Getting Started

Change Your Password!

The first thing to do is to login with the default credentials on your device:

• Username: admin

• Password: admin

Please make sure to change the default credentials, to avoid malicious attacks. You can change the admin password at Preferences → Security or by clicking on the link of the warning message.

The password has to have at least 5 characters, containing at least: one uppercase, one lowercase, one special character (e.g. !#*+!§$%&/()=?) and one number.

The Activate Button / Graceful Activate

Changes made in the GUI do not apply immediately. Depending on the type of change different kinds of "activate" buttons appear:

Only after clicking on the activate button, the changes apply. It is useful to make the needed changes and activate them all at once the configuration is done.

If there are currently active calls going on, you can activate graceful. This will periodically check for active calls and will perform the activate or restart as soon as there are no active calls left.

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In the you can quickly change between the GUI-Mode, the Language and have the status of your beroCloud registration.

• You can visit your beroCloud settings by clicking on the link to beroCloud or in the navigation bar above.

• You can change the GUI-Mode to Advanced, giving you more specific options throughout your gateway.

• You can choose between English or German.

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Configuration path

To configure a beroNet Gateway there are a couple of steps to do. In most cases the configuration starts at Hardware, where the physical parameters of each port are configured. The next step is to group the PSTN ports in PSTN that need to be used. Then SIP Accounts are created under SIP. Finally the different entities are connected in the Dialplan. The Dialplan defines, which calls should be forwarded from SIP to PSTN and vice versa, based on criteria like the dialed number or the CallerID. So the common steps are:

1. Hardware Settings

2. PSTN Groups

3. SIP Accounts

4. Dialplan Rules

Optionally one of the Configuration Wizards can be used for many scenarios. 

Call Handling

Calls are handled by the ISGW service, which is configured through the WebGUI or through provisioning. The ISGW communicates to the hardware through drivers. Every module has a separate hardware driver which has different settings depending on the type of technology. These settings can be configured in the Hardware section.

The physical connections are called ports. ISGW needs to initialise all the ports it should use, this can be configured by putting the ports into PSTN groups.

The next step is to create SIP Accounts and finally the Dialplan rules define how the calls should be routed by ISGW.

PSTN → Hardware

The modules are connected on the beroNet device to a flexible TDM Backplane, which allows sharing of synchronization clocks and bridging of audio-channels between modules and across devices which are connect via the PCM cable.

The hardware section in the webGUI allows to understand which modules are present, to change the synchronization settings and to change technology related settings. 

Here you can see the graphical presentation of the interfaces (in this case BFT-Adapters are required to access all the technologies) and the types of modules that are installed.
Every change in the hardware section requires a red Activate which yields to a reboot of the device.

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Synchronization

It is critical to understand how the synchronization between the different ports, modules and devices should be configured to achieve the highest possible quality for the use case.

In most cases there should be a single synchronization source for one or more beroNet devices. The best results are achieved when receiving the synchronization from an external ISDN Provider through one BRI or PRI port. This synchronization can be shared with the other ports on the same module and with the other modules on the devices and also with another device and its modules via the PCM cable.

If no external synchronization is available, each module can create its own timing. Also in this scenario one module should be chosen as timing source and all the other modules and devices should receive their synchronization from this module to provide a shared timing.

Only in very rare cases it might be necessary to have separate synchronization sources on the same device with 2 modules. This might be the case when 2 different ISDN providers are used which have separated timing sources.

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TDM Clock Master: 

Defines which module should be clock master for all other module slots All other modules will derive their timing from the clock master.

• ALL - each module is clock master for itself. 

• PCM - the external PCM Connector is clock master, so another beroNet gateway provides the timing.
  Thus no synchronization port needs to be defined here. 

• lif0-XXX - the first module is the clock master.

• lif1-XXX - the second module is the clock master.

• lif2-XXX - the third module is the clock master.

The FXS, FXO, LTE and GSM modules can not derive their timing by their interfaces, because these technologies do not provide timing sources. These modules always have a local oscillator to create a timing if they are configured to be TDM Master.

The ISDN modules can receive their timing from the ISDN Provider. Since they have multiple ports it is necessary to define which port should be used a primary synchronization source.
Alternatively they can be set to Crystal, therefore giving their own timing. 

ISDN Settings

Type: Each ISDN Port can be configured separately. The port can be set to te or nt mode.

The type defines whether the device will be te to be connected to an ISDN provider, or whether the port should be nt to be connected to a traditional ISDN PBX. The TE mode is mostly used when an IPBX should be connected to the ISDN network and the nt mode if a traditional PBX needs to be connected to a VoIP Provider. When switching the port from te to nt, the module switches the transmit and receive pins, so that no cross-over cable is required.

Protocol: The protocol can be PTP (Point-to-Point) or PMP (Point-to-Multipoint).

The other settings are more specific and do not require changing in most cases.

The Line Termination can be enabled and a permanent layer 1 timer can be started. 
Furthermore you can set the ISDN call reference value IE to have a length of 2 instead of 1 with CRLEN2 and with EXTCID, you can set the ISDN Caller-ID IE to a length of 3 instead of 1.

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Analog Settings

FXS

When using FXS analog modules (i.e. BF4FXS) you have the following option:

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You can choose between 49V and 89V ring voltage.

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FXO

When using FXO analog modules (i.e. BF4FXO) you can set each country with own settings in terms of signals and tones.
(For Germany 1TR110_DE is often used.)

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PCM Settings

When the PCM Master bridging is enabled the gateway is the PCM-Master (That means this module will generate the clock for all other slaves).
You need to connect a PCM-Slave beroNet Gateway with a PCM cable via the PCM connector to enable the bridging between master and slave.

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On the slave you need to set the Master Settings on all line interfaces to "slave". Then paste the IP-address of the master beroNet Gateway into the slave its PCM-Master IP-address field.

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PSTN

The menu point PSTN gives you an overview about the ports which are provided by the installed modules you are using on this particular beroNet device.

For each technology you will find a sub menu point, like

• ISDN PRI

• ISDN BRI

• ANALOG FXO

• ANALOG FXS

• LTE

• GSM

The sub menu entries are dynamic, and you will only see these which are provided by your modules. These sub menu points or technologies can be grouped together in so called Port-Groups.

As you can see in the picture below, you can add, modify and delete 'Port-Groups' by clicking on the corresponding button.

You can also find the Hardware page here, which is documented in the part above.

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ISDN PRI/BRI options

General ISDN PRI/BRI settings

The picture below shows you ISDN PRI / BRI 'Port-Groups' specific basic settings:

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Group Name
Unique name of the 'Port-Group'

Ports
Ports should be added to the 'Port-Group'

ChanSel
Channel Selection schemes: (Standard / Random / Round Robin) default: Standard
Standard - selects the next free channel in ascending order
Random - selects the next free channel at random
Round Robin - selects the next free channel base on the Round-Robin principle

Tone
ISDN Tones are categorised by country

Interdigit timeout
For every incoming call a interdigit collect timer will be started. After this specified timeout, without getting a digit, the call will be processed to the Dialplan.
Note: This Timer is only started if 'Overlap Dialing' is deactivated. (Default: 3 sec.)

Interdigit timeout initial
This Timer is the initial interdigit timer, that means before we got any digit. This Timer will be stopped after the first digit and the above mentioned Interdigit Timeout Timer will apply.
During this time a dial-tone will be generated. Note this Timer is only started if 'Overlap Dialing' is deactivated. (Default: 15 sec.)

Overlap Dialing
This Option will activate real Overlap Dialing, for instance in ISDN Environments. By activating this option the 'Interdigit' as well as the 'Interdigit timeout initial' Timer will be deactivated.

QSIG support
Enable or disable QSIG support

Link Down behaviour
In some countries like Cyprus the behaviour of ISDN PTP ports regarding Layer1 and Layer2 is different. They deactivate Layer 1 and Layer 2 after a while of inactivity.
With this option you can solve this issue.

Nothing - default
Pull Link Up (2s) - will try to get UP Links up to 2 seconds
Pull Link Up (once) - will try to get UP Links once

Country code
Country calling code with leading 00 or +, e.g. 0049 +49 for calling Germany
This field is required if you set new_source_auto for oad (Caller-ID)

City code
City calling code with a leading 0, e.g. 030 for calling Berlin
This field is required if you set new_source_auto for oad (Caller-ID)

Local area code
Local calling code e.g. 2593890 calling beroNet
This field is required if you set new_source_auto for oad (Caller-ID)

Pcmlaw
Codec setting
Default: using alaw except T1
alaw: using G711-alaw
ulaw: using G711-ulaw

Advanced ISDN PRI/BRI configuration (More)

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EC
This will activate or deactivate the onboard Hardware Echo canceler. default: on

EC tail length
EC tail length [0=8ms,1=16ms,2=24...,15=128ms] (default value is 15=128ms) 16 tabs each 8ms.

Bearer
ISDN Bearer capability to use for outbound calls on this 'Port-Group' speech (default for standard Voice calls) Audio_3.1K (useful for outbound Fax calls) Audio_7K Video Digital_Unrestricted (useful for ISDN digital calls) Digital_Restricted Digital_Unrestricted_Tones

Call Deflection
Call deflection is used to redirect calls on the provider site without using B-Channels on the berofix. If 'Call deflection' is enabled you can use SIP 302 'Move temporarily' to redirect the call on the provider site.

CLIR on OAD
useful for dynamically hide the CallerID in direction of this 'Port-Group' (ISDN). For instance if before detects a call to this 'Port-Group' at which the OAD corresponds with CLIR_on_OAD (after the call left the dialplan), the CallerID will be hidden. That means the remote end doesn't see it. default: empty

Redirected Nr
Defines what the Redirected Nr field should contain. This is usually not needed. (Default: none)

Dialplan Source
The Dialplan Source is used as 'Source' for matching in the 'Dialplan'. That means if a call is initiated from this 'Port-Group' you can use 'Dialplan Source', to tell the Dialplan, which value should be used for the 'Source' in the Dialplan.

Dialplan Source can have the following values OAD:

• auto (This setting checks will automatically choose the settings it thinks is best. Usually oad, but in the case that it is empty, it will choose one of the other)

• oad (default)

• oad2 (oad2 in case you have 2 oad's you can choose with which value you want to use to match with the Dialplan)

• Qsigname (Qsigname to use it at source in the Dialplan)

• Redirected_nr (redirected number)

DAD-Settings

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The dad (Callee ID) gives you the possibility to tell the beroNet Gateway, which field should be used for the DAD for calls to this 'Port-Group'. DAD (Callee ID) can use the the following fields:

dad(Type of number)
It's the 'Type of Number' in terms of ISDN for the Destination Address (DAD). The option defines the number format of the DAD for an outgoing call. Be aware that the remote end has to also support this feature. unknown, international, national, local, subscriber alphanumeric, abbreviated

OAD-Settings

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The oad (Caller ID) gives you the possibility to tell the beroNet Gateway, which field should be used for the OAD for calls to this 'Port-Group'. OAD (Caller ID) can use the following fields:

oad (Type of Number)
Type of Number'TON' is in terms of ISDN for the Originating Address (OAD). This options defines the number format of the OAD for an outgoing call. If you want to use 'CLIP_NO_SCREENING' you have to set this to international, national local subscriber alphanumeric, abbreviated depending on how you are going to send your OAD.

A deeper explanation on how these ISDN and SIP attributes are connected and what they do can be found here: How to modify what goes into From/P-Preferred-Identity/Callerid.

screening/presentation
screening/presentation these are the exact ISDN screening and presentation indicators. default: off screening: off and presentation: off means the callerID is presented but not screened (the remote end does see the callerID) screening: on and presentation: on means callerID presented but screened (the remote end does not see the callerID)

Additional configuration options

The below Box 'Additional configuration options description' contains the possible settings including a small description. You have to enter the setting in the upper box line by line as shown in the picture below.

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Analog FXO options

General analog FXO settings

The picture below will show you analog FXO 'Port-Group' specific basic settings.

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Ports
Ports which have been added to the 'Port-Group'

Interdigit timeout
For every incoming call a interdigit collect timer will be started. After this specified timeout, without getting a digit, the call will be processed to the Dialplan. Note: This timer is only started if 'Overlap Dialing' is deactivated. d default: 3 sec.

Interdigit timeout initial
This Timer is the initial inter digit timer, that means before we got any digit. This Timer will be stopped after the first digit and the above mentioned Interdigit timeout timer will apply. During this time a Dialtone will be generated. Note this Timer is only started if 'Overlap Dialing' is deactivated. (default: 15 sec.

Overlap Dialing
This option will activate real Overlap Dialing, for instance in ISDN environments. By activating this option the 'Interdigit' as well as the 'Interdigit timeout initial' timer will be deactivated.

Tones
Tone sets which are categorised by country

CLIP
To allocate a number for this 'Port Group' that you can use as destination in the Dialplan.

CNIP
To allocate an alphanumeric number for this 'Port Group' that you can use as destination in the Dialplan.

Chan Sel
Channel Selection schemes: (Standard / Random / Round Robin) default: Standard Standard - selects the next free channel in ascending order Random - selects the next free channel at random Round Robin - selects the next free channel base on the Round-Robin principle

ChanSel direction
Ascending or descending direction of the channel selection

Connect
How beroNet Gateway should detect an FXO connect instant (after dialing the state will immediately change to 'connect') polarity (the opposite site send a polarity reversal to detect a 'connect') default: instant

Wait for OAD
wait (default: wait 2sec. to detect the OAD) dontwait (will immediately process without waiting for the OAD)

Dialtone passthrough
default: disabled

Analog call ending signal
the kind of signal must be detected to finish the call. unobtainable tone busy tone

CID Detection mode
The caller ID standard is determined by this setting. Bellcore ETSI ETSI-DTMF-AFTER_RINGING

Advanced Configurations

The picture below will show you analog FXO 'Port-Group' advanced basic settings.

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EC
This will activate or deactivate the onboard Hardware Echo canceler. default: on

EC tail length
EC tail length [0=8ms,1=16ms,2=24...,15=128ms] (default value is 15=128ms) 16 tabs each 8ms.

CLIR on CLIP
To dynamically hide the CallerID in direction of this 'Port-Group' (FXO). For instance if berofix detects a call to this 'Port-Group' with a CLIP value correlates with a CLIR_on_CLIP value (after the dialplan), the CallerID will be hidden, that means the remote end doesn't see it.

CLIP (Caller ID)
The CLIP (Caller ID) gives you the possibility to tell the Gateway, which field should be used for the CLIP (Caller ID)

CNIP (Caller Name)
The CNIP (Caller Name) gives you the possibility to tell the Gateway, which field should be used for the CNIP (Caller Name)

CLIP/CNIP could be applied to the following fields:

Additional configuration options

The mentioned below settings are mostly used and are directly outputted through the WebInterface.
But berofix has a lot of more settings, which are used in very special scenarios. These settings can be found at additional configuration options.

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Analog FXS configuration

General FXS configurations

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Group Name
Unique name of the 'Port-Group'

Ports
Ports which have been added to the 'Port-Group'

Interdigit timeout
For every incoming call a interdigit collect timer will be started. After this specified timeout, without getting a digit, the call will be processed to the Dialplan. Note this Timer is only started if 'Overlap Dialing' is deactivated. d default: 3 sec.

Interdigit timeout initial
This timer is the initial inter digit timer, that means before we got any digit. This Timer will be stopped after the first digit and the above mentioned interdigit timeout timer will apply. During this time a Dialtone will be generated. Note this Timer is only started if 'Overlap Dialing' is deactivated. (default: 15 sec.

Overlap Dialing
This option will activate real Overlap Dialing, for instance in analog environments. By activating this option the 'Interdigit' as well as the 'Interdigit timeout initial' timer will be deactivated.

Tones
Tone sets which are categorized by country

CLIP
To allocate a number for this 'Port Group' that you can use as destination in the Dialplan.

CNIP
To allocate an aphanumeric number for this 'Port Group' that you can use as destination in the Dialplan.

Chan Sel
Channel Selection schemes: (Standard / Random / Round Robin) default: Standard Standard - selects the next free channel in ascending order Random - selects the next free channel at random Round Robin - selects the next free channel base on the Round-Robin principle

ChanSel direction
Ascending or descending direction of the channel selection

Message waiting method
To define which message waiting indication method is used. Stutter Frequency-shift keying (FSK) off

Advanced FXS configurations

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EC
This will activate or deactivate the onboard Hardware Echo canceler. Default: on

EC tail length
EC tail length [0=8ms,1=16ms,2=24...,15=128ms] (default value is 15=128ms) 16 tabs each 8ms.

CLIR on CLIP
To dynamically hide the CallerID in direction of this 'Port-Group' (FXS). For instance if the beroNet Gateway detects a call to this 'Port-Group' with a CLIP value correlates with a CLIR_on_CLIP value (after the dialplan), the CallerID will be hidden, that means the remote end doesn't see it.

Dialplan Source
The PSTN Caller-ID which is used as source for matching in the Dialplan. CLIP or CNIP

Additional configuration options

The below mentioned settings are mostly used and are directly outputted through the webinterface.
But the beroNet Gateway has a lot of more settings, which are used in very special scenarios. These settings can be found at additional configuration options.

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LTE (and GSM)

LTE options

The LTE module behaves like all the other modules, which means its ports need to be grouped, so that they can be used in the Dialplan.

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Group Name
Unique name of the 'Port-Group'

ChanSel
Channel Selection schemes: (Standard / Random / Round Robin) default: Standard Standard - selects the next free channel in ascending order Random - selects the next free channel at random Round Robin - selects the next free channel base on the Round-Robin principle

ChanSel direction
Ascending or descending direction of the channel selection

SMS Extension
The destination number for the SMS

Extension
The destination number

LTE general

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Every LTE Port has some unique configuration which is done in this settings page. These configurations are for example the PIN of the SIM card or the SMSC (SMS center) for the SIM card.

NOTE: The PIN can be left blank if no PIN is stored for the Sim Card.
The SMSC needs to be configured and is different for each provider. Lists can be downloaded on the internet, here are some germans providers SMSCs:

O2 +491760000443 D1 +491710760000 D2 +491722270000

This information is supplied without liability, you should contact your mobile provider.

SIP

By selecting the menu point SIP, you will be directed to the page containing all options regarding SIP. There are 2 menu-points, SIP and SIP Stacks which will be explained in detail in the next chapters.

SIP

In this chapter we will explain SIP specific configuration and settings. Under the menu point SIP you will get an overview about all SIP accounts.
As already mentioned in the chapter 'Port-Groups', you can add, modify and delete SIP accounts by clicking on the corresponding button.

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SIP Configuration

A SIP account always consists of the following settings:

Name
Name you want to use for this SIP account. You can set it as you want, but it can help you with the assignment later.

SIP outbound proxy, SIP registrar and SIP domain
IP address of the SIP server (for instance your 3CX or Asterisk Server).
You can also set an alternative UDP port in the format sip.beronet.com:5061 if required (default is 5060)

If you want to connect this trunk to the SIP-Provider, you have to use the IP given by the Provider.

User
This username will be used as User Part in the SIP From Header. If you need to specify a different SIP From URL (sometimes called fromdomain) you can specify the SIP User in the form from_user@from_url by default the from_url is simply the server address.

If you're going to connect this trunk to the SIP-Provider, this is where the base-number should go.

Authentication user and secret
SIP Username and authentication password used for SIP authentication.

If you connect this trunk to the SIP-Provider, you have use the credentials given by the provider.

Match Type
The 'Match type' give you the possibility to choose which part of the from or To header the beroNet Gateway should use to match.

Note: The settings only match if you set the Match type in the relevant Dialplan to <Default SIP account>. It has the following options and meanings:

Call Transfer Method

SIP Stack

SIP transport
You can choose between the following supported SIP transport modes:
UDP (SIP Transport via UDP)
TCP (SIP Transport via TCP)
TLS (SIP Transport via TCP with TLS and Certificate)

SIP Port
Defines the local SIP Port for this account. The default is 5060 which is shared with all SIP Accounts. 
The values can range from 1025 - 65535.

NAT options

Validate/Keepalive
When enabled, the gateway will regularly send OPTIONS packets for keepalive and monitoring of the remote peer. 
It checks if the link between the two connections is operating, thus preventing the link from being broken.
The interval in which the messages are sent is given in seconds. (default is every 30s)

Register
When enabled, the gateway will send out SIP REGISTER messages to this SIP-accounts server address.
The gateway will automatically changes the following header settings depending on this option:

The interval in which the registration attempts are sent is given in seconds. (default is every 300s)

Advanced SIP settings - Media

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T.38 Support
Check for Fax Tones and tries a T.38 reinvite to make a reliable Fax-Over-IP Connection. The SIP device which is connected to the beroNet must support T.38. Most ATA's and some SIP SoftPBXs support T.38. Default is active.

DTMF Tones
DTMF mode over SIP

RFC 2833: tones transmitted via RTP packet inband: DTMF tones transmitted via inband info: DTMF via SIP-Info packets

SRTP
off: RTP/AVP session without crypto suites for SDP
optional: RTP/AVP session with crypto suites for SDP
mandatory: RTP/AVP session with crypto suites for SDP

Codecs
Defines which codecs are allowed and in which order they should be offered.

Advanced SIP settings - Call-Progress

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Wait for cancel
When the PSTN releases a call with a proper reason and with in-band information, this setting will be reviewed.

If enabled the beroNet Gateway will not immediately send the corresponding SIP response, instead it will make the in-band audio information available, so that the user can hear this message. The beroNet Gateway will wait until the user cancels this call while providing the in-band audio. If it is disabled the beroNet Gateway will immediately finish the call by sending the corresponding SIP response that is mapped for the PSTN Release reason. See also the beroNet Gateway ISDN Cause/SIP Response map for details of this mapping. Default: enable.

Call progress table S2I
Call progress table SIP to ISDN You can define which Call Progress Table should be used for this SIP account. By default (if left empty) the beroNet Gateway build-in Call-Progress Table would be used.

Call progress table I2S
Call progress table ISDN to SIP You can define which Call Progress Table should be used for this SIP account. By default (if left empty) the beroNet Gateway build-in Call-Progress Table would be used.

Advanced SIP settings - Headers

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