain Protection Roof WSRR 50V

164 (6.45) 80 (3.15)

Mounting Rain Protection Roof WSRR 50V

Front panel (metal) with

Rain protection roof Fixing screws required for