Manual coM.sat IP Basic

Transcription

coM.sat
IP Basic
Manual
www.comsat.de
E-Mail: [email protected]
Support-Hotline: 0180 5 633343
Manual
coM.sat IP Basic
Content
CONTENT ....................................................................................................... 2
FIGURES ........................................................................................................ 5
TECHNICAL TERMS ......................................................................................... 7
1.
INTRODUCTION...................................................................................... 8
2.
SECURITY HINTS FOR THE USER ............................................................ 9
3.
WALL MOUNTING OF THE COM.SAT IP BASIC ......................................... 11
4.
FUNCTIONAL DESCRIPTION ................................................................. 12
5.
OPERATION .......................................................................................... 15
5.1
5.2
6.
STARTING THE COM.SAT IP BASIC ......................................................................... 15
SWITCH ON THE COM.SAT IP BASIC ....................................................................... 16
CONFIGURATION.................................................................................. 17
6.1 ACCESS TO THE WEB INTERFACE .......................................................................... 17
6.2 NETWORK PARAMETERS ..................................................................................... 19
6.2.1 Settings ................................................................................................... 19
6.2.1.1
6.2.1.2
6.2.1.3
6.2.1.4
6.2.1.5
6.2.1.6
6.2.1.7
6.2.1.8
6.2.1.9
6.2.1.10
6.2.1.11
6.2.1.12
6.2.1.13
6.2.1.14
6.2.1.15
6.2.1.16
6.2.2
LAN port ...................................................................................................... 20
UMTS channel 1/2 ....................................................................................... 20
WAN port ..................................................................................................... 20
PPPoE Parameter ......................................................................................... 21
Static IP parameter ..................................................................................... 21
DNS Setting ................................................................................................. 22
Dynamic DNS (DDNS): ................................................................................. 23
UDP NAT Keep Alive ..................................................................................... 23
TCP NAT Keep Alive ..................................................................................... 24
Router’s public IP address .......................................................................... 24
STUN ........................................................................................................... 25
Quality of Service (Layer 3) ........................................................................ 26
VLAN/CoS (Layer 2) .................................................................................... 26
SIP/ RTP Port ............................................................................................. 26
SIP-Timer.................................................................................................... 27
Enable Clone MAC Address .......................................................................... 27
Status ...................................................................................................... 28
6.2.2.1
6.2.2.2
6.2.2.3
LAN-Port ...................................................................................................... 28
GSM/UMTS Channel ..................................................................................... 28
WAN Port ..................................................................................................... 29
6.3 ISDN CONFIGURATION ..................................................................................... 31
6.3.1 BRI-NT mode ............................................................................................ 31
6.3.1.1
Configuration of the BRI-NT port ................................................................. 31
6.3.1.2
Incoming calls ............................................................................................. 34
6.3.1.2.1
Route Incoming Calls Port wise
34
6.3.1.2.2
Route Incoming Calls Channel Number wise
36
6.3.2
TE Mode ................................................................................................... 38
6.3.2.1
Configuration of the TE port......................................................................... 38
6.3.2.2
Outgoing Calls ............................................................................................. 40
6.3.2.3
Incoming Calls ............................................................................................. 42
6.3.2.3.1
43
6.3.2.3.2
44
6.3.2.3.3
Route Incoming Calls MSN Number wise
44
6.3.2.4
Routing Table Calling Number Based ........................................................... 45
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Copyright according DIN 34!
Manual
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6.3.2.5
Routing Table DDI Number Based ................................................................ 46
6.4 ISDN STATUS ................................................................................................. 48
6.5 GSM/UMTS KANAL 1/2 .................................................................................. 49
6.5.1 General Channel Parameters ....................................................................... 49
6.5.1.1
6.5.1.2
Prepaid Card ................................................................................................ 52
GSM/UMTS Network Selection ..................................................................... 52
6.5.2 Incoming Calls .......................................................................................... 53
6.5.3 Outgoing Calls .......................................................................................... 56
6.5.4 SMS......................................................................................................... 58
6.5.5 Network Diversion ..................................................................................... 60
6.5.6 Call Barring .............................................................................................. 61
6.6 GSM/UMTS CHANNEL STATUS .......................................................................... 62
6.7 GSM/UMTS CHANNEL DESTINATION NUMBER CALLING NUMBER BASED .................... 64
6.8 SIP-TRUNK 1/2/3/4 ...................................................................................... 66
6.8.1 General .................................................................................................... 66
6.8.1.1
6.8.1.2
6.8.1.3
6.8.1.4
6.8.1.5
6.8.1.6
6.8.1.7
6.8.1.8
6.8.1.9
6.8.1.10
6.8.1.11
Registrar Settings ........................................................................................ 66
Authentication ............................................................................................. 68
Outbound Proxy ........................................................................................... 68
Vocoder Preference ..................................................................................... 69
Use Symmetric RTP? .................................................................................... 69
Send Caller-ID?............................................................................................ 69
Source Port IP Address ................................................................................ 70
DTMF Option ................................................................................................ 70
FAX Options ................................................................................................. 70
Default Transport for Outgoing message .................................................... 71
Call Hold Method using ................................................................................ 71
6.8.2 Incoming Calls .......................................................................................... 72
6.8.3 Outgoing Calls .......................................................................................... 75
6.9 STATUS SIP TRUNK .......................................................................................... 75
6.10 DESTINATION NUMBER SIP TRUNK PORT - BASED ON CALLING NUMBER ................ 78
6.11 SIP PROXY .................................................................................................. 79
6.11.1
General ................................................................................................. 79
6.11.1.1
6.11.1.2
6.11.2
Registration Settings .................................................................................. 80
Codec Preference ........................................................................................ 81
Settings................................................................................................. 82
6.11.2.1
6.11.2.2
6.11.2.3
General Settings ......................................................................................... 82
Authentication of SIP messages .................................................................. 84
Outgoing Calls ............................................................................................. 84
6.11.3
Status ................................................................................................... 86
6.12 ROUTING TABLE ............................................................................................ 88
6.13 STATIC ROUTING........................................................................................... 90
6.14 PEER TO PEER CALLS...................................................................................... 93
6.15 PEER TO PEER ROUTING TABLE ........................................................................ 95
6.16 SYSTEM CALL DIVERSION ............................................................................... 97
6.17 EMERGENCY NUMBERS .................................................................................... 99
6.18 VIP USERS ................................................................................................ 100
6.19 VIRTUAL PBX (VPBX)................................................................................. 102
6.19.1
Call Hold .............................................................................................. 103
6.19.2
Call Retrieve ........................................................................................ 104
6.19.3
Call Transfer ........................................................................................ 104
6.20 LICENSING ................................................................................................. 105
6.21 VOICE MESSAGES ........................................................................................ 106
6.22 SMS NOTIFICATION .................................................................................... 107
6.22.1
Port Failure using SMS .......................................................................... 107
6.22.2
Update IP address using SMS ................................................................. 108
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6.22.3
Calling Number Transfer ........................................................................ 108
6.23 SYSTEM PARAMETERS ................................................................................... 110
6.23.1
General Parameters .............................................................................. 110
6.23.2
Date and Time Settings ......................................................................... 113
6.23.2.1 Real Time Clock ......................................................................................... 113
6.23.2.2 Daylight Saving Time (DST) ...................................................................... 114
6.23.2.2.1 Manual DST
115
6.23.2.2.2 Scheduled DST
115
6.23.2.2.3 Backward Time Adjustments
115
6.23.2.3 SNTP Settings ........................................................................................... 116
6.24 NETWORK CONNECTION ................................................................................ 117
6.25 CALL PROGRESS TONES ................................................................................ 119
6.26 CALL DETAILS RECORD (CDR) FILTERS ........................................................... 126
6.26.1
Call Details Record (CDR) Report ............................................................ 128
6.27 MAINTENANCE ............................................................................................ 130
6.27.1
System Debug...................................................................................... 130
6.27.1.1
6.27.1.2
6.27.1.3
Host Debug Parameters ............................................................................ 131
Host Debug Search Parameters ................................................................. 132
Globale ISDN-Parameter: .......................................................................... 132
6.27.2
PCAP Trace .......................................................................................... 133
6.27.3
Default the System ............................................................................... 135
6.27.4
Soft Restart ......................................................................................... 135
6.27.5
Password Change ................................................................................. 137
6.28 UPLOAD/DOWNLOAD ................................................................................... 139
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Figures
FIGURE 1: WALL MOUNTING ............................................................................................. 11
FIGURE 2: FUNCTIONAL GROUPS ........................................................................................ 13
FIGURE 3: FRONT OF THE COM.SAT IP BASIC .......................................................................... 13
FIGURE 4: TOP OF THE COM.SAT IP BASIC ............................................................................. 14
FIGURE 5: BACK OF THE COM.SAT IP BASIC ........................................................................... 14
FIGURE 6: FRONT OF THE COM.SAT IP BASIC .......................................................................... 15
FIGURE 7: LOGIN PAGE ................................................................................................... 17
FIGURE 8: START PAGE ................................................................................................... 18
FIGURE 9: NETWORK SETTINGS ......................................................................................... 19
FIGURE 10: DNS SETTINGS ............................................................................................. 22
FIGURE 11: STUN SETTINGS ............................................................................................ 25
FIGURE 12: STATUS OF THE NETWORK ................................................................................. 28
FIGURE 13: ISDN NT PORT SETTINGS ................................................................................ 31
FIGURE 14: NT - ROUTE INCOMING CALLS PORT WISE .............................................................. 34
FIGURE 15: NT – ROUTE INCOMING CALLS CHANNEL NUMBER WISE .............................................. 36
FIGURE 16: TE PORT SETTINGS ......................................................................................... 38
FIGURE 17: OUTGOING CALLS IN TE MODE ........................................................................... 40
FIGURE 18: INCOMING CALLS IN TE MODE ............................................................................ 42
FIGURE 19: ROUTING TABLE CALLING NUMBER BASED ............................................................. 45
FIGURE 20: ROUTING TABLE DDI NUMBER BASED .................................................................. 46
FIGURE 21: ISDN STATUS .............................................................................................. 48
FIGURE 22: GSM/UMTS CONFIGURATION............................................................................ 49
FIGURE 23: GSM/UMTS INCOMING CALLS ........................................................................... 53
FIGURE 24: GSM/UMTS OUTGOING CALLS ........................................................................... 56
FIGURE 25: SMS VIA GSM/UMTS..................................................................................... 58
FIGURE 26: NETWORK DIVERSION ...................................................................................... 60
FIGURE 27: CALL BARRING .............................................................................................. 61
FIGURE 28: STATUS OF BOTH GSM/UMTS CHANNELS ............................................................. 62
FIGURE 29: CLI TABLE ................................................................................................... 64
FIGURE 30: SIP TRUNK SETTINGS ..................................................................................... 66
FIGURE 31: SIP-TRUNK INCOMING/OUTGOING CALLS............................................................... 72
FIGURE 32: STATUS OF THE SIP TRUNKS.............................................................................. 75
FIGURE 33: SIP ROUTING FOR INCOMING CALLS ..................................................................... 78
FIGURE 34: GENERAL SETTINGS FOR SIP PROXY .................................................................... 79
FIGURE 35: SIP PROXY SETTINGS ...................................................................................... 82
FIGURE 36: SI EXTENSION STATUS .................................................................................... 86
FIGURE 37: ROUTING TABLE ............................................................................................ 88
FIGURE 38: STATIC ROUTING ........................................................................................... 90
FIGURE 39: STATIC ROUTING ........................................................................................... 91
FIGURE 40: PEER TO PEER CONNECTION .............................................................................. 93
FIGURE 41: ROUTING FOR PEER TO PEER.............................................................................. 95
FIGURE 42: CALL DIVERSION............................................................................................ 97
FIGURE 43: TABLE OF THE EMERGENCY NUMBERS .................................................................... 99
FIGURE 44: VIP USERS ................................................................................................. 101
FIGURE 45: VIRTUAL PBX ............................................................................................. 102
FIGURE 46: LICENSING ................................................................................................. 105
FIGURE 47: VOICE MESSAGES ........................................................................................ 106
FIGURE 48: SMS NOTIFICATION ...................................................................................... 107
FIGURE 49: DISPLAY OF THE SYSTEM PARAMETERS ................................................................ 110
FIGURE 50: DATE AND TIME SETTINGS .............................................................................. 113
FIGURE 51: DATE AND TIME SETTINGS .............................................................................. 114
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FIGURE 52: SNTP SETTINGS .......................................................................................... 116
FIGURE 53: NETWORK CONNECTION ................................................................................. 117
FIGURE 544: CALL PROGRESS TONES CONNECTION ............................................................... 119
FIGURE 55: FILTERS FOR THE CDR ................................................................................... 126
FIGURE 56: CLEAR CALL RECORDS ................................................................................... 127
FIGURE 57: CDR REPORT .............................................................................................. 128
FIGURE 58: CONNECTION DATA ....................................................................................... 129
FIGURE 59: SETTINGS FOR THE SYSTEM DEBUG ................................................................... 130
FIGURE 60: PCAP TRACE .............................................................................................. 133
FIGURE 61: DEFAULT SETTINGS ...................................................................................... 135
FIGURE 62: SOFT RESTART ............................................................................................ 136
FIGURE 63: PASSWORD CHANGE ..................................................................................... 137
FIGURE 64: UPLOAD / DOWNLOAD ................................................................................... 139
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Technical terms
In this manual the keywords ‘coM.sat IP Basic’, ‘System’ and ‘Gateway’ will be used as
synonyms.
Furthermore terms like ‘User’ and ‘You’ will be used as synonyms to the person, which installs
or configures the coM.sat IP Basic unit.
For ‘Caller’ and ‘calling subscriber’ are synonyms for ‘called’ and ‘calling subscriber’.
‘Port’ und ‘branch extension’ are also synonyms for ‘PBX’ and ‘PABX’.
‘Cellular Engines’ are ‘Modules’, ‘Mobile modules’ or ‘Cellular Modules’.
Some of the used terms in this manual are defined as following:
The Caller / called party is a person, which makes calls. Also user can make calls.
Called / called party is a person, which is called / gets the call. User can also be the called
party.
Source / generating branch connection:
branch connection from which the call is generating.
Target / destination connection: a connection, where the call is terminating.
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1. Introduction
coM.sat IP Basic is a gateway with a lot of features, which uses several technologies for voice
services like VoIP, UMTS and ISDN.
coM.sat IP Basic uses GSM/UMTS modules and SIM cards for voice and data communication as
well for SMS transmission. coM.sat IP Basic can be connected to an external or internal BRIport of an ISDN PABX. Furthermore it is possible to connect the unit with an interface for a
IP-PBX over one LAN-Port and an interface for an Enterprise Router/DSL-Modem via WANPort.
Every extension (also virtual extension) has an access (access rights configurable) to the
coM.sat IP Basic Gateway. Vice versa every PBX extension is reachable via mobile subscribers
with the best mobile tariff.
coM.sat IP Basic is built in a stable enclosure and is suitable for vertical or horizontal surfaces.
coM.sat IP Basic is comfortable and user-friendly configurable via a Web-Interface.
The main features, functions and interfaces of the coM.sat IP Basic are described in this
document.
Furthermore the manual provides information for an installation, usage of the Gateways and
diagnostics for malfunctions.
It is recommended to read the usage- and safety information of the unit, before it hast o be
used.
The subject to change, which improves the safety and usage of the unit, is a right of the
manufacturer.
For further technical issues please contact the hotline under +49(0)180-5-NEEDHELP
(+49(0)180-5-633343)1 .
Further information in German or English language is available over the following web site:
www.comsat.de
This information is available in the download area of the company.
1
2
DTMS - only 14 Cent per minute from the German fixed network
DTMS - only 14 Cent per minute from the German fixed network
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2. Security hints for the user
The following hints are related to the coM.sat IP Basic unit. The unit uses the Cellular
Engines/Modules of the company Sierra Wireless (MC8792V). In this case the safety
regulations and manual of this company must be considered.
Electrical safety
The coM.sat IP Basic unit is powered with 12V DC. Furthermore the unit can be connected to
an external port of an ISDN-PBX. There are no further safety arrangements due to high
voltages from the unit.
To avoid serious damages in the unit, it is strictly recommended that user has to discharge
himself before making any interventions.
WARNING
The unit is supplied with a Safety Extra-Low Voltage (SELV) power supply and a Limited
Power Source (LPS). Therefore only Safety Extra-Low Voltage with Limited Power Source
units have to be connected according to IEC 60950-1 / EN 60950-1 / VDE 0805-1.
Air traffic safety
The usage of Cellular-Engines in airplanes can influence the navigation systems and disturb
the radio network. Therefore, their use was prohibited by law.
It is prohibited to use coM.sat IP Basic units on board of any aircraft. The violation against this
allowance can temporarily or full suspension of the Cellular-Engine services and/or lead to
legal action against the offender.
Environment with explosive substances
coM.sat IP Basic is not approved for use in hazardous areas. It is therefore not recommended
to use the unit in the vicinity of such areas, such as gas stations, fuel depots, chemical plants
or with blasting operations. If that should be required, for the operation to ensure that no
hazard can occur.
Road traffic safety
If the units are installed in vehicles used on public roads, so are the rules for calling in
vehicles in the country in which the device is observed in each case.
Non ionizing radiation
As with any radio transmitting equipment, users are reminded that it is advisable to
satisfactory use of the devices and the safety of the operator to use the unit only in the
normal position.
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Medical equipments
The operation of radio stations, which include cellular engines, can affect the viability of not
properly protected medical devices. Please, check with the doctor or the manufacturer of the
medical device
Precautions in case of loss or theft
In case of loss of the Cellular Engines or the inserted SIM-card in the coM.sat IP Basic, the
network provider has to be informed immediately to avoid misuse.
Shipment
The factory offers the device packaging to transport sufficient protection against the bumps
and knocks and should therefore be preserved for future transport. To prevent condensation
or moisture should test the equipment before commissioning can slowly adjust to the ambient
temperature, if they had been stored in a colder or hotter environment
Where to install the units
The units should be installed away from direct sunlight and heat. This increases both the
reliability of the units and their operating life, as the components used are less thermally
stressed.
The units should be operated only with the supplied power devices or a genuine spare part.
The cables to the units should be installed so that no one consult a physical risk may. Power
cables should be routed separately from signal cables.
The installation of the equipment should also be done only by adequately trained personnel
Damage and repairs
With noticeable damage to the device or if the device has been exposed, further operation
should be avoided for safety reasons.
Repairs may only be made by the manufacturer or authorized by it. Should this not be
possible at a time, the repair is run by a qualified technician and only with original parts.
In any case, before any repair, the device is to be disconnected.
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3. Wall mounting of the coM.sat IP Basic
Select an appropriate location on the wall to install the coM.sat IP Basic.
Place the installation template on the wall and mark the hole on the wall.
Drill a hole of corresponding size.
Run the screw and tighten them. You can screw the head a few millimetres out from the
wall.
• Look at the strength of the bolts.
• Hang the coM.sat IP Basic on the wall.
•
•
•
•
Figure 1: Wall mounting
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4. Functional description
The coM.sat IP Basic combines the fixed network ISDN and IP technologies (networks and
terminals) among themselves and with the GSM / UMTS network.
The coM.sat IP Basic is available as a version with two UMTS channels.
Using the Sierra Wireless Cellular Engines (MC8792V) allows not only voice communication
but also data transmission and SMS from a PC. The USB interface provides direct access to
the cellular engines.
The main features are:
GSM/UMTS:
• TS11 Voice in Full Rate and Enhanced Full Rate, DTMF
• TS12 Emergency function
• TS21 SMS in Text and PDU mode
Supporting services:
• CLIP Caller ID of the caller to the called party participants
• CLIR Restriction of the calling number on outgoing
• CFU Call forward
• AOC Advise of Charging Information
• BAOC Bar all outgoing calls
• BOIC Bar all international outgoing calls
• BAIC Barred for all incoming calls
• COLP Caller ID of the calling subscriber at the called party
Hardware interfaces:
• USB Direct access, transparent mode and SMS-Server
• RJ45 WAN for VoIP services and programming
• RJ45 LAN for programming and VoIP
• RJ45 NT for connection to the ISDN PBX according ISO 8877
• RJ45 TE for the synchronisation port and to the ISDN network according ISO 8877
• SIM-card holder for small 1,8V and 3V SIM-cards
• SMA RF-connection for GSM/UMTS antenna
• Connector for DC power supply
• LED-display for operating status
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To realize the above features, a microprocessor circuit was developed that the interplay of
different functional groups of the coM.sat IP Basic controlled device.
These are shown in the sketch:
Figure 2: Functional groups
1
2
3
4
5
6
7
Figure 3: Front of the coM.sat IP Basic
On the front are located:
1
2
3
4
5
6
7
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Jack for the power supply
Socket for the NT port (RJ-45)
Socket connector for the TE or the synchronization (RJ-45)
Socket for USB interface (USB-B)
The reset button to reset the coM.sat IP Basic
Socket for LAN connection (RJ45)
Socket for WAN connection (RJ45)
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Figure 4: top of the coM.sat IP Basic
On the top are located the LED-status display.
1
2
3
4
Figure 5: Back of the coM.sat IP Basic
On the back are located:
1 SMA connection for the antenna of
2 SIM-card reader of the GSM/UMTS
3 SIM-card reader of the GSM/UMTS
4 SMA connection for the antenna of
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the GSM/UMTS channel 2
channel 2
channel 1
the GSM/UMTS channel 1
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5. Operation
5.1 Starting the coM.sat IP Basic
•
•
•
•
•
•
Connect the coM.sat IP Basic as shown below.
Set the coM.sat IP Basic according to a suitable place where a proper power supply is
possible.
Connect the antenna(s) with the coM.sat IP Basic.
Connect the Ethernet port for WAN / LAN IP of the coM.sat IP Basic with the switch /
network of the company.
Connect the ISDN line (if present) with the BRI NT / TE port of the IP coM.sat IP Basic.
Connect the 12V power supply to the coM.sat IP Basic.
GSM/
UMTS
ISDN
LAN
WAN/LAN
PSTN
/PBX
Figure 6: Front of the coM.sat IP Basic
•
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After successful configuration, insert the SIM card in the GSM / UMTS channel of
coM.sat IP Basic and restart the coM.sat IP Basic again, or restart the appropriate
module from the web interface
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5.2 Switch on the coM.sat IP Basic
After connecting the coM.sat IP Basic shown up procedure please switch on the power supply.
The status of each LED is described as follows:
Table 1: LED indications
LED
Status
Meaning
LAN
Off
Port is not connected
Green - on
Port is connected and identified
Green - flashing
Data transmission
Off
Port is not connected
Green - on
Port is connected and the router identified
Green - flashing
Data transmission
Off
No internet connection via WWAN
Green - on
Internet via 3G
WAN
WWAN (3G)
Active 1
Network 1
Power
Sync
Active 2
Network 2
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Green - flashing
Data transmission via 3G
Yellow - on
Internet via GPRS
Yellow - flashing
Data transmission via GPRS
Off
ISDN Layer 1 is up
Yellow - on
Active call on GSM/UMTS-Channel 1
Yellow - flashing rapidly
ISDN Layer 1 is down
Yellow – flashing slowly
Call setup on channel 2
Off
No module available
Green - on
Module 1 is registered in network
Green - flashing rapidly
SIM-Card is not inserted
Green - flashing slowly
Waiting for network registration
Off
Unit off
Green - on
Unit is on
Off
No Synchronisation
Red - on
Unit is synchronised with PBX / PSTN (TE-Mode) or PBX (NTMode)
Off
ISDN Layer 2 is up
Yellow - on
Active call on GSM/UMTS-Channel 2
Yellow - flashing rapidly
ISDN Layer 2 is down
Yellow - flashing slowly
Call setup on channel 2
Off
No module available
Green - on
Module 2 is registered in network
Green - flashing rapidly
SIM-Card is not inserted
Green - flashing slowly
Waiting for network registration
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6. Configuration
6.1 Access to the web interface
The user can program all parameters of the coM.sat IP Basic using the web interface. It may
be necessary that the user must change the port parameters of the network. This is required
if the IP coM.sat IP Basic as part of a network is established and is subsequently accessed on
the web interface to configure the rest of the system. To access the web interface, please
follow the steps below:
•
•
•
•
•
To access the web interface of the coM.sat IP Basic, the LAN port of the device are
connected to the PC. The PC can either be a stand-alone PC or a PC on a network.
Furthermore the coM.sat IP Basic and the PC in the same subnet. The IP addresses of
coM.sat IP Basic and the PC must be different.
The default IP address and subnet mask of the LAN ports of coM.sat IP Basic are
192.168.2.26 and 255.255.255.0.
Open your Web browser at coM.sat IP Basic connected PCs.
Enter the IP address of the coM.sat IP Basic in the address field of your web browser
The following login page will be opened:
Figure 7: Login Page
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Please, enter the default password ‘1234’ on this login page.
Your login was successful, if listed on the left side of the screen the appropriate links.
Figure 8: Start page
Now all parameters of the coM.sat IP Basic can be programmed by clicking on the appropriate
link to the specific property.
After switching on the coM.sat IP Basic, the user should program the parameters with the
given order:
•
•
•
•
•
•
•
•
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Port parameters of the network
Selection of the region and country code
Date and time settings
Time mode for time change
Mobile port parameters
Port parameters for SIP
Port parameters for the ISDN NT or TE mode
routing
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After successful programming of the coM.sat IP Basic, the user is able to take calls as well to
make calls.
Note: The connection of the system for access to the web interface can also be done via the
WAN port.
6.2
Network parameters
6.2.1
Settings
The network conditions can vary from customer to customer. The configuration of the
network parameters is an important step in configuring the coM.sat IP Basic. If the coM.sat IP
Basic is connected to the LAN, the IP address of the coM.sat IP Basic according to the
instructions for setting up the network address can be changed.
Figure 9: Network settings
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The various parameters are:
6.2.1.1
LAN port
•
IP address: In this edit field the user must enter the LAN port IP address of the unit.
The default IP address of the LAN port is 192.168.2.26.
•
Subnet mask: In this edit field the user must enter the subnet mask of the LAN port.
The default subnet mask of the LAN port is 255.255.255.0.
6.2.1.2
UMTS channel 1/2
The following parameters are programmed by the user as specified by the service provider.
•
•
•
•
•
Access point
Subscriber number
User name
Password
Fallback DNS-Server address
6.2.1.3
WAN port
•
Name: In this edit field a name can be assigned to the WAN port of the coM.sat IP
Basic. The name can contain a maximum of 12 characters. All ASCII characters are
allowed. By default the field is empty. This name is an identifier for the network port
and is not used in the coM.sat IP Basic.
•
Connection type: coM.sat IP Basic supports the following connection types:
o Static: Select 'Static', if the user's IP address, subnet mask and gateway address
manually assign wants. Obtain the necessary information from the network
administrator and enter it manually. As the default is 'Static' is selected
o DHCP: Select 'DHCP' if the user's IP address, subnet mask, DNS address and
gateway address assigned by the DHCP server automatically.
o PPPoE: Select 'PPPoE' if the user wants to assign automatically the IP address,
subnet mask, gateway address and the DSN address by the PPPoE server. If this
option is selected, the user must enter the user ID and password, which is from
the service provider in the PPPoE parameters provided.
Note: If the connection type from DHCP or PPPoE is changed to static, then the DNS
connection type should also be changed in static. If DHCP is selected as the connection
type, then the parameters PPPoE, Static IP and DNS settings are not programmable.
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6.2.1.4
coM.sat IP Basic
PPPoE Parameter
If the user has selected the ‘PPPoE’ as the connection type, then the parameters must be
programmed in the following way:
•
User ID: Enter the Service Provider provided PPPoE user ID manually in this edit field.
The user ID can be up to 16 characters long. In this edit field, all ASCII characters are
allowed. By default this edit field is empty.
•
Password: Enter the Service Provider provided PPPoE user password manually in this
edit field. The password can be up to 16 characters long. In this edit field, all ASCII
characters are allowed. By default, this edit field is empty.
•
Service Name: Enter the 'PPPoE Service Name', if this by your Internet Service
Provider (ISP) was made available. Otherwise, leave it blank. By default this edit field
is empty. The PPPoE service name can be up to 16 characters long. In this edit field, all
ASCII characters are allowed
6.2.1.5
Static IP parameter
If the user has selected the connection type ‘Static’, then the following parameters must be
programmed:
•
IP-Address: Enter in this field by the service provider provided 'Static IP Address'. The
default static IP address of the coM.sat IP Basic is 192.168.1.111.
•
Subnet-Mask: Enter in this edit field by the service provider provided 'Static IP
Address'. The default static IP address of the coM.sat IP Basic is 255.255.255.0.
•
Gateway: Enter in this edit field, the gateway address (if provided by your ISP). By
default, the gateway address is blank.
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6.2.1.6
coM.sat IP Basic
DNS Setting
Figure 10: DNS settings
DNS stands for ‘Domain Name Server’, which is used to locate the domain name in the IP
address. Select either ‘Static DNS’ or ‘Automatic DNS’ from.
By default the ‘Static DNS’ is selected.
•
Automatic: Select ‘Automatic DNS’ if the ISP connection type as DHCP or PPPoE
provides.
•
Static: If static DNS is selected, then the user should manually program the following
parameters:
•
DNS address: Enter the Service Provider provided DNS address manually in this field.
By default, the field is empty.
•
DNS Domain Name: Enter the DNS domain name in this edit field. This can be up to
40 characters long. All ASCII characters are allowed. By default the edit field is empty.
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6.2.1.7
coM.sat IP Basic
Dynamic DNS (DDNS):
Dynamic DNS (DDNS) is a service that translates internet domain names to IP addresses.
The DDNS service provider will provide the Host / Domain name for internet devices available
as well as the integrated DDNS client in the internet device. Here, the dynamic DNS client
that runs in the internet host, whenever the internet host a new IP address is assigned to
update its new IP address in Dynamic DNS server.
Once the IP address of the device is updated in the DNS server, any caller can access the IP
network by entering the host name / domain reach the device.
•
Enable dynamic DNS?: Select 'Yes' in this field to enable the dynamic DNS service,
otherwise select 'No'. To use the DDNS service from the DDNS service provider, i.e. by
DynDNS.org, please configure the following parameters:
•
User name: Program of the DynDNS.org provided user name in this edit field, when
the 'Dynamic DNS' is activated. The user name can be up to 40 characters long. All
ASCII characters are allowed. By default, the field 'User Name' is blank
•
Password: Program of the DynDNS.org provided password in this edit field, when the
'Dynamic DNS' is activated. The password can be up to 24 characters long. All ASCII
characters are allowed. By default, the field 'Password' is blank
•
Host name: Program of the DynDNS.org provided host name in this edit field, when
the 'Dynamic DNS' is activated. The host name can be up to 40 characters long. All
ASCII characters are allowed. By default, the field 'Host Name' is blank
By default, 'Enable Dynamic DNS?' option is set to 'No'.
6.2.1.8
UDP NAT Keep Alive
If the coM.sat IP Basic behind the NAT router is connected and the SIP messages over UDP
transport, then select 'UDP NAT Keep Alive' messages should be sent to update the UDP
connection in the NAT router. The 'UDP NAT Keep Alive' settings can be made as follows:
•
Enable UDP NAT Keep Alive?: Select 'Yes' if the coM.sat IP Basic is to send periodic
keep alive messages to update the UDP connection in the NAT router. As the default is
'No' selected.
•
Interval: Select the period after the coM.sat IP Basic should send keep alive messages.
This period should be shorter than the period of the router connection. The valid range
is 1 to 999 seconds. By default the value is 120 seconds.
•
Message type: Select the 'message type', which is sent to update the UDP connection
in the NAT router.
Select either 'entry' or 'notification'. By default 'notification' is selected.
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6.2.1.9
coM.sat IP Basic
TCP NAT Keep Alive
If the coM.sat IP Basic is connected behind the NAT router and the SIP messages has to be
transported over TCP, then the ‘TCP NAT Keep Alive’ messages should be sent to update the
TCP connection in the NAT router. The coM.sat IP Basic sends CR/LF (Carriage Return - Line
Feed) messages as TCP NAT Keep Alive message to update the TCP connection.
The TCP NAT keep alive settings can be made as follows:
•
Enable TCP NAT Keep Alive?: Select 'Yes' to enable the coM.sat IP Basic in a position
to send keep alive messages periodically to update the connection in the NAT router.
By default 'No' is selected.
•
Interval: Select the period after the coM.sat IP Basic should send Keep Alive Messages.
This period should be shorter than the period of the TCP connection of the router. The
valid range is 1 to 9999 seconds. By default the value is 120 seconds.
6.2.1.10 Router’s public IP address
The public IP address of the router is the IP address where coM.sat IP Basic can be achieved.
Enter the public IP address of the router if 'use public IP address of the router' option 'source
port IP address' is set in the SIP parameters. By default, this field is blank.
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6.2.1.11 STUN
Figure 11: STUN settings
STUN is necessary if the coM.sat IP Basic is behind the NAT router. The STUN server relieves
the passage of the most NATs, except symmetric NATs. Don’t program the STUN if the router
has a symmetric NAT.
•
Server’s address: Enter the STUN server address in this field. The server address can
be up to 40 characters long. All ASCII characters are allowed. By default, the STUN
server address is blank.
•
Server’s port: Enter the receiving port of the STUN server in this edit field. The valid
range of server port is between 1024 and 65535. By default the value is 3478.
•
Use SIP Port Fetched using STUN?: This parameter is only applicable if the field
'source port IP address' is set to 'IP address means retrieved using STUN'. If the
gateway is behind a NAT router and the 'will be forwarded SIP listening port on the
NAT router', please select 'No'. Otherwise, select 'Yes'. By default, this field is set to
'No'.
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6.2.1.12 Quality of Service (Layer 3)
•
SIP Diffserve/ToS: The coM.sat IP Basic will send all SIP messages using SIP QoS
setting. Program SIP DiffServe / ToS for SIP signalisation. The valid range for SIP
DiffServe / ToS is between 00 and 63. By default the value is 26.
•
RTP Diffserve/ToS: The coM.sat IP Basic will send all RTP packages using RTP QoS
setting. Program the RTP Diffserve/ ToS for the RTP traffic. The valid range for RTP
Diffserve /ToS is between 00 and 63. By default the value is 46.
6.2.1.13 VLAN/CoS (Layer 2)
• Enable VLAN/ CoS? : Select 'Yes' if the user wants the coM.sat IP Basic L2 VLAN
header to add to the packages is to leave the network port. The L2 VLAN header VLAN
ID and Class of Service (CoS) comprises bits. By default, this field is set to 'No'.
• VLAN ID: This is the virtual LAN identifier. Program the VLAN ID in this field. This ID is
added to the VLAN header, if the option 'Enable VLAN / CoS?' was set to 'Yes'. The
range of VLAN ID is from 0 to 4094. By default, the value is 1.
• SIP CoS: Program SIP CoS in this field. The coM.sat IP Basic will add the CoS bits to
the SIP packets leaving the network port if 'Enable VLAN / CoS?' is set to Yes. The
range of SIP CoS is from 0 to 7. By default, the value is 3.
• RTP CoS: Program RTP CoS in this field. The coM.sat IP Basic will add the CoS bits to
the RTP packets leaving the network port if 'Enable VLAN / CoS?' is set to Yes. The
range of RTP CoS is from 0 to 7. By default, the value is 6.
6.2.1.14 SIP/ RTP Port
• SIP UDP Port: The SIP UDP port defines the port, on which the coM.sat IP Basic listen
SIP messages, are transported over UDP. The range of the SIP UDP ports is from 1024
to 65535. By default, the value is 5060.
• SIP TCP Port: The SIP TCP port defines the port, on which the coM.sat IP Basic listen
SIP messages, are transported over TCP. The range of the SIP TCP port is from 1024 to
65535. By default, the value is 5060.
•
RTP listening port: The receiving RTP port defines the port where the IP coM.sat IP
Basic listens to the RTP packets. The system uses this port as source port sent in RTP
packets for RTP is to the remote peer. The range of the receiving RTP port is between
1024 and 65535. By default, the value is 8000.
•
Enable SIP over TCP? : By default, the option is enabled SIP over TCP to 'Yes'. This
means that coM.sat IP Basic listens to SIP messages over TCP. Select 'No' if you don’t
want that the coM.sat IP Basic sends and receives SIP messages over TCP.
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•
coM.sat IP Basic
Send ICMP Message?: By default, this field is set to 'Yes' to send the ICMP error
message whenever packets are received at the closed TCP / UDP ports. Select 'No' if
you do not want that the ICMP error message has to be sent on receipt of packets to
closed TCP / UDP ports.
6.2.1.15 SIP-Timer
coM.sat IP Basic supports the following three programmable SIP-Timer:
•
SIP INVITE Timer: The coM.sat IP Basic is waiting for a response from the called party
after the INVITE message has been sent for this timer. The range of the SIP INVITE
timer is between 10 and 180 seconds. By default the value is 30 seconds.
•
SIP Provisional Timer: The coM.sat IP Basic is waiting for the final response following
receipt of provisional response from the called party for this timer. The range of SIP
Provisional timer is between 10 and 180 seconds. By default the value is 60 seconds.
•
SIP General Request Timer: The coM.sat IP Basic is waiting for the response of a
transaction request for this timer. The range of the SIP General Request Timer is
between 10 and 60 seconds. By default the value is 20 seconds.
6.2.1.16 Enable Clone MAC Address
The MAC address is the address of the electronic hardware device that is programmed into
the device during manufacture.
This address is unmistakable and cannot be changed. The MAC address is assigned by the
IANA standard.
Nevertheless, it is possible to clone the MAC address if needed.
•
No: The default selection is 'No'. If 'No' is selected, the coM.sat IP Basic uses the unique
MAC address that has been programmed at the factory.
•
Yes: Select 'Yes' if you want to use the cloned MAC address. Enter the desired MAC
address manually in the appropriate field. There are allowed only hexadecimal
characters. By default, this field is blank.
Note: The coM.sat IP Basic restarts when you submit the page after changing the parameters
of the network ports.
After setting all parameters, click on the button 'Submit' to save the changes.
If the user wants to reset all parameters to their default values, click on the button 'Default
All'.
Note: After submitting this page, the gateway can be restarted.
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6.2.2
coM.sat IP Basic
Status
The status of various network parameters is shown as follows:
Figure 12: Status of the network
6.2.2.1
LAN-Port
•
IP Address: This field shows the LAN port currently assigned IP Address.
•
Subnet Mask: This field shows the LAN port currently assigned Subnet Mask.
•
MAC-Address: This field shows the LAN port currently assigned MAC address.
6.2.2.2
GSM/UMTS Channel
If the UMTS module for Internet services and for the SIP-Trunk/Extension registration is
used, the details of this connection will be displayed here. The GSM / UMTS channel that is
used for the WWAN connection has the following details:
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•
IP Address: This field displays the currently assigned IP address, which is the GSM /
UMTS channel (1 / 2) assigned when the Internet connection option for the GSM /
UMTS channel is used. (This field displays the IP address provided by the Service
Provider).
•
Subnet Mask: This field displays the currently assigned subnet mask, which is the
GSM / UMTS channel (1 / 2) assigned when the Internet connection option for the GSM
/ UMTS channel is used. (This field displays the subnet mask provided by the Service
Provider).
•
Gateway IP Address: This field displays the currently assigned IP address of the
gateway, which is the GSM / UMTS channel (1 / 2) assigned when the Internet
connection option for the GSM / UMTS channel is used. (This field displays the IP
address of the gateway provided by the Service Provider).
•
DNS Address: This field displays the currently assigned DNS address, which is the
GSM / UMTS channel (1 / 2) assigned when the Internet connection option for the GSM
/ UMTS channel is used. (This field displays the DNS address provided by the Service
Provider).
6.2.2.3
WAN Port
•
Status: Indicates which interface used on the Internet.
•
IP Address: This field shows the WAN port currently assigned IP address.
•
Subnet Mask: This field shows the WAN port currently assigned Subnet Mask.
•
Gateway IP Address: This field shows the WAN port currently assigned Gateway IP
address.
•
System Unique MAC Address: This filed shows the unique MAC address of the
coM.sat IP Basic.
•
System Clone MAC Address: This field shows the MAC address, which has to be
cloned.
•
DNS Address: This field shows the DNS address of the WAN port.
•
NAT Type: This field shows the NAT type, when STUN is enable in the coM.sat IP Basic.
The mostly used NAT types are:
o
o
o
o
o
o
o
o
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Unknown
Open
Cone NAT
Restricted NAT
Port restricted NAT
Symmetric NAT
Symmetric Firewall
Blocked
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•
Routers Public IP Address: This field displays the public IP address of the router,
which is programmed in the network port parameters.
•
IP Address fetched using STUN: This field displays the IP address, which is fetched
by the STUN, if the STUN Server Address has been programmed.
•
Dynamic DNS Status: This field shows the response that was received by the DDNS
server during query 'update IP address' to the server. The different answers that may
appear in this field are listed as follows:
•
Stack Status: This field shows strings like Idle, DHCP Response wait, PPPoE response
wait, NAT checking response wait, construct and Error.
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Cause of error
Description
Sending the message
failed
Is displayed when a failed registration
request sent to the Registrar.
Creating the Registry
Client failed
Displayed when SIP stack has a memory /
resource limit or the number of registered
SIP clients in the stack exceeds the
programmed.
Separation of the Register
Client failed
Displayed when SIP stack has a memory /
resource limit or the number of registered
SIP clients in the stack exceeds the
programmed.
Sending the request failed
Displayed when the DNS server is not
programmed.
Local Error
Displayed when the DNS request is failed.
Respond Timeout
Displayed when the timer for general
request expired.
Error message 4xx to 6xx
This is the ERROR CODE from 4xx to 6xx
No contact header in 2xx
Displayed when no contact address has
been received by the SIP server in 2xx
response.
Authentication failed
Displayed when the SIP server does not
authenticate the client.
STUN address not
programmed
Displayed when STUN enabled, but the
address is not configured.
STUN request failed
Displayed when the request to the STUN
server failed.
Outgoing address is not
programmed
Displayed when outgoing is enabled but the
outgoing address is not configured.
Routers IP address is not
programmed
Displayed when routers IP address should
be used in the transmission but is not
programmed.
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6.3 ISDN Configuration
The coM.sat IP Basic supports with its two ISDN ports the BRI-NT and the BRI-TE mode.
6.3.1
BRI-NT mode
6.3.1.1
Configuration of the BRI-NT port
Figure 13: ISDN NT Port Settings
To the BRI-NT port an ISDN phone or a PBX can be connected.
The following parameters must be programmed in BRI-NT mode:
•
Port Enable: User can enable or disable the BRI-NT port according to his needs.
Disabling the BRI-NT port is necessary if no calls should be made. The coM.sat IP Basic
route calls only if the BRI-NT port is enabled. If this port is deactivated, the caller gets
an error tone when he tries to access it.
•
Name: can be freely assigned
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coM.sat IP Basic
Progress Indicator: The Progress indicator influences the value of an ISDN
information element. The user can activate the following options:
o If the progress indicator causes a problem in the PBX, it is possible to deactivate
it with the first value 'send no Progress Indicator'.
o The second value ‘Inband Information now Available’ shows that in the voice
channel an announcement or a tone can be heard. This is the usual value.
o The third value ‘Call is not end – end ISDN’. It informs the connected PBX that
the call uses another network, not the ISDN network.
•
Line mode: The BRI-NT operates in two modes:
o Point to Multipoint (PMP): At the point to multipoint connection (multi-terminal),
multiple terminal equipments (up to 8) can be connected to the bus. Of these 8
connected terminal equipments a maximum of two calls can be made at the
same time.
o Point to Point (PP): Only two participants can be connected in the point-to-point
connection. This is the most common connection type with a PBX.
•
Power Feed Control: In NT mode the coM.sat IP Basic provides the terminals
equipments with a separate power supply. Thus the connected terminal equipments
needn’t an own power supply.
Note: The number of terminals can be connected to the BRI port in NT mode depends
on the power consumption of individual terminal equipments. The maximum current
per unit must not exceed 50mA. Because of the signalisation maximum of 8 devices
can be connected to the BRI port.
•
Calling Party Type of Number (TON): In this field the user should enter the type of
number of the calling party number (TON). The various options are:
o
o
o
o
o
o
o
Unknown
International Number
National Number
n/w specific Number
Subscriber Number
Shortened Number
Reserved Number
By default the TON calling party number is ‘Unknown’.
•
Calling Party Number Plan Identification (NPI): In this field the user should enter
the information to numbering plan of the caller (NPI). The various options are:
o
o
o
o
o
o
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Unknown
ISDN Number
Data Number
Telex Number
National Number
Private
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o Reserved
By default the NPI of the calling party number is ‘ISDN Number’.
•
Called Party Type of Number (TON): In this field the user should enter the called
party number type (TON). The various options are:
o
o
o
o
o
o
o
Unknown
National Number
n/w specific Number
Subscriber Number
Shortened Number
Reserved Number
•
Called Party Number Plan Identification (NPI): The user should program in this
field the numbering plan of the called party (NPI). The various options are:
o
o
o
o
o
o
o
Unknown
ISDN Number
Data Number
Telex Number
National Number
Private
Reserved
•
Clock Synchronization Port: This makes it possible to select the source of the
synchronization clock. With the option 'TE port' the external clock source is connect to
the TE port of the coM.sat IP Basic.
With the option 'NT port', the synchronization signal for the coM.sat IP Basic is
generated from the from the NT port. In this case the PBX must be clock master in
Layer 1and the coM.sat IP Basic clock slave. Thus, the PBX generates the clock for the
coM.sat IP Basic.
•
PBX Number: The registration of the PBX number will depend on the PBX. Some PBX
does not require a registration in this point. Other PBX requires the specification of the
root number as a prefix to the extension number.
•
Route incoming calls: The user is able to route incoming calls at the NT port either
Port wise or Channel Number wise.
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6.3.1.2
coM.sat IP Basic
Incoming calls
6.3.1.2.1
Figure 14: NT - route incoming calls port wise
•
Dial Timeout: At an incoming call on the NT port the caller gets a dial tone. The dial
tone will be issued for the duration of the dial timeout. During this time the caller can
reach any extension by dial DTMF digits. After entering a digit the dial timeout restarts.
By default, the time out is set to 7 seconds.
•
Route the call: Once the user has dialled the whole number, the coM.sat IP Basic will
check the following options to route the call:
o If the dialled digits with the programmed value in the 'route the call on receipt of
digits ...' match
o If the dialled digits with ‘#’ or ‘*’ match as programmed
o After expire of the time out, which is programmed as ‘after … seconds of last
digit dialled’.
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Destination Route Selection: coM.sat IP Basic supports the following methods to
route the call:
o As per routing table: For the option ‘As per routing table’ please refer to the
topic ‘Routing Table’.
o Fixed: Fixed routing option gives the user a choice to set three different port
types for routing the call irrespective of the destination number
received/collected on the source port. The user can set the required preference
from the drop-down list. If ports/channels programmed in the first option is
busy, second preference would be selected and so on. The list of port(s) options
are:
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
None
GSM/UMTS Channel 1
GSM/UMTS Channel 2
Any GSM/UMTS Channel (First Free)
Any GSM/UMTS Channel (Alternating)
TE Port (Ch-1)
TE Port (Ch-2)
TE Port (Ch-1,2) (First Free)
TE Port (Ch-1,2) (Alternating)
NT Port (Ch-1)
NT Port (Ch-2)
NT Port (Ch-1,2) (First Free)
NT Port (Ch-1,2) (Alternating)
SIP Trunk 1
SIP Trunk 2
SIP Trunk 3
SIP Trunk 4
Any SIP Trunk (First Free)
Any SIP Trunk (Alternating)
First Free: If there are multiple ports of the same type, the port which is free
first would be used for routing the call each time.
Alternating: If there are multiple ports of the same type, then the first call
would be routed through the first channel and the subsequent call through the
second channel and so on.
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6.3.1.2.2
coM.sat IP Basic
Figure 15: NT – route incoming calls channel number wise
Each ISDN port has two B channels, where two calls can be routed separately.
For both channels 1 and 2 the user must program the following parameters:
•
Dial Timeout: At an incoming call on the NT port the caller gets a dial tone. The dial
tone will be issued for the duration of the dial timeout. During this time the caller can
reach any extension by dial DTMF digits. After entering a digit the dial timeout restarts.
By default, the time out is set to 7 seconds.
•
Route the call: Once the user has dialled the whole number, the coM.sat IP Basic will
check the following options to route the call:
o If the dialled digits with the programmed value in the 'route the call on receipt of
digits ...' match
o If the dialled digits with ‘#’ or ‘*’ match as programmed
o After expire of the time out, which is programmed as ‘after … seconds of last
digit dialled’.
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coM.sat IP Basic
Destination Route Selection: coM.sat IP Basic supports the following methods to
route the call:
o As per routing table: For the option ‘As per routing table’ please refer to the
o Fixed: Fixed routing option gives the user a choice to set three different port
types for routing the call irrespective of the destination number
received/collected on the source port. The user can set the required preference
from the drop-down list. If ports/channels programmed in the first option is
busy, second preference would be selected and so on. The list of port(s) options
are:
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
None
GSM/UMTS Channel 1
GSM/UMTS Channel 2
TE Port (Ch-1)
TE Port (Ch-2)
NT Port (Ch-1)
NT Port (Ch-2)
SIP Trunk 1
SIP Trunk 2
SIP Trunk 3
SIP Trunk 4
•
Apply Allowed-Denied Number Logic: With the option ‘Apply Allowed-Denied
Number Logic’ dialled numbers or number blocks can be allowed or denied at a certain
port.
The user should enter the allowed and denied numbers in different lists (whitelist,
blacklist).
The function 'Apply Allowed-Denied Number Logic' is not applicable in the following
cases:
o destination number is consistent with an emergency number
o For pre-programmed destination numbers
o For the method destination number via CLI based destination number recorded
By default this option is disabled.
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6.3.2
coM.sat IP Basic
TE Mode
6.3.2.1
Configuration of the TE port
Figure 16: TE port settings
•
Port Enable: The user can enable or disable the TE port according to his needs.
Disable the TE port is required if calls are barred on this port. The coM.sat IP Basic will
route calls only if the TE port is enabled. If this is deactivated, the caller gets an error
tone when trying to access.
•
Name:
•
Enable CLIP on TE: The incoming caller ID will be displayed, otherwise the registered
MSN.
•
Line Mode: The BRI-TE operates in two modes:
o Point to Multipoint (PMP): At the point to multipoint connection (multi-terminal),
multiple terminal equipments (up to 8) can be connected to the bus. Of these 8
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connected terminal equipments a maximum of two calls can be made at the
same time.
o Point to Point (PP): Only two participants can be connected in the point-to-point
connection. This is the most common connection type with a PBX.
•
Overlap Receiving Timer: During the dial of digits the time will wait for the last digit.
If the timer has expired, the call will be setup.
The time out is between 1 and 255 seconds. By default the time out is 15 seconds.
•
Calling Party Type of Number (TON): In this field the user should enter the type of
number of the calling party number (TON). The various options are:
o
o
o
o
o
o
o
Unknown
National Number
n/w specific Number
Subscriber Number
Shortened Number
Reserved Number
•
o
o
o
o
o
o
o
Unknown
ISDN Number
Data Number
Telex Number
National Number
Private
Reserved
•
MSN number: In TE mode connected devices can be assigned to multiple MSNs. For
incoming calls, the device responds only to the set MSNs.
One MSN should consist of a maximum of 24 digits. Only the digits 0-9, * and # are
allowed.
Contains the edit field for the MSN no characters, this is allowed for all incoming calls.
For each edit field only one MSN is possible.
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6.3.2.2
coM.sat IP Basic
Outgoing Calls
Figure 17: Outgoing calls in TE mode
•
Allow Outgoing Calls: The TE port can also be used for outgoing calls. This is only
possible if this option is enabled.
By default this option is enabled.
•
Apply Reverse DDI: For outgoing calls, either the extension number or the incoming
caller ID can be transmitted. If the extension number should be displayed, the option
'Reverse DDI' must be activated.
By default this option is deactivated.
•
Apply Automatic Number Translation (ANT) Logic: In certain cases, the number
which has been dialled could not be understood by the ISDN network. For this reason,
this number will be replaced by another.
•
o Unknown
o ISDN Number
o Data Number
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o
o
o
o
coM.sat IP Basic
Telex Number
National Number
Private
Reserved
The user should program both:
o Number string of the dialled number: contains the dialled number (possible with
area code).
o Replace number string replace: contains the corresponding number, which is to
be replaced.
By default this option is deactivated.
Note: Empty numbers cannot be replaced.
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6.3.2.3
coM.sat IP Basic
Incoming Calls
Figure 18: Incoming calls in TE mode
•
Route Incoming Calls: The user can set the incoming calls for the TE port as follow:
o Port wise
o Channel number wise
o MSN number wise
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6.3.2.3.1
coM.sat IP Basic
The option ‘Allow Incoming Calls’ must be enabled to program the routing of incoming calls.
The routing mechanism enfolded two steps:
•
•
Selection of the destination number
Selection of the destination route
By default this field is activated.
•
Destination Route Selection: The coM.sat IP Basic supports the following methods to
route the calls:
o As per Routing Table: For the option ‘As per Routing Table’ please refer to the
o Fixed: The fixed routing option allows the user to route the call using three
several port types.
The user has the possibility to select the routing destination out of a drop down
list. If the port or channel which is programmed for the first routing destination
is busy, then the second routing destination will be taken and so on. The
selection of the routing destination is listed as follow:
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
None
GSM/UMTS Channel 1
GSM/UMTS Channel 2
TE Port (Ch-1)
TE Port (Ch-2)
NT Port (Ch-1)
NT Port (Ch-2)
SIP Trunk 1
SIP Trunk 2
SIP Trunk 3
SIP Trunk 4
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6.3.2.3.2
coM.sat IP Basic
Each ISDN port has two B channels to route two separated calls at the same time. For both
channels 1 and 2 the user must program the following parameters.
•
•
6.3.2.3.3
Route Incoming Calls MSN Number wise
coM.sat IP Basic gives the user the possibility to route the calls according to the MSN.
The option ‘Allow Incoming Calls’ must be enabled to program the routing of incoming calls.
•
•
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6.3.2.4
coM.sat IP Basic
Routing Table Calling Number Based
Figure 19: Routing Table Calling Number Based
In the routing table, the user can enter a destination number, depending on the calling
number. The single table parameters are:
•
Index: In this field shows the index number (001 to 500). The index cannot be
changed.
•
Calling Number: In this edit field the user can enter the number of the calling party
number.
By default the first entry is set to ‘No Match Found’. All other edit fields are blank.
•
Destination Number: In this edit field the user can enter the destination number. The
maximum characters of this number are 24. Valid characters are 0 to 9, *, # and the
dot (.).
Note: If the destination number is a IP Address, then the call will be routed via the
programmed SIP Trunk Group.
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6.3.2.5
coM.sat IP Basic
Routing Table DDI Number Based
Figure 20: Routing Table DDI Number Based
DDI is an ISDN service, which allows the caller to dial directly to an extension of an ISDN PBX
or a private network without switching.
With the help of this table, a call to a specific destination number, depending on the selected
DDI routed.
The user has the possibility to create the table automatically using the 'DDI Number
Generation' button. This can also be created manually.
If desired by the user that the system should create the 'DDI number based' table, then the
following parameters have to be programmed:
•
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All DDI Numbers: The user should enter the total number of DDI, which have been
assigned by the service provider as when assigned by the service provider e.g. if the
service provider has assigned the numbers from 60719100 to 60719199, then the total
number of the DDI entries are 100.
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•
Start Index Number: This is the reference number for the table 'DDI Number Based',
which is created on the basis of the entire table.
•
Start DDI Number: The user should enter the first DDI, which has been assigned by
the service provider.
•
Start Destination Number: The starting target number should be considered here.
These numbers are set by the user
•
Apply Reverse DDI: This option can be enabled if the user desires that each DDI
number is displayed for outgoing calls as CLI.
After the user has set the parameters according to the above requirements, please press the
button 'Apply' to create the table. If the user doesn’t want to save the changes or not create
the table, the button 'Cancel' has to be pressed. If the above option is used, then the ' DDI
Number Based' table generated automatically. The user can also make manual changes, or
create the entire table manually.
The user should then program the following parameters of the table 'DDI Number Based':
•
changed.
•
DDI Number: The user should enter the DDI number, which has been assigned by the
service provider. The maximum characters of the DDI number are 24. Allowed
characters are only from 0 to 9.
•
Destination Number: The user should program in this field the destination number
for each DDI number. The destination number may be up to 24 characters. These are
the characters 0 through 9, #, *, and the dot (.).
•
Apply Reverse DDI: This option can be enabled if the user desires that each DDI
number is displayed for outgoing calls as CLI.
For each DDI, an associated destination number is specified. Thus, the incoming calls
arrive directly at the desired destination.
The DDI table allows the user to program 1000 entries.
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6.4 ISDN Status
The ISDN Status displays the status of the Layers for the NT and TE port.
Figure 21: ISDN Status
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6.5
coM.sat IP Basic
GSM/UMTS Kanal 1/2
The coM.sat IP Basic supports 2 GSM/UMTS Channels. The GSM / UMTS channels provide the
flexibility to route calls efficiently and cost effectively according to user requirements in
mobile networks.
6.5.1
General Channel Parameters
The general channel parameters are declared below:
Figure 22: GSM/UMTS Configuration
•
GSM/UMTS Channel: This option is used to activate or deactivate the channel. If the
channel is deactivated, all incoming and outgoing calls are barred.
By default this field is deactivated.
Note: When the GSM / UMTS channel is activated and 'allow incoming calls' function
and 'allow outgoing calls is deactivated', then the channel for GSM / UMTS call is also
barred.
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•
Name: The user can assign the GSM / UMTS channel a name. This can be a maximum
of 12 characters. The ASCII characters such as 0-9, A...Z, a...z, %, ', *, -, + and the
point(.) are allowed in this edit field. By default, the edit field is blank.
•
SIM PIN: The SIM PIN is a security feature of the SIM card, which is used in the
device to protect the data stored on the SIM card from misuse. The SIM PIN must
contain at least four digits and a maximum 8 digits. If the SIM card uses access
without PIN, then the edit field remains blank. By default, the edit field is blank.
If the PIN is activated every time the coM.sat IP Basic is switched on, the PIN will be
requested. If an incorrect PIN is entered, you get the message 'SIM PIN wrong'.
Enter the wrong PIN three times or rebooting three times the unit with the wrong
programmed PIN, the card is locked. After the message 'SIM PUK' will be displayed. To
unlock the SIM card, the PUK that came with the SIM card must be entered. If the
wrong PUK is entered ten times, the card is useless.
•
Microphone Gain (Tx): The GSM / UMTS channel offers the possibility to adjust the
transmit volume. The following level settings are possible:
o
o
o
o
o
Very low
Low
Normal
High
Very high
By default the level is ‘Normal’.
•
Speaker Volume (Rx): The GSM / UMTS channel offers the possibility to adjust the
receive volume. The following level settings are possible:
o
o
o
o
o
Very low
Low
Normal
High
Very high
By default the level is ‘Normal’.
•
Access/Band Selection: Here the user can select the GSM band as well the access to
the module. The following options are possible:
o Only GSM Network
o All Networks allowed
o Direct USB access
Register only in the GSM network
Register in GSM or UMTS
transparent connection to the GSM module
By default the access/band is ‘All Networks allowed’.
•
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SMS Service Centre Number: The number of 'SMS Service Centre' is entered here.
This number is provided by the network provider and is depending on the use of SIM
card. The SMS Service Centre number may also be selected from a list of all known
network providers.
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•
AOC Timer: If the GSM / UMTS network cannot provide any real terrifying, an artificial
will be generated. This is determined by the 'AOC Timer'. The timing intervals are 0000
- 9999 seconds. By default the value is 60 seconds.
•
Enable CLIR: Allows the suppression of the caller ID at outgoing calls.
•
Enable CLIP: Allows the presentation of the CLIP of an incoming call.
•
Allow Roaming: If ‘allow Roaming’ is activated, the unit is able to register in foreign
networks. By default this field is deactivated.
•
Ignore Incoming SMS: This option determines the response to incoming SMS. In
some cases many SMS for advertising purposes are sent from service providers. To
burdened unnecessary coM.sat IP Basic with the processing of meaningless text, it is
recommended not to respond to SMS at all. If this option is activated the SMS will not
be processed by the GSM / UMTS module. By default the field is disabled.
•
Delete All Incoming SMS after processing: If this option is activated all incoming
SMS will be deleted after have been processed.
By default the field is disabled.
•
Direct Extension: The option 'Direct Extension’ is a special function for certain GSM /
UMTS networks, not for all. In this mode, all numbers called via GSM / UMTS get '#'
and the calling MSN is appended. This allows the called party to see the extension of
the caller.
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6.5.1.1
coM.sat IP Basic
Prepaid Card
Prepaid cards are used in the respective channels, so the settings for the balance are set as
follows:
•
Enter prepaid minute to set: The user defines the total number of minutes available
on this channel. By default the field is blank.
•
Remaining Minutes: This displays the remaining minutes. These are:
Remaining minutes = set minutes
•
– elapsed minutes.
If Prepaid Minutes reaches below: When prepaid minutes fall below the minimum
balance a message will be sent to the number, which is set in the field 'send SMS to'.
Any text message can be programmed by the user, for example ‘minimum balance to
GSM / UMTS channel [x] achieved’.
6.5.1.2
GSM/UMTS Network Selection
The coM.sat IP Basic offers the following two options to select the network:
•
Automatic: This option allows the GSM/UMTS module to make a free (automatical)
network selection.
•
Manual: This option allows the GSM / UMTS module in each channel to make a manual
network selection. This can be done by selecting the desired network from the priority
table option. Then, the button 'Select' are operated. To remove a network from the
'Priority Selected' list needs to press the 'Remove' button.
•
Net Lock: The user can block the selected networks with the option 'Net Lock'.
•
Change Password: The net lock password can be changed from the user by clicking the
option 'Change Password'. The default password is 12345678
By default this field is set to ‘Automatic’.
•
Copy Parameters to GSM/UMTS Channel 2: The user can activate this option, if the
same parameters should be set in channel 2.
Note: For the GSM/UMTS channel 2 exist the same copy option, if the user want to
copy all parameters for channel 1. The following parameters will not be copied:
o SMS Service Centre Number
o GSM/UMTS Network Selection
If all parameters are programmed, the user should press the ‘Submit’ button. To set all
parameters to default, the user should press the button ‘Default All’.
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6.5.2
coM.sat IP Basic
Incoming Calls
The configuration for incoming calls from the GSM/UMTS network will be set via the menu
selection ‘Incoming Call’.
Figure 23: GSM/UMTS incoming calls
To program the routing of the incoming calls to the appropriate user requirements, the
selection 'incoming calls' has to be enabled. By default the field is activated.
• Selection of the destination number
• Selection of the destination route
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•
coM.sat IP Basic
Destination Number Selection: The selection of the destination number is only for
incoming calls available. coM.sat IP Basic supports three different to determine the
destination number:
o Number selection using CLI table: If the selection of the destination number
‘Number selection using CLI table’ has been programmed, the unit checks the
GSM/UMTS channel target number based on calling number (refer to the topic
‘GSM/UMTS channel destination number based on calling number).
If no match found in the CLI table, the user has two possibilities to determine
the destination number:
Fixed destination number to use: The user has to enter the destination
number in this edit field. The number may be up to 24 characters. These
are the characters 0 through 9, #, *, and the dot (.).
By default this edit field is blank.
Answering the call with dial tone: At an incoming call, the unit
answers the call with a dial tone. The caller is now able to dial any number
per DTMF. If the caller doesn’t dial any number within the timeout, the
unit can either disconnect the call or route it to the fixed destination
number. The fixed destination number is defined in the edit field ‘Fixed
destination number to use’. By default the timeout is 7 seconds.
If the option ‘Answering the call with dial tone’ is programmed, then the unit will
recognize the end of dialling under the following criteria:
If the maximum dialled digits meets the value in ‘Route the Call on receipt
of <> digits’.
If the dialled digits meets the programmed characters ‘#’ or ‘*’.
After the timer expired in ‘after <> seconds of last digit dialled’.
•
Incoming call without CLIP should be: If calls has been received without CLIP, the
user can program the following options:
o Reject
o Route to a fixed destination number
o Answer the call and route it to the dialled destination number
•
Destination Route Selection: The coM.sat IP Basic supports the following methods to
route the calls:
o As per Routing Table: For the option ‘As per Routing Table’ please refer to the
o Fixed: The fixed routing option allows the user to route the call using three
several port types.
The user has the possibility to select the routing destination out of a drop down
list. If the port or channel which is programmed for the first routing destination
is busy, then the second routing destination will be taken and so on. The
selection of the routing destination is listed as follow:
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•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
•
None
GSM/UMTS Channel 1
GSM/UMTS Channel 2
TE Port (Ch-1)
TE Port (Ch-2)
NT Port (Ch-1)
NT Port (Ch-2)
SIP Trunk 1
SIP Trunk 2
SIP Trunk 3
SIP Trunk 4
•
Reject Incoming Calls from Callers: This option allows the user to program the list
of the caller, which have to be rejected.
The whole numbers can be programmed in the table ‘Caller Number’.
•
Apply Allowed-Denied Number Logic: With the option ‘Apply Allowed-Denied
Number Logic’ dialled numbers or number blocks can be allowed or denied at a certain
port.
The user should enter the allowed and denied numbers in different lists (white list,
black list).
The function 'Apply Allowed-Denied Number Logic' is not applicable in the following
cases:
o For the method destination number via CLI based destination number recorded
•
Copy Parameters to GSM/UMTS Channel 2: The user can enable this option, if he
wants to copy the same parameters in channel 2.
Note: For the GSM/UMTS Channel 2 the same copy option is available, if the user
wants to copy all parameters for channel 1.
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6.5.3
coM.sat IP Basic
Outgoing Calls
The user can make outgoing calls using the GSM/UMTS channel.
To use this possibility the option ‘Allow Outgoing Calls’ must be enabled.
Figure 24: GSM/UMTS outgoing calls
•
Return Call Handling (RCOC): When an outgoing call is placed from this channel to
any number and the called party is busy or not responding, the number of the caller is
stored internally in the memory of the device in the form of a RCOC table. If the called
party makes the call based on the received CLI and the option RCOC is enabled, then
the incoming call is routed to the before called party.
•
Allow following Callers to make Calls through this port: If the user is using that
channel for outgoing calls, and can select only certain numbers, then this option can be
activated. The numbers, which may be selected in the table ‘Caller Number’, is
entered. All other numbers are rejected. By default the field is disabled.
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•
Numbers Free of Charge: The user can enter the table 'call number' the number,
which is free of charge, such as the numbers that begin with the prefix 0800. These
numbers are free of charge (Advice of Charge). By default the field is blank.
•
Apply Automatic Number Translation (ANT) logic: In certain cases, the number
dialled by the ISDN network is not understood. For this reason, this number will be
replaced by another.
•
Note: Empty numbers cannot be replaced with any number string.
•
Net Access Numbers: The user can define in this list box the net access numbers.
That means here are the prefix code numbers of operators, with which calls are
generated. The registration of these numbers is as followed described:
0170
0171
For example, if only these two numbers have to be used by Deutsche Telekom.
It should be noted that only one prefix code may be entered per line.
Note: It is recommended that only make entries, when in fact restrictions will be
made. Otherwise, it is recommended to leave this list field empty.
•
Numbers always with CLIP: The list of 'Calling Number' contains numbers, which
transmit the number to your display (CLIP). Even if the option CLIR for this GSM /
UMTS channel is activated. For VIP users the CLIR function is not applicable.
•
Copy Parameters to GSM/UMTS Channel 2: The user can enable this option, if he
wants to copy the same parameters in channel 2.
Note: For the GSM/UMTS Channel 2 the same copy option is available, if the user
wants to copy all parameters for channel 1.
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6.5.4 SMS
The coM.sat IP Basic can send and receive short messages (SMS).
Figure 25: SMS via GSM/UMTS
The SMS option for the GSM / UMTS channel is set here. For the sending and receiving two
input fields are available.
Note: SMS-Messages have a length of maximum 160 characters.
•
Received SMS: The user can display all on the SIM card stored messages by pressing
the button 'Query'. However, in this field, only the following information is displayed:
o
o
o
o
•
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Date of the received text message
Time of the received text message
the content of the received SMS will appear in the 'text messages'
The number of the sender is in the 'phone number' is displayed.
Phone Number: Displays the number of the sender. When to send a text message
when the recipient's number is entered.
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•
Message Text: The message text of the received short message is displayed here.
When to send a SMS the text message and the destination number in field 'Phone
Number' is entered. To send the SMS the button 'Send' must be pressed.
•
Delivery Reports: When an SMS was sent and delivered to the recipient, a report is
sent to the sender. However, in this field, only the following information is displayed:
o date of delivery of the short message
o time of delivery of the short message
o The number to which the short message was delivered
To delete a short message the button 'Delete' must be pressed. To delete all short
messages the button 'Delete All' must be pressed. In this case all stored in 'Received
SMS’ as well the 'Delivery Reports' are deleted.
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6.5.5
coM.sat IP Basic
Network Diversion
Figure 26: Network diversion
At a call forward the network manages the call to another number.
The different redirection options are described as follows:
•
•
•
•
•
•
•
unconditional (always)
When Busy
No answer
unreachable
all calls
all calls unconditionally
calls under certain conditions
The necessary call forwards must be activated and registered, then the button 'Submit' must
be pressed to activate it to the network.
Use the 'Query' button to request for the current status of the call forwarding.
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6.5.6
coM.sat IP Basic
Call Barring
Figure 27: Call Barring
The GSM / UMTS network offers the user the ability to barr incoming and outgoing calls. To
activate the bar, a password must be created.
•
Change Password: This parameter allows the user to change the password for the
call barring.
•
Last Password Change Status: After the user has changed the password, the last
changing status will be displayed.
The various options are:
• Barr all outgoing calls (BAOC)
• Barr all international outgoing calls (BOIC)
• Barr all international outgoing calls except home country (BOIC-exHC)
• Barr all incoming calls (BAIC)
• Barr all incoming calls, when roaming out of home country (BIC-Roam)
After the parameters have been transmitted with the button ‘Submit’, it is recommended to
query the status with the button ‘Query’.
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6.6 GSM/UMTS Channel Status
The user can request the status of both GSM/UMTS channels.
Figure 28: Status of both GSM/UMTS Channels
•
Activity Status: The activity status of both GSM/UMTS channels will be displayed in
this field. The various status are described below:
o
o
o
o
o
o
o
Module initialisation:
SIM-PUK required:
SIM PIN wrong:
SIM not available:
SIM available:
Network not available:
Network available:
•
Module Firmware: The current firmware version of both GSM/UMTS modules will be
displayed in this field.
•
IMEI: The IMEI of the both GSM/UMTS modules, which are built in the coM.sat IP Basic
will be displayed in this field. The IMEI is a unique programmed code in the GSM/UMTS
module, which is used for identification. If e.g. the coM.sat IP Basic has been stolen, the
network provider is able to lock the GSM/UMTS module with help of the IMEI.
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Initialisation of the GSM/UMTS modules
PUK must be entered
wrong PIN has been entered
The SIM card is missing or is not recognized
The SIM card is recognized
Searching for network to register
Network and channel are available
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•
IMSI: This is the unique international mobile subscriber identity (IMSI), which is
programmed on the SIM card.
•
Network Operator Code: In this field the numeric network operator code (MCC-MNC)
of the network will be displayed.
•
Network Operator Name: In this field the name of the network operator, in which
the SIM card of the module in the coM.sat IP Basic is registered, will be displayed.
•
Registered with Network: The type of network, in which the coM.sat IP Basic is
registered, will be displayed in this field. The possible network types are:
o GSM
o 3G/UMTS
o Not registered (if the GSM/UMTS channel is not registered in any network).
•
SMS Service Centre Number: This is the Service Centre Number, which is provided
by the service provider for the SMS service. In most cases this number is already
programmed on the SIM card.
•
Signal Strength: The coM.sat IP Basic supports the display of the signal strength of
both GSM/UMTS channels. After successful installation it is recommended to check the
signal strength at the installed location. With this measurement it is possible to find a
appropriate location to improve poor voice and transmission quality.
•
Prepaid Minutes: The remaining minutes of the balance of the prepaid card.
•
SIM Lock Password: If the user want to use certain SIM card for a channel, it is
possible with the SIM Lock password. As soon as the user has entered the password,
no other SIM card will be accepted in this channel. The SIM Lock password can be
changed via the button ‘Change Password’. The default password is 12345678.
•
SIM-PUK: The PIN is a safety function of the SIM card. If the access with PIN is
activated, the GSM/UMTS module requests the PIN after every restart. If a wrong PIN
is entered three times or the unit is restarted three times, the SIM card is locked. The
SIM card can be unlocked only with the PUK, where a new PIN has to be defined. If a
wrong PUK is entered ten times, the SIM card is useless.
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6.7 GSM/UMTS Channel Destination Number Calling Number based
Figure 29: CLI table
If the destination number based on CLI table' is selected, then the user should be
programmed in the destination number for each phone number of a calling party. Here, the
calling party is automatically routed to the destination number. This table is also used for the
call-back function.
The CLI table parameters are:
•
changed.
•
Calling Number: In this edit field the user can enter the number of the calling party
number.
By default the first entry is set to ‘No Match Found’. All other edit fields are blank.
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•
Destination Number: In this edit field the user can enter the destination number. The
maximum characters of this number are 24. Valid characters are 0 to 9, *, # and the
dot (.).
•
Allow Call-back on receipt of Call?: The call-back function allows to recall a specific
GSM numbers and therefore save costs for the caller. The recall is carried out due to a
call (which is rejected) if the signalling mode was selected 'Allow call-back on receipt of
Call’. Will be entered in the 'Destination Number' field one number, a call will be setup
after the call-back subscriber has accepted the call. Otherwise, the subscriber is routed
to the destination number that was specified in the channel parameters.
Note: This function is only given if the CLI is received during an incoming call.
The call-back will be initiated from the same channel, from which the call has been
received.
•
Allow Call-back on receipt of SMS?: The call-back is initiated by receiving a SMS if
the signalling type is ‘Allow Call-back on receipt of SMS’ has been selected. The callback will be initiated to number of the sender. If a number is entered in the edit field
‘Destination Number’, a call is setup to the call-back subscriber if he accepts the call.
Is this edit field blank and the SMS text has a valid number, so this call-back
subscriber will be routed automatically to this number. The connection to the
destination number will be established, if the call-back subscriber accepts the call.
Otherwise the subscriber will be routed to the destination number, which is
programmed in the channel parameters.
Note: The call-back will be initiated from the same channel, from which the call has
been received.
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6.8
SIP-Trunk 1/2/3/4
6.8.1
General
The coM.sat IP Basic supports 4 SIP Trunks. All SIP Trunks could be registered in one Internet
Service Provider (ISP) or in various ISPs.
The various SIP Trunk parameters are declared as followed:
Figure 30: SIP Trunk Settings
6.8.1.1
Registrar Settings
•
Enable SIP Trunk: To make incoming and outgoing calls the SIP Trunks must be
enabled.
By default the SIP Trunks are disabled.
•
Name: The user can enter a name for the SIP Trunk. This should have a length of
maximum 12 characters. All alphanumeric characters are allowed.
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This name will be displayed at the called party number, if an outgoing call is initiated.
By default, this edit field is blank.
•
SIP ID: In this edit field the ISP provided SIP ID will be is entered. The SIP ID must
contain maximum 40 characters. All ASCII characters are allowed. If for example the
ISP provided SIP URI [email protected] is, then the SIP ID is 12345.
By default for each SIP Trunk ID is the ‘*’ programmed.
Note: In case of peer to peer connection the first SIP Trunk is always used to route all
incoming calls, which is configured with a SIP ID as '*' programmed SIP Trunk.
•
Registar Server Address: Enter the ISP provided address of the SIP registration
server. This can be an IP address or domain. The address of the SIP registration server
may contain up to 40 characters.
Extended ASCII characters are allowed.
By default, the address of the registry server is empty.
•
Port of the Registrar Server: Enter the ISP provided the receiving port of the
registry server. The valid range of the port of registry server is between 1024 to 65535
The default setting of the port of registry server is 5060
•
Send REGISTER: coM.sat IP Basic sends a REGISTER message to the registry /
outbound proxy server. If the user does not want the device sends the message, so
this option is deactivated. By default, this parameter is enabled for all SIP Trunks
•
Add ‘rinstance’ in REGISTER Message?: If the user does not want the coM.sat IP
Basic rinstance in the REGISTER message is included, this option must be deactivated,
otherwise the coM.sat IP Basic rinstance will be inserted in all REGISTER messages.
By default this parameter for all SIP trunks is enabled.
•
Gateway Mode: The coM.sat IP Basic is for all SIP-ID and accept the registration /
outbound proxy server programmed incoming calls. If the user wants the gateway
accepts the incoming call to a SIP ID, but only for the registry / outbound proxy server
programmed versions, this field should be activated.
For this reason, the SIP trunk when activated as a gateway would be used.
By default this parameter for all SIP trunks is disabled.
•
Allow Outgoing Calls without Registration? : The unit does not allow outgoing calls
when the SIP trunk is not registered. However, in order to allow outgoing calls when
the device is not registered in the registry server, the parameter must be activated.
By default this parameter for all SIP trunks is disabled.
•
Re-Registration Timer: The user must enter the value for the re-registration timer.
Before the timer expires, the coM.sat IP Basic should send the registration request to
the registry server to be registered again. The valid range of this timer is from 00001
to 65535. By default the value of re-registration timer is 3600 seconds.
•
Registration Retry Timer: This is the period between the renewed attempts to
register with the SIP server. The coM.sat IP Basic unit sends the registration request
until the SIP trunk is registered with the SIP server. The valid range of this timer is
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00001-65535 seconds. By default, the value for the Registration Retry Timer is 10
seconds
•
Maximum Calls: The user must program the maximum number of calls that are
allowed on each SIP trunk. The valid range of allowed calls is between 01 and 08.
By default, the maximum value on all SIP trunks calls is 8.
6.8.1.2
Authentication
•
User ID: For the registration of the SIP trunk with the SIP server a user ID is required.
This can be up to 40 characters long. All ASCII characters are allowed.
•
User Password: Enter the ISP provided authentication password. The password can
be up to 24 characters long. By default, the field is empty
6.8.1.3
Outbound Proxy
•
Enable Outbound Proxy: The outbound proxy server, depending on the requirements
can be activated or deactivated. This can only be activated if the ISP is a SIP outbound
server for processing of voice calls.
By default, the outbound proxy is disabled.
•
Outbound Proxy Server Address: The server address is the address of the outbound
proxy server. This may be the SIP server address, an IP address or domain. The
address of the outbound proxy server can be up to 40 characters long and contain all
ASCII characters.
By default, the address of the outbound proxy server is empty.
•
Outbound Proxy Server Port: The port of the outbound proxy server is listed on the
outbound port of SIP proxy server. This may correspond to the port of the SIP server.
The range of ports is between 1024 and 65535. By default the outbound proxy server
port is 5060.
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6.8.1.4
coM.sat IP Basic
Vocoder Preference
The coM.sat IP Basic supports for outgoing calls the following 7 codec:
1.
2.
3.
4.
5.
6.
7.
G.729
G.723.1
GSM FR
iLBC_30ms
iLBC_20ms
G.711 (µ-Law)
G.711 (A-Law)
By default the preferred codec are in the same row as listed before.
Notes: The codec for outgoing calls should be selected according to the programmed
preferences. This parameter is checked only in case of outgoing calls. The codec of the
called party is requested to take into account only if this is supported from coM.sat IP
Basic, otherwise the call is rejected.
•
G.723.1 Bit rate: For the codec G.723.1, which is selected for an outgoing call, the
two preferred bit rates can be selected:
o 5,3 kbps
o 6,3 kbps
By default the preferred bit rate for the G.723.1 codec is 6,3 kbps.
6.8.1.5
Use Symmetric RTP?
This option is enabled, when the coM.sat IP Basic is registered on a public IP and all outgoing
calls to the SIP client, which is located behind a NAT router, be made or received calls from
SIP client behind the NAT router is situated, be accepted. This parameter is applicable only
for peer-to-peer calls.
By default this parameter is disabled for all SIP trunks.
6.8.1.6
Send Caller-ID?
If the user does not want to send the Caller-ID when making calls over the SIP trunk, this
parameter has to be disabled. The coM.sat IP Basic will send the Caller-ID, which is configured
in the SIP trunk, only if this parameter is enabled. By default the parameter is enabled.
•
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Send Caller-ID received on Source Port: If this option is enabled, the coM.sat IP
Basic is transmitting on the source port in the 'From' of the INVITE message received
Caller-ID, if an outgoing call using the SIP trunk.
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6.8.1.7
coM.sat IP Basic
Source Port IP Address
The IP address of the source port is the IP address of the source of the system are sent over
the IP address for sending data packets. The coM.sat IP Basic provides three options for
selecting the IP address of the source ports:
•
Use WAN IP Address: Select this option, if the coM.sat IP Basic should be connected
with the World Wide Web (Internet).
•
Use IP Address Fetched using STUN: STUN is the most common protocol used by
SIP clients, when they are installed behind the NAT router. STUN is used to set the
public IP address and port mapping on the NAT router, behind which is the SIP client.
Select this option if the coM.sat IP Basic behind the NAT router is not arranged
symmetrically. This option works only if it is disabled on the SIP outgoing trunk
function. If you choose this option, program the address and the port of the STUN
server in the 'Network Port Parameters'.
•
Use Router’s public IP Address: Select this option, if the coM.sat IP Basic is behind
the NAT router. This option works only, if the function for outgoing calls is disabled in
the SIP trunk. If you choose this option, program the public IP address of the router in
‘network port parameters’. By default, the option ‘Use IP address of the network port’
is selected as the IP Address of the source port.
6.8.1.8
DTMF Option
This option decides how the DTMF tones will be transmitted over the IP network, if a DTMF
digit has been pressed. The coM.sat IP Basic supports in SIP the following options for DTMF
signalisation:
• RTP (RFC 2833)
• SIP INFO
• In-Band
By default, DTMF tones will be transmitted via RTP (RFC 2833).
6.8.1.9
FAX Options
This parameter indicates how fax massages will be processed. The coM.sat IP Basic supports
the following fax protocols:
• T.38 (UDPTL)
• T.38 (RTP)
• Pass through
By default the protocol is T.38 (UDPTL).
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6.8.1.10 Default Transport for Outgoing message
The coM.sat IP Basic supports the following three options for outgoing SIP messages. These
options will be checked only, if the user has activated SIP over TCP in the network
parameters. Otherwise, all outgoing messages will be transported via UDP.
•
UDP: UDP will be used as the Default Transport Protocol for outgoing messages.
•
TCP: TCP will be used as the Default Transport Protocol for outgoing messages.
•
TCP: (with call-back on UDP): TCP will be used as the Default Transport Protocol for
outgoing messages.
If the TCP connection failed, the coM.sat IP Basic will try to send the messages again over
UDP. By default, for outgoing SIP messages UDP will be used.
6.8.1.11 Call Hold Method using
Select the one option to hold a SIP call. There are two options:
•
RFC 2543: The coM.sat IP Basic sends the connecting information:
0.0.0.0 to ITSP.
•
RFC 3261: The coM.sat IP Basic sends the connecting information:
The IP address using for the connection.
By default the option to hold a call is ‘RFC 3261’ for every SIP-Trunk.
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6.8.2
coM.sat IP Basic
Incoming Calls
The field ‘Allow Incoming Calls’ must be activated to route the incoming calls.
Figure 31: SIP-Trunk incoming/outgoing calls
The routing mechanism includes two steps:
• Selection of the destination number
• Selection of the destination route
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Destination Number Selection: The various options of routing to the destination
number is described as follows:
o
Routed to a fixed destination number: If 'Routed to a fixed destination number’
is selected, then in the edit field 'Fixed destination number to use' the
destination number has to be entered. The destination number may be up to 24
digits. Valid characters are 0 through 9, *, #, and the dot (.).
By default, the edit field is empty.
o Routed as per Calling Party Number: Please refer to the topic ‘destination
number SIP Trunk: Based on calling number'.
o Routed as called party number: A call is routed to the destination number as a
function of the received number of the called party. The number of the called
subscriber is routed to the destination number from the unit.
o Answered and routed as per the number dialled by the caller: If the option is
‘Answer call with dial tone’ is used, then the system answers the incoming call
and emits a dial tone. Now the caller can dial any number per DTMF tone.
If the caller doesn’t dial any number within the set time out, the unit can either
disconnect the call, or route them to a fixed destination number. The fixed
destination number is removed from the box 'Fixed to use destination number'.
The timer is programmed by default to 7 seconds.
If the option 'Respond the call with dial tone' is programmed, the unit will
consider the following criteria as the end of dialling:
If the maximum of the selected characters with the programmed value in
the 'route the call on receipt of <> digits' match
If the selected character is matched with '#' or '*' as programmed
After the expiry of the timer, the 'after last <> seconds of digit dialled' is
programmed
•
Destination Route Selection: The unit supports the following options to route the
destination port:
o As per routing table: For the option ‘as per routing table’ refer to the topics
‘Routing Table’.
o Fixed: The fixed routing option allows the user to connect three different types of
port to route the call to be set independently of the source port received /
recorded to the destination number. The user can select the required option
from a drop-down list set. If the first option, programmed in the ports / channels
are busy, would select the second choice and so on. The list of port options are
as follows:
None
GSM/UMTS channel 1
GSM/UMTS channel 2
Every GSM/UMTS channel (first free)
Every GSM/UMTS channel (alternately)
TE-Port (Ch-1)
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TE-Port (Ch-2)
TE-Port (Ch-1,2) (first free)
TE-Port (Ch-1,2) (alternately)
NT-Port (Ch-1)
NT-Port (Ch-2)
NT-Port (Ch-1,2) (first free)
NT-Port (Ch-1,2) (alternately)
SIP-Trunk 1
SIP-Trunk 2
SIP-Trunk 3
SIP-Trunk 4
Every SIP-Trunk (first free)
Every SIP-Trunk (alternately)
First free: If multiple ports contain the same routing destination, then it is
always the first available port used for routing the call.
Alternately: If it contains multiple ports, the same routing destination, then
the first call on the first channel and the subsequent call routed via the second
channel, and so on.
•
Anonymous Calls should be: If the calls has been received without CLIP, the user
can program the following options:
o Rejected
o Routed to a fixed destination number
o Answered and routed as per the number dialled by the caller
•
Rejected calls from Callers: This option gives the user the ability to program the list
of callers to be rejected. The overall numbers are in the table caller numbers' to
program. By default this option is disabled.
•
Apply Allowed-Denied Number Logic: The option 'Apply Allowed-Denied Number
Logic' provides the ability to dial a specific number or range of numbers from a specific
port to allow or reject. This property is applicable only to the source port. The user
should program the allowed and rejected phone numbers in different lists (black list /
white list).
The option 'Apply Allowed-Denied Number Logic' is not applicable in the following
cases:
o
o
o
The destination number is consistent with an emergency number.
For pre-programmed destination numbers.
For the destination number via CLI method based destination number recorded.
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6.8.3
coM.sat IP Basic
Outgoing Calls
The user can use the port for making outgoing calls. This field 'Allow Outgoing Calls' must be
activated.
•
Apply Automatic Number Translation (ANT) logic: In certain cases, the number
dialled by the ISDN network is not understood. For this reason, this number will be
replaced by another.
Note: Empty numbers cannot be replaced with any number string.
6.9 Status SIP Trunk
Figure 32: Status of the SIP trunks
This option displays the user the following parameters:
•
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SIP Trunk Number: This field displays the SIP Trunk number.
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Status: This field displays the status of the SIP Trunk. The different status options that
may appear in this field are explained below:
o Disabled:
o Registered:
o Active:
o Failed:
SIP Trunk is disabled
SIP Trunk is active and waiting for an answer from the SIP
Server
SIP Trunk is registered at the SIP Server
An error has occurred in the SIP Trunk and no calls can be
done (meets only the Proxy Account).
•
Registration Time: This field displays the remaining time to re-register the SIP
trunks, after it is successfully registered. This time is provided from the registration
server. If the duration of the registration expires, the SIP trunk will be re-registered.
•
Registration Retry Count: This field displays the total number of registration
messages that were sent to the registration server to register the SIP trunk.
•
Failed Reason: This field indicates the reason of a failed registration. The various
reasons for a failed registry, which can be displayed in this field are given below:
o Sending Message failed: Is displayed when a sent request to the registry server
failed.
o Create Registry Clients failed: Appears when the SIP stack has a memory /
resource limit or exceeds the number of registered SIP clients programmed in
the stack.
o Disconnecting the client registration failed: Appears when SIP stack has a
memory / resource limit or the number of registered SIP clients exceeds the
programmed in the stack.
o Send the request failed: Appears when the DNS server is not programmed.
o Local error: is displayed when the DNA is request failed.
o Response Timeout: This is displayed after the expiry of the timer for general
inquiries.
o Error message from 4xx to 6xx: This is the code of the error response.
o No contact header in 2xx: Appears when no contact has been received by the
SIP server in 2xx response.
o Authentication failed: This appears when the SIP server does not authenticate
the client.
o STUN address not programmed: If you see, when STUN enabled, but the address
is not configured.
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o STUN request failed: Appears when a request has failed at the STUN server.
o Outgoing address is not programmed: Appears if enabled outgoing but the
outgoing address is not configured.
o IP address of the router is not programmed: If you see, if IP address of the
router to be used in transmission, but it is not programmed
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6.10
coM.sat IP Basic
Destination number SIP Trunk Port - Based on calling
number
This method is used when a call should be routed to a specific number depending on the
number of the caller.
Figure 33: SIP routing for incoming calls
In the routing table, the user, depending on the calling number enter a destination number.
The single table parameters are:
•
Index: This field displays the index number (001 to 500) is indicated.
•
Calling Number: In this field the number of the calling subscriber is registered.
By default, the first entry 'No match' with shows. All other entries are empty.
•
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Destination Number: In this field the destination number is entered, which may
consist of a maximum of 24 characters. Valid characters are 0 through 9, *, #, and the
dot (.).
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Note: If the destination number is an IP address, then the call is programmed via the
SIP trunk group routed.
6.11
SIP Proxy
6.11.1
General
Figure 34: General Settings for SIP Proxy
The coM.sat IP Basic supports up to 8 SIP trunks. Each SIP trunk can carry out the registration
of a maximum of 3 different locations. The SIP trunk can cause the registration over WAN /
WWAN, and the LAN port of coM.sat IP Basic. If the call is made via the SIP trunk, the coM.sat
IP Basic will signal the call to all locations (if the SIP trunk is registered from different
locations).
The various parameters for the SIP trunk are explained below. The user should have this
program according to the required standards.
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6.11.1.1 Registration Settings
•
Server Domain: This parameter is only when you configure the SIP Trunk. The coM.sat
IP Basic is able to support one domain from the registration of SIP clients and SIP
trunk. By programming the server domain, the SIP client is also registered with the
domain.
Note: If both server domain and host name for the dynamic DNS to be configured,
view the coM.sat IP Basic host name as a domain server. The configured domain server
will be ignored.
•
IP Address of the Source Port: This parameter is the one to use NAT 'traversal
mechanism' on, when the coM.sat IP Basic is behind a NAT router installed. The
following options of the IP addresses of the source ports are available:
o WAN IP address: The WAN port of the coM.sat IP Basic is connected directly with
the public IP network.
o Public IP address of the router to use: Here the coM.sat IP Basic is behind a NAT
router and the router supports a symmetric NAT.
o Retrieved using the IP address using STUN: Here is the coM.sat IP Basic behind a
NAT router where a symmetric NAT is not supported.
•
Minimum Registration Timer (sec): This is the minimum timer, which the SIP trunk
in its registration requests should be sent. If the discharge value in the REGISTRATION
message is smaller than this value, the request is rejected. This value is used by the
system if UA is located behind the NAT router. The valid range for this timer is between
10 and 99999 seconds.
By default the value is 45 seconds.
•
Maximum Registration Timer (sec): This is the maximum timer, which will accept
the coM.sat IP Basic in the received REGISTER request. If the coM.sat IP Basic is
receiving more than this value, the value configured in the SIP message is sent. The
valid range of this timer is between 10 and 99999 seconds.
The default value is 3600 seconds.
•
Private Key: The Private Key is a security feature that uses the coM.sat IP Basic
authentication for the SIP messages. It is the MD5 authentication key that uses the
coM.sat IP Basic for encryption, decryption of SIP messages. The private key can
contain a maximum of 24 characters. It allowed all ASCII characters.
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6.11.1.2 Codec Preference
The coM.sat IP Basic selects an appropriate codec, while it makes a call to the SIP extension
or receives a call from the SIP extension. It selects a codec from the different order of
preferences by the coM.sat IP Basic supported by codecs.
•
Preferences 1 – 5: The basic codec supported by the coM.sat IP Basic are arranged as
follows according to the preferences:
o G.729
o G.723.1
o GSM FR
o G.711 µ-Law
o G.711 A-Law
•
G.723.1 Bitrate: Is the outgoing call for one of the G.723.1 codec is selected, then
one of two preferred bit rates are predefined:
o 5,3 kbps
o 6,3 kbps
As a setting for the G.723.1 codec, the bit rate of 6.3 is defined kbps.
Note: When G.723.1 is selected, the selected bit rate is used only if the RTP packets are
sent. Upon receiving the RTP packets from different IP participants, both the G.723.1 bit
rates will be accepted.
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6.11.2
Settings
6.11.2.1
General Settings
Figure 35: SIP Proxy Settings
•
Enable SIP Extension: Enables the option of a SIP extension. If this option is not
enabled, then no extension will be used.
By default, all SIP extensions are enabled.
•
Name: Enter the name for the SIP extension number. The name in this field will be
displayed on the phone of the called party the caller's name as the SIP extension. You
can configure each name with a maximum of 18 alphanumeric characters
•
SIP ID: The SIP ID to register the SIP extension for the filing of the coM.sat IP Basic
necessary. It is the number to which the SIP extension can be called. Each user of a
SIP extension of the coM.sat IP Basic can call another SIP extension by dialling the
assigned SIP ID. The SIP ID of a SIP extension has a unique number string with a
maximum of six characters (minimum 2). It is any combination of the characters from
0 to 9 permits.
A SIP-ID cannot be reused.
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•
Authentication ID: The coM.sat IP Basic authentication ID used to authenticate the
message received from the SIP extension SIP messages. The number can be a string
with a maximum of six characters (minimum 2). It permits any combination of the
characters from 0 to 9. The authentication ID is only applicable if one of the
'Authentications' of the SIP message options, REGISTER or INVITE is activated.
Otherwise, the authentication ID is not used.
•
Authentication Password: Where in the coM.sat IP Basic authentication of SIP
messages required. The password can be up to 24 characters long and contain all
ASCII characters.
The default authentication password is: 1234
•
DTMF Options: This option determines how the DTMF signals are transmitted over
between IP extensions, if a DTMF digit has been pressed. The coM.sat IP Basic supports
the following options for SIP DTMF choice:
o RTP (RFC 2833)
o SIP INFO
o In-Band
By default, the DTMF signals via RTP (RFC 2833) transferred.
•
FAX Options: This parameter can be determined, such as fax messages through the
coM.sat IP Basic handled by the SIP Extension: FAX options.
The coM.sat IP Basic supports the following protocols FAX:
o T.38 (UDPTL)
o T.38 (RTP)
o Pass through
By default, T.38 (UDPTL) is set.
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6.11.2.2 Authentication of SIP messages
•
Authentication of REGISTRATION request: These can be configured as an
activation or deactivation. Upon activation, the resultant of the SIP extension
REGISTER message is authenticated, otherwise the message is not authenticated.
By default the field is activated.
•
A Authenticate the INVITE request: This can be configured as an activation or
deactivation. Upon activation, the resultant of the SIP extension INVITE message is
authenticated, otherwise the message is not authenticated.
6.11.2.3 Outgoing Calls
•
Destination Route Selection: The device supports the following options to select the
destination port to route the call:
o As per Routing Table: For the option 'As per routing table', please see the
chapter 'routing table'.
o Fixed: The fixed-routing option allows the user to connect three different types
of port to route the call to be set independently of the source port received /
recorded to the destination number. The user can select the required option
from a drop-down list set. If the first option, programmed in the ports / channels
are occupied, would select the second choice and so on. The list of port options
are as follows:
None
GSM/UMTS channel 1
GSM/UMTS channel 2
TE-Port (Ch-1)
TE-Port (Ch-2)
NT-Port (Ch-1)
NT-Port (Ch-2)
SIP-Trunk 1
SIP-Trunk 2
SIP-Trunk 3
SIP-Trunk 4
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channel, and so on.
•
Apply Allowed-Denied Number logic: The option 'Apply Allowed-Denied Number
logic' offers the possibility to dial a specific number or range of numbers from a specific
port to allow or reject.
This property is applicable only to the source port. The user should program the
allowed and rejected phone numbers in different lists (blacklist / white list).
The option 'Allowed Denied Number' is not applicable in the following cases:
o For the destination number via CLI method based destination number recorded.
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6.11.3
coM.sat IP Basic
Status
The status of the various SIP extensions will be shown on this page as follows:
Figure 36: SI Extension Status
•
SIP Extension: This field can not be edited. It displays the SIP extension number for
which the various parameters.
•
Name: This field displays the name, which was configured for the SIP extension.
•
SIP ID: This field displays the SIP ID that was configured for the SIP extension.
•
Status: The following status will be displayed:
o Disabled: If the configuration of SIP extension is disabled.
o Registered: When configuring the SIP extension is enabled and that it is
registered in the system.
o Not registered: If the configuration of SIP extension is enabled and this is not
registered in the system.
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coM.sat IP Basic
Contact 1/2/3: The SIP extension can be registered from different locations. The
coM.sat IP Basic displays the details of three locations from which the SIP extensions
have been registered.
The details displayed are as follows:
o The Contact header in the REGISTER message received URI.
o The registration period after the coM.sat IP Basic user registration to delete from
its database.
o IP address: the port from which the SIP extension is registered.
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6.12 Routing Table
If the routing was from users under 'routing table' is selected, the following parameters can
be programmed:
Figure 37: Routing Table
•
Index Number: The index number indicates the current number of entries for the
destination number and port options. There can be up to 100 entries in this table. The
index starts with ‘001’ . If no match is found, the index is used 001.
•
Destination Number: In this field the frequently called numbers entered. The
destination number may be up to 24 characters. These are the characters allowed 0 to
9, *, #, and (.) Is the point. By default the field is empty.
•
Minimum Digits: In this field the minimum length of the destination number is
entered. Is in an election the number string less than the minimum length the call will
be rejected. The range of the minimum length of the target number is 1 to 24
By default, the minimum length is determined by three digits.
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•
Maximum Digits: In this field the maximum length of the destination number is
entered. Is in an election the number is greater than the maximum string length, then
the number string ‘cut off’ means The remaining digits are discarded and the call is
routed. The range of the maximum length of the target number is 1 to 24 Default is 16
•
Port 1/2/3: The user can select three different types of port to route the call. This list
can be adjusted according to the desired preference from a drop-down. Are
programmed in the first option ports / channels occupied, the second choice is selected
and so on. The list of port options are as follows:
None
GSM/UMTS channel 1
GSM/UMTS channel 2
TE-Port (Ch-1)
TE-Port (Ch-2)
NT-Port (Ch-1)
NT-Port (Ch-2)
SIP-Trunk 1
SIP-Trunk 2
SIP-Trunk 3
SIP-Trunk 4
channel, and so on.
Note: The routing table is the same for all port types.
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6.13
coM.sat IP Basic
Static Routing
The static routes to the table helps the calls between point-to-point locations (connected over
MPLS, Frame Relay, etc.) while in route to the public Internet.
Figure 38: Static Routing
In general, the static routing must not be configured when the coM.sat IP Basic behind a NAT
router is connected.
The LAN interface of the NAT router acts as the default gateway for the coM.sat IP Basic. All
calls initiated by coM.sat IP Basic are routed from the LAN port of the NAT router.
But if you have multiple offices connected via MPLS, Frame Relay, etc., and peer to peer calls
between different offices and at the same proxy calls to the public Internet or wish to receive
calls, the user must configure the table static routing in the coM.sat IP Basic.
In the above figure two offices over MPLS or frame relay are connected. The user would like
peer to peer calls between two offices over a Frame Relay / MPLS network and also proxy
calls over the public Internet.
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The user should program the following parameters of the 'static routing table':
Figure 39: Static Routing
You can program a total of 8 entries in the Static Routing Table.
Each entry has three parameters.
•
Destination Address: can be the IP address of the called SIP device or the network
address of the location of the called party's SIP device.
•
Subnet Mask: is the mask that is applied to the destination address.
•
Gateway Address: is the IP address of the LAN port on your router, from where you
want to route the call to a specific target. This address must be on the same subnet, in
which the calling coM.sat IP Basic is located.
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By programming this table, the coM.sat IP Basic can route the calls based on the destination
address. Calls for a particular network can be routed to a specific gateway.
As indicated in the above figure should perform the following programming in the Static
Routing Table of the coM.sat IP Basic location A are made as an example.
Index
Destination Address
Subnet Mask
Gateway Address
1
192.168.2.0
255.255.255.0
192.168.1.1
2
3
4
5
There is only one entry to be configured in the Static Routing Table.
The destination address 192.168.2.0 specifies the network address of location B.
The subnet mask specifies the destination address on the mask to be applied.
The gateway address 192.168.1.1 is the LAN address of the router that connects the location
A with location B.
The IP address of the LAN interface of the router that connects to the location A with the
public Internet should be configured as the default gateway. Using the above programming all
calls through the coM.sat IP Basic at 192.168.2.0 / 24 made calls routed to the router, the
location A links to the location B. Whereby all calls from the coM.sat IP Basic not the
192.168.2.0 / 24 address to be made, will be routed to the default gateway.
At location B is in a similar coM.sat IP Basic programming necessary to allow calls to location
A.
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6.14
coM.sat IP Basic
Peer to Peer Calls
A call over the IP network without intervention of the proxy (SIP) server is referred to as
peer-to-peer call. Since the peer to peer calls needn’t a SIP server this kind of voice
communication is free of charge. For companies, this cost saving factor offers an attractive
voice communication within departments or offices.
The following example helps to understand the functionality of this option is better:
•
•
•
•
•
A company is headquartered in Hamburg and a branch in Berlin.
Both offices are on VLAN services of the ISP connected.
The user must configure the network settings in both office settings so that the peer to
peer calls between two offices are possible.
Suppose the IP address of the coM.sat IP Basic in the Hamburg office is 192.16x.1.xx
and the coM.sat IP Basic in the Berlin office 192.16y.1.yy.
The extension numbers are programmed in Hamburg 3001, 3002, 3003 and so on, and
the extension numbers, which are programmed in Berlin, are 2001, 2002, 2003 and so
on.
Figure 40: Peer to Peer Connection
Now, if the user wishes to call in the Hamburg office with the number 3001, the Berlin office
with the number 2002, then he should select either [email protected] or
2002@domainname. This procedure must be at peer to peer calls via the coM.sat IP Basic
considered.
Peer to peer table contains a total of 500 entries. Thus, 500 numbers for Peer to Peer calls
can be programmed. The first entry in the Peer to Peer table is used for the event, if no
match is found.
Each entry in the Peer to Peer table contains the destination number, destination address, the
minimum characters, the maximum characters and the name.
•
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Destination Number: The destination number is the DDI of the coM.sat IP Basic or a
PBX, which is connected to the coM.sat IP Basic (to the called party number).
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•
Minimum Characters: This defines the minimum number of characters that are
necessary to dial in order to allow routing of the call to its destination. The minimum
length of the selected characters should be less than or equal to the programmed
maximum characters.
•
Maximum Characters: This defines the maximum number of characters from the
coM.sat IP Basic to route the call to the desired destination will be accepted. If the user
selected destination number is greater than the maximum number, then the remaining
characters are discarded and the call is routed.
•
Destination Address: The destination address matches the IP address or domain
name of the coM.sat IP Basic, to which the call is made for the programmed number.
•
Name: In the edit field 'Name' the name is entered, the contact person to whom the
call is made. This name is used for informational purposes only and is not otherwise
used in the coM.sat IP Basic. The coM.sat IP Basic uses the Best Match Found logic to
compare the destination number to be dialled in with the peer to peer table entries
programmed. The user dialled number string should meet the criteria of the minimum
or maximum characters that are programmed for this destination number. If the
selected destination number the entries that are programmed into the peer to peer
table, not equal, then the criterion should be 'No matches found for minimum and
maximum characters' verified. If the selected destination number of the entry of the
best match is found, but that the minimum criteria of character, which was
programmed for such an entry is not met; the coM.sat IP Basic will transmit the user an
error tone. If the selected destination number of the entry of the best match is found,
but the destination number string is greater than the programmed for such a record
maximum characters, then the coM.sat IP Basic discard the rest of the characters and
use the destination address to route the call.
When to dial the destination number is associated with multiple entries in the Peer to
peer table as another criterion, the minimum and maximum character length is
compared. This account of characters avoids a conflict in peer to peer application.
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6.15
coM.sat IP Basic
Peer to Peer Routing Table
Figure 41: Routing for Peer to Peer
In order to set up peer to peer routing, the following parameters must be programmed:
•
Index Number: The index number indicates the current number of entries for the
Peer to Peer connection. There can be up to 100 entries in this table, starting 001-100.
If no match is found, the index is used 001.
•
Destination Number: In this field the frequently called numbers entered. The
destination number may be up to 24 characters. These are the characters allowed 0 to
9, * and #. By default all fields are empty.
•
Minimum Digits: In this field the minimum length of the destination number is
entered. If the dialled number string is less than the minimum length, then the call will
be rejected. The range of the minimum length of the target number is 01 to 24
By default, the minimum length is determined by 3 digits.
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•
Maximum Digits: In this field the maximum length of the destination number is
entered. The range of the maximum length of the target number is 01 to 24 Default is
16.
•
Destination Address: The destination address can be up to 40 characters long. It
allowed all ASCII characters. By default all fields are empty.
•
Name: The name of the called destination is stored in this edit field. This can be a
maximum of 12 characters. All ASCII characters are allowed.
Notes: The peer-to-peer routing table is used only if the destination port is a SIP trunk.
When an incoming call on the peer to peer is signalled in the coM.sat IP Basic unit, it will be
routed first through the SIP trunk.
Using the coM.sat IP Basic can be made peer-to-peer calls between two clients if:
• Both clients are in the public network.
• Both clients are in the same private network and make calls only on the private
network (STUN is disabled).
• Both clients are on different private networks and are not using the symmetric NAT
(STUN here must be used by both clients involuntary).
• A client is in the public network (The use of symmetric RTP is required) and the other
client is on the private network.
Using the coM.sat IP Basic cannot be made peer-to-Peer Calls:
• Between the clients in different private networks, private networks, if both use the
symmetric NAT router.
• Between clients in a different private network, while a public network.
• Between clients on the same private network and public network simultaneously.
• Between clients in the same private network, different private network and public
network simultaneously.
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6.16
coM.sat IP Basic
System Call Diversion
This feature is applicable, when a called number is routed via the NT port.
Figure 42: Call Diversion
The user must program the destination number are routed to the called numbers. The various
options for call diversion are:
•
Call Forward - unconditional: All calls are always routed to the programmed
destination number.
•
Call Forward - Busy: The calls are diverted to the programmed destination number
when the called party number is busy.
•
Call Forward – No Reply: The calls are forwarded to the programmed destination
number, if there is no response for the programmed ring timer.
If the called number is 3001 and the unconditional call forwarding is to 3002, then the
device will automatically route the call to the number 3002.
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To activate call forwarding, the user has to program the following parameters in the table
for each of the above programmed options:
•
Called MSN: This is called number from the caller.
•
Status: To enable the corresponding call forwarding, the user has to enable this field.
•
Number: The number to which the call has to be diverted.
•
Ring Timer (in Seconds): When call forwarding 'Call Forward No Reply' has been
selected, the call is redirected to the programmed number after the expiration of this
timer. After all parameters are programmed, the 'Submit' must be pressed button to
transfer the changes. If all parameters have to be reset, the button 'Default All' must
be pressed.
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6.17
coM.sat IP Basic
Emergency Numbers
The coM.sat IP Basic supports without any barring the direct dialling of emergency numbers at
all ports. The user has to enter this emergency numbers in the following table:
Figure 43: Table of emergency numbers
The table of the emergency numbers has ten entries, where each entry has two parameters:
•
Emergency Number: Enter the emergency number.
•
Port/Group Type: From the drop-down menu / the corresponding port (s) / group is
selected to route the call. The first four entries in the index 01-04 are assigned to the
preset emergency numbers in their own country. The emergency number should
consist of a maximum of 4 digits. You may use only the digits 0 to 9. Each emergency
number is selected through the assigned routing port or through the assigned routing
group. The emergency number is only routed to the port when this is enabled. The
coM.sat IP Basic will route the emergency number on the mobile channel route, even if
the SIM card is missing or not registered in the network.
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Note: If the coM.sat IP Basic is dead, it is not possible to dial the emergency number. The
ANT logic is not applied to the emergency numbers.
Dialling the emergency number in GSM/UMTS network is described as follows:
The coM.sat IP Basic can make an emergency call via the GSM/UMTS module and the SIM
card. Some number strings are selected as emergency numbers that are in the GSM/UMTS
module known as an emergency call message.
These emergency numbers are by coM.sat IP Basic chosen in the following situations:
•
•
•
•
•
6.18
If the SIM card is not inserted.
If the SIM card is is not valid (there are only a few emergency numbers allowed).
If the wrong PIN has been entered.
If the SIM card is barred.
If the GSM/UMTS module is not registered, it will try to make an emergency call over
another available network.
VIP Users
The VIP number list contains numbers which are to undergo a special treatment.
Each line contains a telephone number and an optional associated MSN. The MSN can be used
to define shortcuts for members of the VIP list. If this MSN from any source (such as NT, TE,
SIP or GSM) is selected, the call according to the routing table is set up. A call from a mobile
number will be associated with the MSN for the ISDN call as the calling number.
VIP users also have the right to make outgoing calls on the channels even it is not allowed to
make post dialling. Thus it is possible to allow the post dialling function for a particular user
group. This option 'Answer the incoming call from VIP user' is set in the menu 'System
Parameters'.
VIP callers can also use the routing function. While regular callers are always connected to
the PBX, users can make VIP calls according to the chosen number and programmed routing
calls to landline or via the second GSM channel to other GSM subscriber.
If a VIP party number is called, the CLIP will be always transmitted, even if it is switched off
for the channel. This does not permit the use of call-back processing; otherwise the VIP could
not use anymore the VPBX.
You can program a total of 100 VIP numbers.
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Figure 44: VIP users
•
Index: This is the number through which the other parameters are stored. The index is
not changed.
•
VIP number: The number with a special treatment.
•
MSN number: The short number for the number in the VIP number list.
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6.19
coM.sat IP Basic
Virtual PBX (VPBX)
The functionality of the virtual PBX is not available unless a special license was purchased and
their status in the virtual PBX option can be displayed. Licenses can be requested under
[email protected].
Figure 45: Virtual PBX
The functions of the virtual PBX extend the functionalities of the coM.sat IP Basic with the
following features:
•
•
•
Call Hold
Call Retrieve
Call forwarding
The VPBX functions can be used in combination with a suitable PBX or as a standalone
solution.
For access to these additional services the user must program each of these functions to the
appropriate access code.
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The access code shall consist of a maximum of 4 characters. The numbers from 0-9, and the
*, # characters can be programmed.
By default, the access code for each VPBX function is as below defined:
•
•
•
Call Hold '1 '
Call Retrieve '2 '
Call Forward '3 '
The VPBX services can be activated by the user during the following conditions:
•
•
•
•
•
If calls are received from the NT port or built
If calls from the SIP extension will be accepted or processed
Calls to or from VIP User
When an internal user makes a call or receiving a call from an external user, then the
internal users are able to access the services.
If two internal users are connected, only the calling user is able to use the services.
Note: If two internal users are connected and if the called user is a VIP user, the IP Basic is
programmed so that the user can use these VIP services.
This can be done in the following cases, where the features are activated according to their
requirements:
•
•
For calls from a PBX to the VIP number in NT mode, the called party (VIP) is
permitted, VPBX instead of the caller (PBX) to use.
For calls from a SIP extension to the VIP number, the called party (VIP) is permitted to
use VPBX instead of the caller (SIP extension).
6.19.1
Call Hold
During an active call the dialling of the configured character leads to hold the call. The audio
connection will be separated during a UMTS / GSM / or NT / TE call and held party number
will receive a music on hold. During a SIP connection the INVITE message with a link a =
send only has to be sent. The subscriber, who initiates the hold function hears a dial tone and
is capable to dial an internal or external number. This number is dialled using DTMF tones.
During the dialling, it is not possible to retrieve the call on hold, because the retrieve digit
would be interpreted as a number of the called party. Only if the new number has been
dialled and the call remains in place, it is possible to retrieve, get back or forwards it. If an
error while dialling occurs, the dialling van be cancelled by using the '#' character. If after
holding no further characters during the "First Digit wait timer ' have been dialled, the active
call on the call on hold will be disconnected.
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6.19.2
coM.sat IP Basic
Call Retrieve
A held call can be retrieved if the call on the third party or is connected. In this case, the
retrieved call leads that the third party is disconnected and the audio connection to the held
user is restored.
6.19.3
Call Transfer
A held call can be forwarded, if the call to the third party is connected. In this case, select the
access code for call forwarding by the participants holding means that the holding is
disconnected and that the audio connection is established between the user and the third
held. Call Forwarding is also initiated, when the holding party hangs up.
Note: If the VPBX is used in the TE mode, then the entries of the VIP number list must be
entered in the destination number.
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6.20
coM.sat IP Basic
Licensing
The details of the licensing must be entered in the 'licensing' as follows.
Figure 46: Licensing
•
IMEI Number: As indicated by coM.sat
•
Deadline: As indicated by coM.sat
•
Licence Key: As indicated by coM.sat
•
Product: Please, select the product from the drop down list.
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6.21
coM.sat IP Basic
Voice Messages
Figure 47: Voice Messages
In certain operating situations can be recorded voice announcements to inform the calling
party number during the call setup or forwarding.
Three different announcements can be stored in the coM.sat IP Basic. With regard to these
announcements, please note that this may be a total of not more than 60 seconds.
Furthermore these announcements are stored as'. WAV' files in the format CCITT A-Law 8
bits and mono recording at 8000 samples per second. The announcements are assigned to
the corresponding functions:
•
At an incoming call the caller is informed to dial a number, such as 'Please Dial.
•
At an incoming call the caller is informed that the requested extension of the subscriber
being called, such as 'Calling'.
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coM.sat IP Basic
At an incoming call the caller is informed that the requested extension of the subscriber
is busy and that after pressing the '*' can re-dial such as 'Busy, Re-Dialling with '*'.
6.22
SMS Notification
Figure 48: SMS Notification
The coM.sat IP Basic can send an SMS notification to the party number, if this has been
programmed. The notification can be programmed for the following reasons:
•
•
•
Port Errors
IP address update
Calling Number Transfer
6.22.1
Port Failure using SMS
The following messages are sent, when the user wants to receive an SMS in the event of a
port error. The following notifications are supported:
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•
•
•
•
coM.sat IP Basic
SMS send, if the ISDN NT port link is not working
SMS send, if the ISDN TE port link is not working
Send SMS, if the WAN port link is not working
SMS send, if the WWAN port link is not working
6.22.2
Update IP address using SMS
If the user received a text message to update the IP address like the following messages are
sent:
•
•
IP address of the WAN port
IP address of the WWAN port
The user can activate the above functions as required.
For all port errors and to update the IP addresses for both ports are the parameters for 'error
on SMS' and 'IP addresses update' to be programmed individually:
•
To: The user has the option of specifying any three destination numbers, to which the
SMS should be sent.
Note: Not all three target numbers are needed.
•
SMS Text: The user can review the text to display, as follows:
o Update of coM.sat IP Basic: The WAN port link is not working.
o Update of coM.sat IP Basic: The WAN IP Address is [ipaddr]
Note: Use [ipaddr], where to insert the IP address.
•
Channel: The user can select the channel from which to send the SMS. Two options
are available:
o GSM / UMTS channel 1
o GSM / UMTS channel 2
Note: If you enable the function after the port error occurred, no notification is sent.
6.22.3
Calling Number Transfer
Send phone number if the call could not be established.
If this option is enabled, then an SMS is sent to the destination number of unsuccessful calls.
•
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When will the SMS be sent: The user can program, when the coM.sat IP Basic should
send an SMS. Therefore, the user can set the following options according to its
requirements:
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o Disconnect and send if the call is not answered (the number of seconds to be
programmed in this example are 5 seconds).
o Send only if the character is dialled by the calling party number (the character is
to program e.g. # etc.).
•
Channel to use for SMS: The user can select the following options:
o GSM/UMTS Channel 1
o GSM/UMTS Channel 2
o Same GSM/UMTS Channel as for call
•
Report sending of SMS: The user can program, when the coM.sat IP Basic should
send an SMS. Therefore, the user can set the following options according to its
requirements:
o Just disconnect and send SMS (the number of seconds to be programmed in this
example is 5 seconds).
o Beep, disconnect, send SMS
o Beep, send SMS, report sending success (confirmation tone) or error (Error
Tone), then disconnect
o Send only if the character is dialled by the calling party number (the character is
to program e.g. # etc.).
•
Text to be send as SMS: The user can program the text for the message.
Note: Use [Number], where the number is inserted. If the text is empty, the SMS is
not sent.
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6.23
coM.sat IP Basic
System Parameters
In coM.sat IP Basic certain global parameters are displayed / configured, which refer to the
entire coM.sat IP Basic and not only on a specific port. These parameters are known system
parameters.
6.23.1
General Parameters
Figure 49: Display of the System Parameters
•
Software Revision Version: The current version of the software revision of the
coM.sat IP Basic is displayed in this field.
•
Kernel Date: The date of the current Kernel of the system will be displayed in this
field.
•
System Name: The user can assign the coM.sat IP Basic a name. This name is known
as a 'system name'. This name is meaningful only when two or more coM.sat IP Basic
connected in a LAN network. The system name can contain up to 40 characters. All
ASCII characters are allowed.
By default the field is empty.
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•
Region/Country: The user can select the country / region, in which the coM.sat IP
Basic is installed. By default, this field is selected for Germany.
•
Language: The Web interface of the des coM.sat IP Basic supports eight languages:
o
o
o
o
o
o
English
French
Italian
German
Portuguese
Spanish
The user can select the desired language. The contents of the Web interface are
displayed in the selected language. As the default is 'German' is selected as language.
Note: After changing the 'languages', the user is redirected to the login page.
•
Country Code: The user should enter the appropriate Country Code.
•
Area Code: The user should enter the appropriate Area Code.
•
Companding Type: For voice calls, the compression is used to improve the signal-tonoise ratio (S / N). The amplitude of the analogue signal is digitized by Pulse Code
Modulation (PCM) and compressed before it is transmitted. The coM.sat IP Basic
supports two algorithms for compressing the voice band: A-law and µ-law.
•
Call Release Timer (Minutes): The Call Release Timer is a timer that is used to
unlock the ports that are involved in a call after a specified time if they are not
unlocked for a reason. This timer is loaded when the call expires. This timer is stopped
when one of the call related port is unlocked. The valid range of the call release timer
is between 1 and 999 minutes. The default value is 999 minutes
•
SIP Trunk Group for IP Dialling: Using the drop down menu the 'SIP trunk group'
for routing of IP calls will be selected. If an IP address from a port of the coM.sat IP
Basic is selected, it will be the desired destination via the programmed choice for IP
SIP trunk group routed. The various options can be selected:
o Null
o SIP Trunk 1
o SIP Trunk 2
o SIP Trunk 3
o SIP Trunk 4
o Any SIP Trunk (first free)
o Any SIP Trunk (rotation)
o
By default the option ‘Any SIP Trunk (first free)’ is set.
•
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Play Routing Tone?: During an outgoing call the routing tone signals that the call is
set up.
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Select 'No' to disable the routing tone. By default, this field is set to 'Yes'.
• Return Call Record Delete Timer (Minutes): This is the duration of an entry in the
table RCOC. This timer is activated as soon as a number in the RCOC database is
entered. After this time the entry is deleted.
•
Replace ‘+’ from the CLI? : In most GSM / UMTS networks, the number is
transferred in an international phone number format and displayed. In some cases,
however, the country-specific code character '+' is not processed. The coM.sat IP Basic
provides the ability to remove the '+' and replace it with an appropriate number string.
For example: If the caller ID '+4962046070952' has to be replaced and the number
string '00' is programmed, then the number is converted to 004962046070952. If the
box is an empty string, then the number would be converted to 62046070,952.
•
Replace ‘+’ from the CLI with number string: If the above option is selected, the
desired number string to be replaced with the '+' is entered here.
•
VoIP Silence Disconnect Timer (Seconds): The duration (in seconds) after which
the SIP call is disconnected when a permanent silence, i.e. RTP packets are not
detected for this period. The 'VoIP Silence Disconnect Timer' is started, when silence is
detected during a SIP call. The SIP call is disconnected after the expiry of this timer.
This timer is used for all types of calls that are made via the SIP trunk applied or
received.
The valid range of 'VoIP Silence Disconnect Timer' is between 1 and 999 seconds. By
default the value is 999 seconds.
•
Clock Synchronisation Port: If the PBX has a solid network connection, and these
reports too many errors (frame slips, bit slips), then synchronisation is necessary.
These types of errors are not important for speech communication. However, with most
phone systems with advanced troubleshooting processes, the problem occurs during
operation. Some telephone systems can lead to the shutdown of the 'faulty' port. This
can be avoided by synchronisation. In this case, the synchronisation input of the
coM.sat IP Basic is directly connected to the PBX port. The clock synchronisation is then
derived from this signal. The user can derive the clock synchronisation from the NT /
TE port by selecting the option from the Port Clock Synchronisation Port'. If
synchronisation is selected on the NT port, then use the connected device has a clock
on this port, which is synchronised to another source.
•
Answer Incoming call from VIP user?: If this option is activated, incoming calls are
answered by the VIP users with a dial tone.
•
Terminal (USB Connection): The user must either activate 'GSM / UMTS Transparent
GSM / UMTS channel 1' or the ' GSM / UMTS Transparent GSM / UMTS Channel 2' to
access directly to the GSM / UMTS module.
By default this option is ‘Transparent Mode Off’.
•
Clear Return Call Table: This option allows the user to delete all entries in the table
RCOC. After setting all required parameters, click on the button 'Submit' to save the
changes. In case to reset all parameters click on the button 'Default All'.
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6.23.2
coM.sat IP Basic
Date and Time Settings
Various functions such as call detail records (CDR), the time change etc. use the date and
time parameters for their proper function. For this reason, the programming of the 'date and
time parameters' is one of the most important steps in programming the coM.sat IP Basic. The
coM.sat IP Basic has its own real time clock (RTC), which saves the time and date. This uses a
simple Network Time Protocol (SNTP) to get the Greenwich Mean Time (GMT) from the time
server. Depending on the country of installation the coM.sat IP Basic sets the Real Time Clock
so that it coincides with the period of time server. The coM.sat IP Basic also offers an option to
synchronise the date and time for the system with the SNTP server. This is done
automatically at startup.
6.23.2.1 Real Time Clock
The user can edit the following details:
Figure 50: Date and Time Settings
•
Created:
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Current Date: Enter the current date in the format DD-MM-YYYY. The valid range for
the date is between 01 to 31, for the month of January to December and for the year
of 2009 to 2099.
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•
Current Time: Enter the current time in the format HH-MM-SS. The valid range for the
hour is 00 to 23, for the minutes 00-59 and the seconds from 00 to 59.
•
Current Day: Depending on the current date the coM.sat IP Basic automatically sets
this field to the current day. Click on the link 'Submit' to save the changes.
6.23.2.2 Daylight Saving Time (DST)
The real time clock of the coM.sat IP Basic automatically sets itself in conformance with the
requirements of the country at the time of conversion back and forth. Typically, clocks are set
one hour forward in the spring and one hour back in the autumn. The start and end dates of
daylight saving will vary depending on location and year.
The coM.sat IP Basic provides the option to enable or disable the time change. In most
countries in Asia and Africa as well as certain countries in South America, the time change is
ignored.
Figure 51: Date and Time Settings
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The time change should only be programmed when the coM.sat IP Basic is installed in a
country where the time change is also observed. By observing the time change, the time
difference between countries in different time zones can also be different. The time change
can be activated automatically or by manual programming.
6.23.2.2.1
Manual DST
If the time change is manually activated, then the user must program the parameters
according to the requirements. These parameters are:
•
Day-Month wise: If this option is selected, the user must then enter the following
information:
o Ordinal: Choose the 1st, 2nd, 3rd, 4th or 5 Day of month as ordinal.
o Day: Select a day from Sunday to Saturday as the day on which the time should
be changed.
o Month: Select a month from January to December on which the time should be
changed.
o Change Time from: This is the time from which the time change is to be changed.
The time is specified in 24-hour format and the valid range for the hour is 00-23
and for the minutes from 00 to 59.
o To Time: The time at which to write the time change. The time is specified in 24hour format and the valid range for the hour is 00-23 and for the minutes from 00
to 59.
By default the time change is 'No'. No is selected when the coM.sat IP Basic unit is
installed in a country, where the time change is not observed.
6.23.2.2.2 Scheduled DST
When the scheduled time change is enabled, please select the region from the list in which
the coM.sat IP Basic is installed. All other parameters are set automatically.
6.23.2.2.3 Backward Time Adjustments
Go to 'Backward Time Adjustments' and program the appropriate parameters in the same
manner as above for 'Backward Time Adjustments'.
After the user has set all parameters, the 'Submit' button must be pressed to save the
changes.
If the option 'Default All' is selected, all parameters are reset.
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6.23.2.3 SNTP Settings
Once the RTC is set, these details are presented here.
The user can program the following information:
Figure 52: SNTP Settings
•
Auto Date & Time Sync with SNTP During Power ON? : Select 'Yes', and it is
synchronized date and time with the SNTP server automatically at power up.
By default, this field is set to 'No'.
•
SNTP Server Address: Program the SNTP server address in this field if the option
date and time with the SNTP server to synchronize automatically at power up. The
coM.sat IP Basic will synchronize the date and time with the server whose address was
entered in this field. The SNTP server address may contain a maximum of 40
characters. All ASCII characters are allowed.
•
Time Zone: Select the country / region in which the coM.sat IP Basic is installed.
By default, 'Germany (GMT +01.00)' is selected.
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6.24
coM.sat IP Basic
Network Connection
The device must be connected with ISP, the LAN or the Internet to operate the SIP trunks or
to provide Internet access.
The user can, depending on the requirements, set three priorities. This can be done via the
option ‘Internet Connection using’.
Figure 53: Network Connection
•
Priorities: The priorities can be set as followed:
o Priority 1: The user can choose the required option from a drop-down list. The
Internet connection is set here as a first preference. The options on the dropdown list are as follows:
Non
WAN
GSM/UMTS Channel 1
GSM/UMTS Channel 2
By default WAN is selected.
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o Priority 2: Can be set like Priority 1.
By default GSM/UMTS Channel 1 is selected.
o Priority 3: Can be set like Priority 1.
By default GSM/UMTS Channel 2 is selected.
Note: Two priorities cannot be equal. It is not necessary for the user to set all three
priorities, but priority one is mandatory.
•
Fallback to another network connection when connection is unavailable: If the
network connection goes down for the set priority 1 and the user has activated this
option, the coM.sat IP Basic will automatically switch to the next priority to restore the
network connection.
•
Switch back to priority 1 when connection is available: If this option is enabled,
the coM.sat IP Basic automatically switches back to priority 1, when the Internet
connection is restored in priority 1.
•
The coM.sat IP Basic must monitor the functionality of the Internet connection, this can
be done by setting the following parameters:
o Server to check internet connectivity: The user specifies the public IP address /
domain name such as 'Google.com' one. The coM.sat IP Basic would ping this
address programmed to check whether the Internet connection is present or not.
As the default is 'google.com' is selected
o Interval: The user defines the times at which the public IP address / domain
name from IP Basic is pinged. By default the value is 60 seconds.
o Number of retries: The coM.sat IP Basic specifies that the connection is not
working, if the response from the ping is not received for the programmed
number of retries. The default is 1. If the above function is set, the coM.sat IP
Basic automatically pings the server at the above specified time intervals. In this
type the coM.sat IP Basic monitors, whether the links are intact or not intact. The
coM.sat IP Basic can make the link again, either by selecting the next priority, or
by recourse to it.
After setting all required parameters, click on the button 'Submit' to save the
changes.
In the event that it wants the user to reset all the parameters, the option
'Default All' will be selected.
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6.25
coM.sat IP Basic
Call Progress Tones
Figure 544: Call Progress Tones Connection
While setting up the call, tones to the calling subscriber will be issued to indicate the status of
the phone call.
The following tones are audible:
•
•
•
•
•
•
•
•
•
Created:
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Dial Tone
Ring Back Tone
Busy Tone
Error Tone 1
Error Tone 2
Confirmation Tone
Feature Tone / Programming Tone
Intrusion Tone
Routing Tone
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The standards for the ring tones are used for different situations and for different countries.
For example there is according to the ITU standard for Germany from the dial tone 425 Hz
modulated by 25 Hz, and this is for the USA / Canada at 350 +440 Hz without modulation.
The coM.sat IP Basic supports tones for 34 different countries. The user can customize the
sounds to the coM.sat IP Basic call progress to the standards of the countries in which the
device is installed. Use 'notes to Ruffortschritt' the user can edit the different tones and to
Ruffortschritt.
The user can select from the following options:
•
CPTG Type: If the option 'CPTG type' is selected, then the combo box to select the
country will be edited. Select the country code in which the coM.sat IP Basic is installed
(See also CPTG table, which appears at the end of this chapter for the codes of
different countries). If you select this option, you can no longer edit the Call Progress
Tone Cadence table. By default, 11 (code of Germany) is selected. In the field
'Supported Country' the supported country will be displayed.
•
Customized:
If 'Customized' is selected, then you can edit the combo box (by country) and not the
Call Progress Tone Cadence table. The frequency and cadence for each tone can be
programmed.
The default values for all the tones corresponding to the last selected country are
displayed.
The various parameters of the Call Progress Tone Cadence table are as follows:
•
Frequency 1 (Hz): Frequency 1 for all tones in this field. The frequency 1 range is
300-1400 Hz for all tones.
•
Frequency 2 (Hz): Frequency 2 for all tones in this field. The frequency range is 201400 Hz for all tones. This activated and editable if the ‘Operator’ has been selected.
•
Operator: The operator parameters consist of the following settings:
o No: If ‘No’ has been programmed, frequency 2 is not applicable.
o Modulation: If ‘*’ (modulation) has been programmed, frequency 1 and
frequency 2 can be used as modulation in the form F1 * F2.
o + (additional): If additional (+) has been programmed, frequency 1 and
frequency 2 will be added in the form F1 + F2.
•
Cadence: Cadence ON Time 1-OFF Time 1, ON Time 2-OFF Time 2T and ON Time 3OFFTime 3 for all tones. The valid range for ON time and OFF time for all tone is:
0000-9999 msec. Click on the link 'Submit' botton to save the changes. The button
'Default All' will be used to reset all parameters.
Note: Resetting of the coM.sat IP Basic or its modification will not reset the tone for Call
Progress. Even the country is not reset. The tone for hold is fixed for all countries and is not
programmable.
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The default values for all tones for call progress for all countries according to the ETSI
standards are programmed in the system. The table with the default values for all the call
progress tones can be found below:
CB
TG
T
Y
P
E
Country
Dial tone
Ring tone
Busy tone
Error tone 1
Error tone 2
Freq.
Cad.
(Sec.)
Freq
Cad.
(Sec.)
Freq.
Cad.
(Sec.)
Freq.
Cad.
(Sec.)
Freq.
Cad.
(Sec.)
1
CPTG Type 1
440
Continu
ous
Tone
350+
440
0,4on
0,2off
0,4on
2,0off
440
0,75on
0,75off
440
0,25on
0,25off
440
1on
1off
2
CPTG Type 2
400
Continu
ous
Tone
400
0,6on
0,2off
0,2on
2,0off
400
0,5on
0,5off
400
0,25on
0,25off
400
1on
1off
3
CPTG Type 3
350+
440
Continu
ous
Tone
440+
480
2,0on
4,0off
480+
620
0,5on
0,5off
440
0,25on
0,25off
440
1on
1off
4
Argentina
425
Continu
ous
Tone
425
1,0on
4,0off
425
0,3on
0,2off
425
0,3on
0,4off
425
1on
1off
5
Australia
425*2
5
Continu
ous
Tone
400*2
5
,4on
,2off
,4on
2,0off
425
0,375on
0,375off
425
0,375on
0,375off
425
1on
1off
6
Brazil
425
Continu
ous
Tone
425
1,0on
4,0off
425
0,205on
0,25off
425
0,25on
0,25off
425
1on
1off
7
Canada
350+
440
Contin
uous
Tone
440+
480
2,0an
4,0aus
480+
620
0,5on
0,5off
480+62
0
0,25on
0,25off
480+62
0
1on
1off
8
China
450
Contin
uous
Tone
450
1,0on
4,0off
450
0,35on
0,36off
450
0,7on
0,7off
450
1an
1aus
9
Egypt
425*5
0
Contin
uous
Tone
425*5
0
2,0on
1,0off
425*5
0
1,0on
4,0off
450
0,5on
0,5off
450
1an
1aus
10
France
440
Contin
uous
Tone
440
1,5on
3,5off
440
0,5on
0,5off
440
0,25on
0,25off
440
1on
1off
11
Germany
425
Contin
uous
Tone
425
1,0an
4,0aus
425
0,48on
0,48off
425
0,24on
0,24off
425
1on
1off
12
Greece
425
425
1,0on
4,0off
425
0,3on
0,3off
425
0,15on
0,15off
425
1on
1off
13
India
400*2
5
Contin
uous
Tone
400*2
5
,4on
,2off
,4on
2,0off
400
0,75on
0,75off
400
0,25on
0,25off
400
1on
1off
14
Indonesia
425
Contin
uous
Tone
425
1,0on
4,0off
425
0,5on
0,5off
425
0,25on
0,25off
425
1on
1off
15
Iran
425
Contin
uous
Tone
425
1,0on
4,0off
425
0,5on
0,5off
425
0,25on
0,25off
425
1on
1off
16
Iraq
400
400
Continu
ous
Tone
400
1,0on
1,0off
400
0,25on
0,25off
400
1on
1off
17
Israel
400
400
1,0on
3,0off
400
0,5on
0,5off
400
0,25on
0,25off
400
1on
1off
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0,2an
0,3aus
0,7an
0,8aus
Contin
uous
Tone
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18
Italy
425
Contin
uous
Tone
425
1,0on
4,0off
425
0,5on
0,5off
425
0,25on
0,25off
425
1on
1off
19
Japan
400
Contin
uous
Tone
400*2
5
1,0on
2,0off
400
0,5on
0,5off
400
0,25on
0,25off
400
1on
1off
20
Kenya
425
Contin
uous
Tone
425
0,67on
3,0off
1,5on
5,0off
425
0,2on
0,6off
0,2on
0,6off
425
0,2on
0,6off
425
1on
1off
21
Korea
350+
440
Contin
uous
Tone
440+
480
1,0on
2,0off
480+
620
0,5on
0,5off
480+62
0
0,3on
0,2off
480+62
0
1on
1off
22
Malaysia
425
Contin
uous
Tone
425
0,4on
0,2off
0,4on
2,0off
425
0,5on
0,5off
425
2,5on
0,5off
425
1on
1off
23
Mexico
425
Contin
uous
Tone
425
1,0on
4,0off
425
0,25on
0,25off
425
0,25on
0,25off
425
1on
1off
24
New Zeeland
400
Contin
uous
Tone
400+
450
0,4on
0,2off
0,4on
2,0off
400
0,5on
0,5off
400
0,25on
0,25off
400
1on
1off
25
Philippines
425
Contin
uous
Tone
425+
480
1,0on
4,0off
480+
620
0,5on
0,5off
480+62
0
0,25on
0,25off
480+62
0
1on
1off
26
Pole
425
Contin
uous
Tone
425
1,0on
4,0off
425
0,5on
0,5off
425
0,5on
0,5off
425
1on
1off
27
Portugal
425
Contin
uous
Tone
425
1,0on
5,0off
425
0,5on
0,5off
450
0,33on
1,0off
450
1on
1off
28
Russia
425
Contin
uous
Tone
425
0,8on
3,2off
425
0,4on
0,4off
425
0,25an
0,25aus
425
1on
1off
29
Saudi Arabia
425
Contin
uous
Tone
425
1,2on
4,6off
425
0,5on
0,5off
425
0,25on
0,25off
425
1on
1off
30
Singapore
425
Contin
uous
Tone
425*2
4
0,4on
0,2off
0,4on
2,0off
425
,75on
0,75off
425
0,25on
0,25off
425
1on
1off
31
South Africa
400*3
3
Contin
uous
Tone
400*3
3
0,4on
0,2off
0,4on
2,0off
400
0,5on
0,5off
400
0,25on
0,25off
400
1on
1off
32
Spain
425
Contin
uous
Tone
425
1,5on
3,0off
425
0,2an
0,2aus
425
0,25on
0,25off
425
1on
1off
33
Thailand
400*5
0
Contin
uous
Tone
400
1,0on
4,0off
400
0,5on
0,5off
400
0,3on
0,3off
400
1an
1aus
34
Turkey
450
Contin
uous
Tone
450
2,0on
4,0off
450
0,5on
0,5off
450
0,2on
0,2off
0,6on
0,2off
450
1on
1off
35
VAE
350+
440
Contin
uous
Tone
400+
450
0,4on
0,2off
0,4on
2,0off
400
0,375an
0,375au
s
400
0,4on
0,35off
0,225on
0,525off
400
1on
1off
36
United Kingdom
350+
440
Contin
uous
Tone
400+
450
0,4on
0,2off
0,4on
2,0off
400
0,375on
0,375off
400
0,4on
0,35off
0,225on
0,525off
400
1on
1off
37
USA
350+
440
Contin
uous
Tone
440+
480
2,0on
4,off
480+
620
0,5on
0,5off
480+62
0
0,25on
0,25off
480+62
0
1on
1off
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38
CPTG Type 4
400
Contin
uous
Tone
400
1,0on
2,0off
400
0,5an
0,5aus
400
0,25on
0,25off
400
1on
1off
39
Belgium
425
Contin
uous
Tone
425
1,0on
3,0off
425
0,5on
0,5off
425
0,167on
0,167off
425
1on
1off
40
CPTG Type 5
350+
400
Contin
uous
Tone
350+
440
0,4on
0,2off
0,4on
2,0off
400
0,75on
0,75off
400
0,25on
0,25off
400
1on
1off
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CB
TG
T
Y
P
E
Country
1
coM.sat IP Basic
Confirmation
Tone
Feature Tone /
Programming Tone
Routing Tone
Freq.
Cad.
(Sec.)
Freq.
Cad.
(Sec.)
Freq.
Rhythm
(Sec.)
Freq.
Cad.
(Sec.)
CPTG Type 1
350+44
0
0,1on
0,1off
350+440
0,1an 0,9off
350+440
0,1on
1,9off
440
0,1on
2,9off
2
CPTG Type 2
400
0,1on
0,1off
400
1,5an 0,1on
400
0,1on
1,9off
400
0,2on
4,8off
3
CPTG Type 3
350+44
0
0,1on
0,1off
350+440
0,1an 0,9off
350+440
0,1on
1,9off
440
0,1on
2,9off
4
Argentine
425
0,1on
0,1off
425
0,1an 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
5
Australia
425*25
0,1on
0,1off
425*25
0,1an 0,9off
425*25
0,1on
1,9off
425
Cont.
Tone
6
Brazil
425
0,1on
0,1off
425
0,1an 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
7
Canada
350+44
0
0,1on
0,1off
350+440
0,1an 0,9off
350+440
0,1on
1,9off
480+
620
0,5on
0,5off
8
China
450
0,1on
0,1off
450
0,1an 0,9off
450
0,1on
1,9off
450
0,2on
0,2off
0,2on
0,6off
9
Egypt
425*50
0,1on
0,1off
425*50
0,1an 0,9off
425*50
0,1on
1,9off
450
0,5on
0,5off
10
France
440
0,1on
0,1off
440
0,1an 0,9off
440
0,1on
1,9off
440
0,1on
2,9off
11
Germany
425
0,1on
0,1off
425
0,1an 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
12
Greece
425
0,1on
0,1off
425
0,1an 0,9off
425
0,1on
1,9off
425
0,15on
0,25off
0,15on
1,45off
13
India
400
0,1on
0,1off
400*25
0,1on 0,9off
400*25
0,1on
1,9off
400
0,15on
4,85off
14
Indonesia
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
15
Iran
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
16
Iraq
400
0,1on
0,1off
400
0,1on 0,9off
400
0,1on
1,9off
400
0,1on
2,9off
17
Israel
400
0,1on
0,1off
400
0,1on 0,9off
400
0,1on
1,9off
400
0,1on
2,9off
18
Italy
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
19
Japan
400
0,1on
0,1off
400
0,1on 0,9off
400
0,1on
1,9off
400*
25
0,1on
2,9off
20
Kenya
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
21
Korea
350+44
0
0,1on
0,1off
350+440
0,1on 0,9off
350+440
0,1on
1,9off
350+
440
0,1on
2,9off
22
Malaysia
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
23
Mexico
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
24
New Zeeland
400
0,1on
0,1off
400
0,1on 0,9off
400
0,1on
1,9off
425
0,1on
2,9off
25
Philippines
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
440
0,1on
2,9off
26
Pole
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
27
Portugal
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,2on
1,4off
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28
Russia
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
29
Saudi Arabia
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
30
Singapore
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,25on
2,0off
31
South Africa
400*33
0,1on
0,1off
400*33
0,1on 0,9off
400*33
0,1on
1,9off
400
0,15on
0,25off
0,15on
1,45off
32
Spain
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
425
0,1on
2,9off
33
Thailand
400*50
0,1on
0,1off
400*50
0,1on 0,9off
400*50
0,1on
1,9off
400
0,1on
2,9off
34
Turkey
450
0,1on
0,1off
450
0,1on 0,9off
450
0,1on
1,9off
450
0,1on
2,9off
35
VAE
350+44
0
0,1on
0,1off
350+440
0,1on 0,9off
350+440
0,1on
1,9off
350+
440
0,1on
2,9off
36
United Kingdom
350+44
0
0,1on
0,1off
350+440
0,1on 0,9off
350+440
0,1on
1,9off
400
0,2on
4,8off
37
USA
350+44
0
0,1on
0,1off
350+440
0,1on 0,9off
350+440
0,1on
1,9off
480+
620
0,5on
0,5off
38
CPTG Type 4
400
0,1on
0,1off
400
1,75on 0,1off
400
0,1on
1,9off
400
0,2on
0,2off
0,2on
2,5off
39
Belgium
425
0,1on
0,1off
425
0,1on 0,9off
425
0,1on
1,9off
440
0,1on
2,9off
40
CPTG Type 5
350+44
0
0,1on
0,1off
350+440
0,1on 0,9off
350+440
0,1on
1,9off
350+
440
0,5on
0,5off
1,0on
5,0off
Note: The frequency designations are as follows:
• F1 * F2: F1 is modulated by F2.
• F1+ F2: the interference of the two frequencies F1 and F2 without modulation
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6.26
coM.sat IP Basic
Call Details Record (CDR) Filters
The user can activate the filters in various combinations to generate many different reports.
The filters are described below:
Figure 55: Filters for the CDR
•
Calls originated from: This filter can be set when the user requires a report based on
calls that are generated by specific port (s) or equipments.
By default, the filters for all ports and devices are enabled.
•
Calls terminated on: This filter can be set when the user requires a report based on
calls that are destined for certain TE port (s) or equipment.
By default, are enabled for all ports and devices.
•
Calls made from: The user selects the date ‘from to’ to create the report.
By default, this value is from 01 May 2010 to current date.
•
Calls made between: The user enters the start time and end time, for example: 9:00
to 23:30 clock.
By default the value is between 00:00 bis 23:59 clock
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•
Call Duration equal to and greater than: The report is created based on the
duration of the call.
By default the value is zero.
•
Called Party Numbers Matching with: In dieser Tabelle sind die Nummern der
angerufenen Teilnehmer für welche der Bericht erstellt wird.
Als Voreinstellung sind alle Nummern programmiert.
•
Calling Party Numbers Matching with: This table shows the numbers of the called
party for which the report is created.
By default, all numbers are programmed.
•
Clear Call Records: The call detail record is deleted only if ‘Clear Call Records’ button
is clicked. If the 'Clear Call Records' button has been clicked, the screen will display an
alert message.
Figure 56: Clear Call Records
Click on the 'OK' button to delete all call records.
After the call records were deleted, the field ‘Calls made from’ changes ' to the date, on
which the records has been deleted.
After the user has activated the filter according to the requirements, the reports can be
displayed by clicking the option ‘Call Detail Records report’.
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6.26.1
coM.sat IP Basic
Call Details Record (CDR) Report
Figure 57: CDR Report
The coM.sat IP Basic facilitates the creation of various reports by the user with the required
settings. These reports may be necessary for various reasons, e.g. to cut costs, increase
productivity, security and data protection.
The coM.sat IP Basic supports the default filter to display the reports, e.g. Report for calls for
certain duration or for calls only from certain ports or for calls with specific dates, where the
calls have been made or received etc.
The coM.sat IP Basic supports up to 2000 entries for call records and these records are stored
by first in first out (FIFO) method. On one page, a total of 20 records are displayed. To view
the next 20 records, please click on the button 'Next'. On the first page the buttons 'First
Page' and 'Back' are disabled. On the last page, the buttons 'Last Page' and 'Next' is disabled.
Click on the buttons 'Back', 'First Page', 'Next' and 'Last Page' to check all the call records.
The alarm message ‘No calls to display’ appears on the screen, when no records are
available.
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Note: The filter parameters are not applicable to store the call records in the system. If the
coM.sat IP Basic is set to the default settings, the records are not deleted the call details.
The feature call details record will be used to prepare a report on call details record with the
following parameters:
Date
Start time
Calling Number
Called Number
Bill
Duration in seconds
From Port
To Port
AOC
Disconnected by
Cause
Remarks
•
•
•
•
•
•
•
•
•
•
•
•
Sr.
Nr.
Date
Start
time
Calling
Number
Called
Number
Bill
Duration
(sec.)
From
Port
To
Port
AOC
Disconnected
by
Cause
Rema
rks
1
29.1.2000
0:52
[email protected].
1.1 13
2526
+
62
SIP
EXT7
NT
BCH
1
0
SIP EXT7
Normal
call
clearing
(16)
N
2
29.1.2000
0:58
[email protected].
1.1 13
123556
-
0
SIP
EXT7
NT
BCH
255
0
NT BCH255
Call
rejected
(21)
U
3
29.1.2000
1:13
[email protected].
1.1 13
9879095390
-
0
SIP
EXT7
NT
BCH
2
0
SIP EXT7
Normal
call
clearing
(16)
U
4
29.1.2000
1:27
[email protected].
1.1 13
9879095390
-
0
SIP
EXT7
NT
BCH
2
0
SIP EXT7
Normal
call
clearing
(16)
U
5
29.1.2000
1:29
[email protected].
1.1 13
1
+
22
SIP
EXT7
NT
BCH
1
0
SIP EXT7
Normal
call
clearing
(16)
N
6
29.1.2000
1:30
[email protected].
1.1 13
1
+
29
SIP
EXT7
NT
BCH
1
0
SIP EXT7
Normal
call
clearing
(16)
N
7
29.1.2000
1:28
[email protected].
1.1 13
9879095390
+
33
SIP
EXT7
GSM
CH1
1
SIP EXT7
Normal
call
clearing
(16)
N
8
29.1.2000
1:30
9879095390
1
+
35
GSM
CH1
NT
BCH
1
0
NT BCH1
Normal
call
clearing
(16)
AT
9
29.1.2000
2:58
[email protected].
1.1 13
192.168.1.1
3
+
19
SIP
EXT7
SIP
TRK
1
0
SIP EXT7
Normal
call
clearing
(16)
N
10
29.1.2000
2:57
[email protected].
1.1 13
9879095390
+
89
SIP
EXT7
GSM
CH1
2
SIP EXT7
Normal
call
clearing
(16)
N
…
Figure 58: Connection data
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6.27
Maintenance
6.27.1
System Debug
Troubleshooting is a method that is used to record events and processes of the system.
Search errors are principally records of system and network activities. The troubleshooting
has certain advantages, which include fault detection, security and system administration.
Debugging messages are sent to the remote server on the IP network to determine the
number of errors or defects in the system or to reduce errors. Syslog is one of the protocols
used to send the debugging messages on the IP network. This is a client / server protocol
that uses UDP as a transport protocol for the debugging process. The coM.sat IP Basic has a
built syslog-client that allows sending debugging messages.
Figure 59: Settings for the System Debug
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6.27.1.1
coM.sat IP Basic
Host Debug Parameters
•
Debug Enable: If this option is enabled, the Debug Search Parameter is enabled.
By default this field is In der Voreinstellung ist das Feld deaktiviert.
•
System Debug: If ‘Debug Enable’ option is enabled, then the user can select the
troubleshooting from the following options:
o
System:
Call
Config
Media Channel
VoPP Packet Recording (Power On)
Time
Web Interface
o SIP-Port:
SIP
SIP Message
STUN / NAT
o GSM/UMTS Port 1/2:
The user can activate this depending on the requirements.
o NT-Port:
Level 4
DTMF
D-Channel
o TE-Port:
Level 4
DTMF
D-Channel
By default, all the above troubleshooting steps on, except VoPP Packet Recording
(power on), and NT Port TE Port.
Note: When debugging is enabled, it should at least show an error level can be
selected. If no debug level is selected, the coM.sat IP Basic displays the following error
message: 'Please select debug level'.
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6.27.1.2 Host Debug Search Parameters
•
VoPP Packet Recording IP Address: Enter the Syslog server address where the
VoIP packets are recorded.
•
Syslog Server Address: Enter in this field, the Syslog server address when debugging
is enabled.
It is only possible to program the IP address. This can be max. 15 characters. The valid
range for the syslog server address is between 000 to 255, and dot (.). An empty field
is allowed as well.
•
Server Port: Enter the syslog server port in this field. The valid range of server ports
is between 514, 1024 to 65535
The default is 514
6.27.1.3 Globale ISDN-Parameter:
Level and code for the ISDN port as follows:
•
•
Level 1
Not
used
Run
Code
Meaning
001
Primitives
002
Status
004
Variable
008
SVC Primitives
016
LAP
032
NLS
064
Error search run
000
Error search off
LAP SVC
Primitives
Variable
Status Primitives
Level 2
Not
used
Created:
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NLS
Not
used
Not
used
Not
used
Code
Meaning
001
Netzwerk Schnittstelle (NS)
002
OS Task
004
ISDN Terminal
000
Error search off
Not
used
OS Task
ISDN
Terminal
NI
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The coM.sat IP Basic supports the cleanup of the parameters (debug codes) depending on the
cleanup stage.
Note: Cleanup parameter 1, cleanup parameter 2 and cleanup parameters 3 are reserved for
future use and are ignored at this time.
6.27.2
PCAP Trace
The PCAP Trace is a built-in function in the coM.sat IP Basic that is used to solve problems
related to SIP trunk with the help of volatile packages.
Figure 60: PCAP Trace
The following parameters must be programmed here:
•
Created:
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Filter Settings: Set the filter according to one of the packages that need to be
recorded. The filter settings can be up to 60 characters long. All ASCII characters are
allowed. By default the field is empty.
An empty field means that all packets are captured.
Examples of filter settings:
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o Capture only SIP Traces: Filter settings = port 5060, where 5060 is the SIP port
number for which the error logs are recorded.
o To capture packets that were transmitted by the system: i.e. from the IP address
192.168.1.191: 192.168.1.191 src = filter settings
o To capture packets that were received for the system: i.e. to the IP address
192.168.1.191: 192.168.1.191 dst = filter settings
o To capture only the packets that were transmitted by the system or the system
received: i.e. from the IP address 192.168.1.191: filter settings = src or dst
192.168.1.191 or 192.168.1.191
•
Enable Promiscuous mode: Enable this filed to activate the promiscuous mode.
If this field is enabled coM.sat IP Basic will capture the entire traffic.
Is the promiscuous mode disabled, the coM.sat IP Basic only to collects from or through
the host routed traffic.
By default, the promiscuous mode is disabled.
Press the Start button to start capturing the packets. The coM.sat IP Basic will stop
collecting IP packets if the 'stop' button is pressed or the maximum amount of space
allotted is used.
Note: The process continues to capture the packets, even if other pages are clicked
on.
Once the capturing of packages is finished, please save the trace file (the captured
packets through the system) by clicking the button 'Save Trace File ".
o It now opens the dialog box 'File download'.
o Click on the button 'Save'. Choose the appropriate location to store the trace file
and press 'Save'.
o The trace file is stored at the specified location.
o To view the saved file, click 'Open'.
Note: To open the trace file, the corresponding software (Ethereal or Wireshark), with
the trace file can be opened, must be installed on your computer.
•
Created:
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Last Status: After the process is completed to capture the packets, the status of the
process in the following fields is displayed.
o
Captured packets: The number of captured packets is displayed in this field.
o
Total Bytes: The total bytes recorded are displayed in this field.
o
Status: This field displays the current capture status of the packages.
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6.27.3
coM.sat IP Basic
Default the System
The coM.sat IP Basic allows the user to make the default settings of the device and the default
values for all programmable parameters using the option 'default settings of the system'
results.
Thus, all parameters are set to their factory settings with the exception of the following
values:
• Call detail records
• Date and Time
• Tones
• SIM PIN (GSM / UMTS port parameters)
• IP address
If this option is selected, a window appears with an error message with the following
warning:
‘This option will assign Default values to all the programmable parameters and will restart. Do
you want to continue?’ ‘OK’ / ‘Cancel’.
Figure 61: Default Settings
Click on the 'OK' button. The coM.sat IP Basic will now restart. All programmable parameters
with the exception of the above are now loaded with the defaults values.
6.27.4
Soft Restart
Sometimes it is necessary for the user to restart the coM.sat IP Basic. It is often difficult to get
to the installation location of the coM.sat IP Basic to restart it. For this reason it is possible to
restart the coM.sat IP Basic via the Web interface.
The restart of the coM.sat IP Basic has followed the same effect as turning off the power
followed by a new switch on. This option is used by the user for a soft restart.
If the user has clicked on the option 'soft restart, a warning message will appear on the
screen: ‘This will restart system. Do you want to continue?’.
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Figure 62: Soft Restart
Click on the 'OK' button to restart the coM.sat IP Basic, otherwise click 'Cancel'.
Note: The restart of the coM.sat IP Basic will not change the programming parameters.
However, all active calls are disconnected and the ports released.
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6.27.5
coM.sat IP Basic
Password Change
The coM.sat IP Basic can only be configured via the web interface.
Figure 63: Password Change
To program the coM.sat IP Basic, the user must first enter the password using the
programming mode. The password consists of four characters.
By default, the password is '1234 '.
After selecting the option to change the password, the following message will be displayed:
•
Enter New Password: Here the new password will be entered.
•
Confirm New Password: Enter the new password for confirmation.
To save the password change, click on the button ‘Submit’.
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The user can reset the password if he has forgotten this. This is done by changing the jumper
position. To do this, please follow the steps below:
•
•
•
•
•
•
•
Swith off the coM.sat IP Basic.
Find the mini-jumper J10 on the PCB.
Change the jumper position from BC to AB.
Switch on the coM.sat IP Basic and wait until it starts to initialize.
Switch off the coM.sat IP Basic and put the jumper back into its original position.
Switch the coM.sat IP Basic again on.
The password is now reset to 1234.
Note: If the password is by changing the jumper settings reset to default, the following
parameters will be reset back to default.
•
•
•
•
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Network Connection (Static)
IP address (192.168.1.111)
subnet mask (255.255.255.0)
VLAN
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6.28 Upload/Download
If an upgrade of the software should be necessary, the coM.sat IP Basic provide the possibility
to update the software with the new version by clicking a button.
Figure 64: Upload / Download
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Glossar
GSM
Global System for mobile Communication
UMTS
Universal Mobile Telecommunication System
SIM
Subscriber Identity Module (Chip card in the mobile)
SMS
Short Message Service
SMSC
Short Message Service Centre
ISDN
Integrated Services Digital Network (Digital Telephone Standard)
TC System
Private Telecommunication Switch over System
NT-Mode
Network Termination; In this case the device as a network termination
(NT) is operated, be aligned with both the electrical and physical
parameters (layer 1), as well as for the layers 2 and 3, which are
responsible for data link and addressing.
TE-Mode
Terminal Equipment; the device is in this case as a TE device, i.e. as
an ISDN-capable device on the IP Basic operated.
PMP
Point-to-Multi-Point, point-to-multipoint connection is the variant of
the configuration for the S0 bus. This can have several (max. 8)
terminals are connected to the same port. Of these 8 devices can
establish, a connection each 2 simultaneously.
AOC
Advice of Charge: Feature of EDSS1. Display of the connection fees as
a bargaining unit according to the classification of the operator during
and at the end of a link that has been.
RJ45
With RJ45 the eight-pin plug technique is called, which has a very
simple but effective operating structure. This connector system is used
in the ISDN S0 port. The connector is standardized in ISO 8877.
MSN
Multiple Subscriber Number -> Multiple Numbers in Point-ToMultipoint mode.
CLIP
Calling Line Identification Presentation; displays the number of the
caller.
CLIR
Calling Line Identification Restriction; suppresses the display of the
CLIP.
CFU
Call Forwarding Unconditional
BAOC
Barring All Outgoing Calls
BOIC
Barring All International Calls
BAIC
Barring All Incoming Calls
COLP
Connected Line Identification Presentation; displays the number of
the called party number
DDI
Direct Dial In
BRI
Basic Rate Interface
ANT
Automatic Number Translation; translates the dialled number string
into a number that is understood by the network to achieve the
desired target
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coM.sat IP Basic
CPTG
Call Progress Tone Generator, generates dial tone, ring tone busy tone
and so on
NAT
Network Address Translation; translation of the network address
STUN
Session Traversal Utilities for NAT; tools for NATs
CoS
Class of Service
ToS
Type of Service
QoS
Quality of Service
SIP
Session Initialization Protocol; network protocol for set up, controlling
and disconnect a communication session between two or more
participants
RTP
Real Time Protocol
TCP
Transmission Control Protocol
UDP
User Datagram Protocol network protocol
LAN
Local Area Network
WAN
Wide Area Network
WWAN
Wireless Wide Area Network
VLAN
Virtual Local Area Network
ICMP
Internet Control Message Protocol; supports the exchange of
information and error messages via the Internet Protocol
100rel
A user wishes to use reliable, tentative answers can be triggered by
the 100rel mark that is required for the value of the initial signaling
header added.
PRACK
Provisional Response Acknowledgement - SIP PRACK method
guarantees reliable and orderly delivery of provisional responses in SIP
SNTP
Simple Network Time Protocol
RTC
Real Time Clock
Created:
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File:
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Manual coM.sat IP Basic

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