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Getting Started

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

username: admin
password: admin

please make sure to change these default settings, to avoid malicious attacks. You can change the admin password at Preferences->Security.

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

red  -> reboot  (hardware and network settings) 
orange -> restart of ISGW and call drops (PSTN Settings)
yellow -> applies immediately without call drops  (SIP, Dialplan and Cloud settings)

Only after clicking on the activate button, the changes apply. It is useful to make several changes and activate them once.

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". Finaly the different entities are connected in the "Dialplan". The dialplan defines, which calls should be forwared from SIP to PSTN and vice versa, based on criterias like the dialed number or the callerid. So the common steps are:

  1. Hardware Settings
  2. PSTN Groups
  3. SIP Accounts
  4. Dialplan Entries

Otionaly 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. ISGW comunicates to the hardware through drivers. Every module has a separte hardware driver which has different settings depending on the type of techonlogy. These settings can be configured in the Hardware section.

The physical connections are called ports. ISGW needs to initialize 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 finaly dialplan rules define how the calls should be routed by ISGW.

Hardware

The modules are connected on the beroNet device to a flexible TDM Backplane, which allows sharing of synchronization clocks and briding 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 syncronization settings and to change technology related settings. Every change in the hardware section requires a red activate which yields to a reboot.

Hardware-overview.png

The picture highlights the red activate button, the graphical presentation of the interfaces (in this case T-Adpators are required to access all the techonologies) and the types of modules that are installed.


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 scenarion one module should be choosen 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 seldom cases it might be necessary to have seperate synchronization sources on the same device with 2 modules. This might be the case when 2 different ISDN providers are used which have seperated timing sources.

The FXS, FXO and GSM Modules can not derive their timing by their interfaces, because these technologies don't provide timing sources. Thus these modules always have a local oscilator 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 defining which port should be used a primary synchronization source.

Synchronization-master.png


The picture shows a 4 port BRI module which is set to be TDM master and the first port should be used to derive timing for the whole module.

Synchronization-slave.png

The picture shows how a 4 port BRI module is set to be TDM slave and thus expects timing from another module or via the PCM cable. No synchronization port needs to be defined here.

ISDN Settings

Each ISDN Port can be configured separately. The port can be set to TE or NT mode, the protocol can be Point-to-Point or Point-to-Multipoint, the Line Termination can be enabled and a permanent Layer 1 timer can be started.

BRI-settings.png

Mode

The mode defines wether the device will be TE to be connected to an ISDN provider, or wether 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 an 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.

Analog Settings

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

Hardware analog FXS.png

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

Hardware ringvoltage.png


When using FXO Analog modules (i.e. BF4FXO) you can set each country with own settings regarding signals and tones

Hardware analog FXO.png

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.

Hardware pcm.png

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.

Hardware pcm slave.png

PSTN

The menu point PSTN+ gives you an overview about the ports which are provided by the plugged modules you are using on this particular berofix. For each technology you will find a sub menu point, like 'ISDN PRI' / 'ISDN BRI' / 'Analog FXO' / 'Analog FXS' / 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 next picture, you can add, modify and delete 'Port-Groups' by clicking on the corresponding button.

PSTN menu.PNG


You can add Ports to the 'Port-Group' by selecting available available ports. A port which is already assigned to a 'Port-Group' will be displayed grey.

PSTN portgroup.PNG

Each configured 'Port-Group' has technology specific settings. The settings on the 'Port-Groups' will apply to all ports, which are member of this 'Port-Group'. Furthermore if you dial out on a 'Port-Group' the system will automatically find the next free channel in this 'Port-Group', regardless if it is a BRI/PRI, Analog or GSM Port. You need to put all the ports you want to be able to configure into a 'Port-Group' (even if you only want or have one). As mentioned before, depending of the technologies you are using, different settings will be available.

ISDN PRI/BRI options

This Chapter will explain the ISDN configuration possibilities and ISDN specific settings.

General ISDN PRI/BRI settings

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

PSTN ISDN general.PNG

Group Name

Unique name of the 'Port-Group'

Ports

Ports which are member 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

Tone

ISDN Tone sets which are categorized by country

Interdigit timeout

For every incoming call a inter digit 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.

QSIG support

Enable or disable QSIG support

Link Down behavior

In some countries like Cyprus the behavior of ISDN PTP ports regarding Layer1 and Layer2 are different. They deactivate Layer 1 and Layer 2 after a while of inactivity. With this option you can solve this issue.
Nothing
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 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 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 settings

PSTN ISDN advanced2.PNG

EC

This will activate or deactivate the onboard Hardware Echocanceler
off,on 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.

dton

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

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.

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)

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.

Clear on OAD

useful for dynamically hide the CallerID in direction of this 'Port-Group' (ISDN). 
For instance if berofix 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

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: orginator address (default) 
OAD2 (OAD2 in case you have 2 OAD's you can choose with which value you want)
Qsigname (Qsigname to use it at source in the Dialplan)
Redirected_nr (redirected number)

Caller-ID Mapping

PSTN ISDN mapping2.PNG


The OAD (Caller ID) gives you the possibility to tell berofix, which Field should be used for the OAD for calls to this 'Port-Group'. oad (Caller ID) could be applied to the following fields

new_source (use new_source from the Dialplan as OAD for calls to this 'Port-Group') 
from_user (use SIP from_user as OAD for calls to this 'Port-Group')
from_display (use SIP from_display as OAD for calls to this 'Port-Group')
pai_all (use P-Asserted-Identities as OAD for calls to this 'Port-Group')
pai_user (use the P-Asserted-Identity user part: "berofix" <sip:gateway@beronet.com>)
pai_display (use the P-Asserted-Identity display part: "berofix" <sip:gateway@beronet.com>)
ppi_all (use P-Preferred-Identities as OAD for calls to this 'Port-Group')
ppi_user (use the P-Preferred-Identitiy user part: "berofix" <sip:gateway@beronet.com>)
ppi_display (use the P-Preferred-Identity display part: "berofix" <sip:gateway@beronet.com>)
none (use nothing for the OAD)
manual (use a constant string for as OAD for calls to this 'Port-Group')

A deeper explanation on how theses ISDN and SIP Attributes are connected can be found here: Caller-ID Mapping Howto

Additional configuration options

The lower 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.

PSTN ISDN additional.PNG

Analog FXO options

This Chapter will explain analog FXO configuration possibilities and analog FXO specific settings.

General analog FXO settings

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

PSTN FXO general.PNG


Ports

Ports which are member of the 'Port-Group'

Interdigit timeout

For every incoming call a inter digit 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 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

Connect

How berofix should detect a 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.

PSTN FXO advanced2.PNG

EC

This will activate or deactivate the onboard Hardware Echocanceler
off,on 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.

Dialplan Source

The PSTN Calller-ID which is used as source for matchin in the dialplan.
CLIP
CNIP

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.

PSTN ISDN additional.PNG

Analog FXS configuration

This Chapter will explain analog FXS configuration and analog FXS specific settings.

General FXS configurations

PSTN FXS general.PNG

Group Name

Unique name of the 'Port-Group'

Ports

Ports which are member of the 'Port-Group'

Interdigit timeout

For every incoming call a inter digit 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 hhannel selection

Message waiting method

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

Advanced FXS configurations

PSTN FXS advanced.PNG

EC

This will activate or deactivate the onboard Hardware Echocanceler
off,on 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 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/CNIP (Caller ID) gives you the possibility to tell berofix, which field should be used for the CLIP/CNIP for calls to this 'Port-Group'. CLIP/CNIP (Caller ID) could be applied to the following fields

new_source (use new_source from the Dialplan as CLIP/CNIP for calls to this 'Port-Group') 
from_user (use SIP from_user as OAD for calls to this 'Port-Group') 
from_display (use SIP from_display as CLIP/CNIP for calls to this 'Port-Group') 
pai_all (use P-Asserted-Identity as CLIP/CNIP for calls to this 'Port-Group') 
pai_user (use the P-Asserted-Identity user part: "berofix" <sip:gateway@beronet.com>) 
pai_display (use the P-Asserted-Identity display part: "berofix" <sip:gateway@beronet.com>) 
ppi_all (use P-Preferred-Identity as CLIP/CNIP for calls to this 'Port-Group') 
ppi_user (use the P-Preferred-Identitiy user part: "berofix" <sip:gateway@beronet.com>) 
ppi_display (use the P-Preferred-Identity display part: "berofix" <sip:gateway@beronet.com>) 
none (use nothing for the CLIP/CNIP) 
manual (use a constant string for as CLIP/CNIP for calls to this 'Port-Group')

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.

PSTN FXS additional.PNG


GSM

GSM options

The GSM Module behaves like all the other modules, which means it's ports need to be grouped, so that they can be used in the dialplan.

PSTN GSM.PNG

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

GSM additional options

PSTN GSM additional.PNG

GSM general

PSTN GSM general.PNG

Every GSM Port has some unique configuration which is done in this Setting 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.


SMS

PSTN GSM SMS.PNG

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 General 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 you can see in the next picture. As already mentioned in the chapter 'Port-Groups', you can add, modify and delete SIP accounts by clicking on the corresponding button.

SIP configuration summary.png

SIP Configuration

A SIP account consist always the following settings.

SIP general2.PNG

Name

Name you want to use for this SIP account (berofix internal)

Server address

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 requuired (default is 5060)

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.

Authentication user

SIP Username used for SIP authentication.

Secret

SIP authentication password

Match Type

The 'Match type' give you the possibility to choose which part of the from or To header the beroFix 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:

IP Address: By choosing this option, the IP-address from where the call is originated has to match to the 'Server address' of the SIP account, which has been chosen in the 'Server address' field.

From User: By choosing this option, additionaly to the IP-Adress, the SIP From_user part, of a call has to conform with the 'user' field of the SIP-account. 

To User: additionaly to the IP-address choosen in the server address field, the beroFix will match if the user part of the To Header conforms with the 'user' field of the SIP-account.

Contact User: matches if the contact header conforms with the user field of the SIP-account.

Manual: with this option, you can set a manual address (IP-address) which should be used instead of an existing SIP-account. Matching is the same like described in 'IP Addres' above. 
This field also could contain regular expressions. [192.168.2.15|192.2168.2.16]

Advanced SIP settings (more)

SIP configuration advanced1.png

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 beroFix must support T.38. Most ATA's and some SIP Softpbxes 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

IE on SIP

Information elements on SIP activates the possibility to send additional ISDN Information Elements over SIP-Headers.

If set to active beroFix will encode ISDN Information Elements like the 'Bearer Capability', 'Type of Number' or the 'Release Cause' etc. as X-BF SIP Headers. 
beroFix will also look for X-BF Headers on incoming SIP Messages to decode them into ISDN Information Elements. See Howto to use X-BF Headers for more details. Default: off.

Codecs

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

By default codecs is set to 'G.711 Alaw'.

Wait for cancel

When the PSTN releases a call with a proper reason and with inband information, this setting will be reviewed.

If enabled the beroFix will not immediately send the corresponding SIP response, instead it will make the inband audio information available, so that the user can hear this message. 
The berofix will wait until the user cancels this call while providing the inband-audio. If it disabled the beroFix will immediately finish the call by sending the corresponding SIP response that is mapped for the PSTN Release reason. 
See also the beroFix 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 (when left empty) the beroFix 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 (when left empty) the beroFix build-in Call-Progress Table would be used.


Failover account

This feature allows to provide a backup SIP Peer which is used in case the primary SIP Peer will not be reachable or will return a Server Error (503).

sip accounts

Failover timeout

The timne without interruption of reconnection process. After this time the fail over account will be activated.
0 (off) - 60 seconds


Dialplan Source

Determines which part will be used for the dialplan source.

from_user (use SIP from_user as dialplan source for calls to this 'Port-Group') 
from_display (use SIP from_display as dialplan source for calls to this 'Port-Group') 
pai_all (use P-Asserted-Identity as dialplan source for calls to this 'Port-Group') 
pai_user (use the P-Asserted-Identity user part: "berofix" <sip:gateway@beronet.com>) 
pai_display (use the P-Asserted-Identity display part: "berofix" <sip:gateway@beronet.com>) 
ppi_all (use P-Preferred-Identity as dialplan source for calls to this 'Port-Group') 
ppi_user (use the P-Preferred-Identitiy user part: "berofix" <sip:gateway@beronet.com>) 
ppi_display (use the P-Preferred-Identity display part: "berofix" <sip:gateway@beronet.com>) 
rpi_all (use Remote-Party-ID as dialplan source for calls to this 'Port-Group') 
rpi_user (use the Remote-Party-ID user part: "berofix" <sip:gateway@beronet.com>) 
rpi_display (use the Remote-Party-ID display part: "berofix" <sip:gateway@beronet.com>)

Called Party Number

To user part

The value which will be used in the To user part in the SIP-Invite sending to the SIP-PBX.

new destination: use the value for the TO: user part which results in the new destination part of the dial plan.
dad: use the DAD which comes in
account_username: use user from this SIP-account as SIP from_user
manual: enter a manual value to be used as SIP from_user

Caller ID Mapping

From User Part

The value which will be used in the From user part in the SIP-Invite sending to the SIP-PBX.

new_source: use this value for the FROM: part which results in the new source part of the dial plan.
oad: use the orginated number
oad2: use the OAD2 if an OAD2 exists
qsigname: use the qsigname
account_username: use the user which is indicate in the SIP configuration
none: none
manual: enter a manual value to be used in the SIP From user part

From Display Part

The value which will be used in the From display part in the SIP-Invite sending to the SIP-PBX.

new_source: use this value for the FROM: part which results in the new source part of the dial plan.
oad: use the orginated number
oad2: use the OAD2 if an OAD2 exists
qsigname: use the qsigname
account_username: use the user which is indicate in the SIP configuration
none: none
manual: enter a manual value to be used in the SIP Display part

Provider International Number Format National Code

National code without leading zero 

International Code

International code without leading zero

International prefix

International prefix (Default:00)

'ISDN to SIP' Destination number format

(00)(int)(nat)(num) 00 49 30 2593890
(+)(int)(nat)(num) + 49 30 2593890
(int)(nat)(num) 49 30 2593890
unknown (default)

OAD prefix setting

none (default)
remove prefix: removed prefix set in "International prefix" 
replace: "+" by "prefix": "+" will replaced by value set in "International prefix" 

'SIP To ISDN' DAD prefix setting

replace: "+" by "prefix": "+" will replaced by value set in "International prefix" 
remove prefix: removed prefix set in "International prefix" 
none (default)

OAD prefix setting

replace: "+" by "prefix": "+" will replaced by value set in "International prefix"
prepend prefix: removed prefix set in "International prefix" 
none (default)

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

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.

SIP additional.PNG

SIP General

Under SIP General settings you can set the following values

SIP general3.PNG

Bind port

The Port on which the berofix device should listen for SIP traffic. Default is 5060

RTP port range

RTP port range which should be used for RTP Traffic. Default is (5000-5059,5062-6000)

TOS RTP

Type of service for RTP traffic. Useful for prioritization of the RTP traffic. Default: 160

TOS SIP

Type of service for SIP traffic. Useful for prioritization of the SIP traffic. Default: 160

Reject calls under load

RTP port selection

standard 
roundrobin

SIP transport

The beroFix devices support the following SIP transport modes:
udp (SIP Transport via UDP)
tcp (SIP Transport via TCP)
tls (SIP Transport via TCP with TLS and Certificate)

Dialplan

The dial plan is one of the most important things to set up during the configuration of a beroFix device. The dial plan defines rules concerning how calls should be routed under certain circumstances. The beroFix dial plan engine is based on regular expressions ('Howto RegEx') and reads the dial plan entries from the top to the bottom. After the first match beroFix will leave the dial plan and execute the rule. Therefore it is important to know that special dial plan rules should be placed on top while the general ones should be placed below them. This can be done by using the position arrows.

DialplanV3.png


As you can see you can add, modify, copy and delete a dial plan rule by pressing on it's corresponding button on the right in the table. Furthermore you access the the advanced options for a particular dial plan rule by pressing the 'tool icon' also on the right-hand side in the table. Above the table you can set some filters for the dial plan table like 'direction' or 'entries per page' or even search for a special character inside all fields of a dial plan rule. As you can see that there are several columns. Before explaining the meaning of each column, it is important to know that some of these columns are matching criteria and the others are executing ones. Matching columns are taken into account to decide if a particular dial plan rule matches while the executing columns would be executed when a dial plan rule has matched. Only if all matching columns are true a particular dial plan rule will it be executed.


Matching Columns:

Direction: The direction of the call. Here you can set the direction a call has to be from for this rule to qualify and to where the call is to be routed. 
For instance in the first row you see that the “call” has to originate from SIP and will then be routed to Analog. In the the second row it is vice versa.
FromID: The ID from where the call is originated. Depending of the direction that is chosen the FromID could be the name of a 'SIP-account', a 'PSTN-port-group' or a manual value like an IP-address.
Destination: Also called "CalledID" or DAD. This is the number which the caller dialed.
Source: Also called "CallerID" or OAD. This is the number of the device of the caller.

Execution Columns:

ToID: The ID where the call should be routed to. Depending on the direction this could be a particular 'PSTN-port' a 'PSTN-port-group', the name of a SIP-account or an IP-address
New Destination: The new destination after the dial plan rule has been executed (see 'Destination' above).

New Source: The new source after the dial plan rule has been executed (see 'Source' above).


New dialplan rule

You can add,modify or copy a dial plan rule by pressing it's corresponding buttons. The next picture will show you the input form. On the left-hand side you will find all of the matching fields and on the right all of the executing fields. Each matching field has one executing field, that means there is always a pair of both.

Dialplan newrule.PNG


From direction/To direction

The first pair is the direction pair, consisting of the fields From direction and To direction. From direction is the matching field and it checks from where the call originated while the "To direction" field (executing field) makes necessary changes so that the call can be routed to where you would like it to go. In the picture above we have chosen the direction SIP-to-ISDN. Depending on which modules are plugged in, the fields 'From direction' and 'To direction' could have the following values::

SIP
Analog (FXS/FXO)
ISDN (PRI/BRI)
GSM
In other words if haven't plugged in a GSM module, you will not see the GSM option in 'From direction' and 'To direction' pair.
Note: You can combine every technology direction with each other except the direction SIP-to-SIP.

From Direction is set to ISDN / Analog / GSM):

In this case the From ID is a list of all configured PSTN port-groups indicated by their names. For instance if you select ISDN you will only see the ISDN port-groups in the list. Alternatively to the port-group you could also select a single port, identifiable by it's port number. But this port has to be a member of a port-group anyway, otherwise you won't be able to select it. All calls which originated from this single port or the port-group would match the criteria 'From_ID'.

From Direction is set to SIP:

If 'From Direction' is set to SIP, the 'From ID' is a list of all SIP-accounts, configured on the beroFix, indicated by their 'Names'. In this case the a new field called 'Match type' would be added. The 'Match type' give you the possibility to choose how beroFix should match for the From_ID. The match type has the following options and meanings:

From_IP: By choosing this option, the IP-address from where the call is originated has to match to the 'Server address' of the SIP account, which has been chosen in the From_ID Field.
From User: By choosing this option, additionaly to the IP-Address (see From_IP), the SIP From_user part, of the a call has to match to the 'user' field of the SIP-account, which has been choosen in the From_ID field.
Manual Address: By choosing this option, you can set a manual address (IP-address) which should be used instead of a existing SIP-account. Matching is the same like described in 'From_IP' above. This field also could contain regular expressions like our example below:
[172.20.0.2|172.20.0.3]

To Direction is set to ISDN / Analog / GSM):

The To ID field is a bit easier, because it is the execution part. If 'To direction' set to PSTN (ISDN / Analog / GSM) a list of the corresponding configured PSTN ports groups or PSTN ports is shown and you can select the desired direction. If 'To direction' is set to SIP you can select one of the configured SIP accounts or enter a manual address as described above.


Destination / New Destination

This pair 'Destination / New Destination' is one of the most important matching criterias. The destination is the calledID (the number which was originally dialed) also named DAD, 
this means the number which enters the dial plan.  While the new destination is the calledID, when we leave the dial plan. Both destination and new destination are based on regular expressions.

Source / New Source

This pair 'Source / New Source' very similar to the pair 'Destination / New Destination' but apply to the callerID (OAD)

Comments

For each dial plan rule you can leave a small comment. You can see the comment as a tooltip Box in the 'Dialplan overview', when you hover over a dial plan rule with the mouse.

Active

Each dial plan rule can be active or inactive. You can see a deactivated rule in the dial plan overview, it will be indicated as inactive by a grey background.

Advanced dial plan options

When you have created a dial plan rule you can customize it to be more advanced by clicking on the tool icon on the right-hand side of the dial plan overview page (the fourth icon on the right). Depending on what is set for 'From' and 'To Direction' you will have corresponding buttons to reach the more advanced options for ISDN / Analog / GSM and SIP.

Dialplan advanced configuration1.png

The picture above shows you the advanced option's in this case for ISDN. These settings are exact the same like described in the Chapter 'Advanced ISDN PRI / BRI options' , which are attached to the 'ISDN Port Group'. In the dial plan you have the possibility to make special settings for one particular rule. This means on the 'ISDN-Port-Group' you have the default settings while in the dial plan we can overwrite one or more values for a specific dial plan rule. In the right column of the above picture you can see the default button. By default this is activated. For instance if we check the option EC (Echo canceling) you will see that the value EC itself and the default column are activated. This means that the default on the ISDN-port-group for the option EC is activated and the dial plan inherits this value. To change this particular value you have to deactivate the 'default' option and then deactivate the value EC, as you can see in the next picture.

Dialplan advanced configuration2.png

This should illustrate how you can make very specific dial plan settings to particular rules. Note: The dial plan always has the highest priority. For instance in the above mentioned example EC is deactivated in the dial plan but activated in the 'ISDN-port-group'. If the dial plan rule matches, the value EC would be 'off' although this value is set to be 'on' by the ISDN-port-group. All other options (Analog / GSM / SIP) follow the same mechanism as described here. For more information please read their respective Part.

Examples of dial plan rules

The best way to explain the beroFix device handles dial plan rules, is to show you some examples. And before we start showing you examples, this might help to understand them (in case you didn't read the 'Howto RegEx'):

() will make everything contained by parentheses referable.
  You can refer to it by the use of the \<DIGIT> i.e. \1 for the first parentheses \2 for the next etc.
. is a special symbol that matches any single symbol
* is a multiplier that changes the statement after which it stands 
  (i.e. .* would mean <any_symbol>AND<any_number_of> which would mean literally any string of any length)
therefore (.*) would cause anything to match and make it referable by it's respective \<DIGIT>

Example1: Incoming call from SIP with the following settings

SourceIP:    172.20.0.1
CallerID:    2593890
CalledID:    025938912

Dialplan entry values:

Direction:          “SIP->ISDN”
FromID:             “.*”      matches anything
ToID:               “(g:te)”  routed to ISDN-port-group g:te
Destination:        “0(.*)”   matches any calledID starting with '0'
New Destination:    “\1 “    \1 will reference to parameter 1 ( \1 the value in the first parenthesize of destination )
Source:             “(.*)”    matches any callerID 
New Source:         “\1 “    \1 is the value in the first parenthesize of source

With these settings the the “Call” will be routed to ISDN-port-group “g:te”. The calledID will be changed to 25938912, that means the first '0' will be stripped while the callerID will routed transparent and still is 2593890.

Example2: Incoming call from SIP with the following settings

SourceIP:    172.20.0.1
CallerID:    12
CalledID:    0176242XXXXX.

Dialplan entry values:

Direction:         “SIP->ISDN”
FromID:            “172.20.0.1”  matches if source IP-address is 172.20.0.1
ToID:              “1”           routed to ISDN-port 1
Destination:       “0176(.*)”    matches if calledID is starting with 0176
New Destination:   “0049176\1”   will cut 0176 from calledID and add 0049176 to the calledID, followed by reference to parameter 1 
Source:            “(..)”        matches callerIDs with exact 2 digits.
New Source:        “25938912”    CallerID will be overwritten by 25938912

With these settings the the “Call” would be routed to ISDN-port 1”. The calledID will be modified to 0049176XXXXX. The callerID will be changed to 25938912.

Example3: Incoming call from ISDN with the following settings

ISDNPort:    g:teports
CallerID:    0176242XXXX
CalledID:    25938912

Dialplan entry values:

Direction:         “ISDN->SIP”
FromID:            ”g:teports”           matches if the call is originated from the ISDN-port-group named 'teports'
ToID:              “p:mysipserver”       routed to the SIP account named 'mysipserver'
Destination:       ”259389([0-8][0-9])”  matches all numbers starting with 259389 followed by 2 digits in the range from [00-89]
New Destination:   “\1”                  will cut 259389 from calledID and add the 2 digits referenced by parameter 1
Source             “(.*)”                matches any callerID
New Source:        “\1 “                 \1 is the value in the first parenthesize of 'Source'

With these settings the the “Call” would be routed to SIP account p:mysipserver. The calledID will be changed to 12. The callerID will routed transparent and still is 0176242XXXX.

Example4: Incoming call from ISDN with the following settings

ISDNPort:    1
CallerID:    12
CalledID:    02593890 

Dialplan entry values:

Direction:         “ISDN->SIP”
FromID:            ”1”                matches the call is originated from the ISDN-port 1
ToID:              “p:mysipserver”    routed to the SIP account named 'mysipserver'
Destination:       ”0([2-9])(.*))”    matches all numbers starting with 0. The second digit has to be in range[2-9] followed by the any digits.
New Destination:   “\1\2”             will cut 0 from calledID and add the parameter 1 followed by parameter 2 
Source             “(.*)”             will match any callerID
New Source:        “\1 “              \1 is the value in the first parenthesize of 'Source' field

With these settings the the “Call” would be routed to SIP account p:mysipserver. The calledID will be changed to 2593890. The callerID is untouched.


With the above examples you should be able to handle almost every situation in the real world. If this is not enough and you need some special things, feel free to implement more complex regular expressions. More informations about this can be found in the corresponding chapters.

Preferences

This chapter will explain the possible settings you can make in the preferences part of the web-interface.


Network settings

By selecting the menu point “Network” you can configure all network options like IP-address, netmask as well as the default gateway.

Preferences network2.PNG

DHCP / Static IP

Choose to manually configure the beroFix network settings or use DHCP to get them from a DHCP server.

IP-Address

If you using static IP you can enter the IP-address the beroFix device should use.

Netmask

If you using static IP you can enter the subnetmask the beroFix device should use.

Gateway

If you using static IP you can enter the default gateway the beroFix device should use.

Primary DNS-Server

IP-Address of the first DNS-server

Secondary DNS-Server

IP-Address of the second DNS-server

Hostname

Hostname of the beroNet gateway

Advanced

MTU size

Maximum transmission unit. A value between 0-1500. Default is 1500.

Zeroconf Support

Disable bfdetect

VLAN Enable

If you want you add the beroFix device to a VLAN then enable this option.

Route target

Route gateway


Time settings

This is what the time settings look like:

Preferences time.PNG

NTP Host

If you want to get the time for your beroFix device from a server using hosting ntp, enter it's IP address here.

Timezone

Please select the time zone you are using the beroFix device in.

Summer / Winter change

Enable this option if you want the beroFix device to automatically adjust the systemtime to summer / winter changes.

Time from ISDN

If you want the beroFix device to get it's systemtime from ISDN, you can do so by enabling this option. Y
You have to choose a specific port. All incoming ISDN calls are checked for the time and if it differs from the current system time, the system time is set to the new time.


Provisioning

Preferences provisioning.PNG

For using the provisioning feature, you can choose either TFTP or HTTP provisioning.

TFTP / HTTP Host

Enter the IP-address of the respective TFTP / HTTP host. (The host that provides the configuration. )

TFTP / HTTP URL

Enter the URL of the configuration file in question of the respective TFTP / HTTP server.

Use boot TFTP / HTTP

If this option is activated, beroFix will try to start provisioning process during the boot phase.

beroFos Heartbeat

Please read beroFos manual for more information!

Preferences berofos heartbeat.PNG

IP-Address

If you want to use the beroFos heartbeat feature, you need to enter the IP-address of the beroFos device here.

MAC address

If you want to use the beroFos heartbeat feature, you need to enter the MAC-address of the beroFos device here.

Interval (sec.)

Here you can set the interval in which the beroFix device sends a heartbeat signal to the beroFos device.

Heartbeat on boot

Enable this option if you want the beroFix device to send a heartbeat during and after booting automatically. (Alternatively you can also manually start the heartbeat)

beroFos heartbeat state

Activates / deactivates the heartbeat process


Logging

Preferences logging.PNG

Logging server

The IP-address where the beroFix device should send it's logging information to.

Logging server port

The UDP port of the logging server

Logging Active

enable / disable the logging feature.

Log level

Here you can set the logging level. Log level 1 is very low while log level 9 is very high.

System log

Syslog facility

An information field associated with a syslog message. It is defined by the syslog protocol. It is meant to provide a very through clue from what part of a system the message originated from. 
LOCAL_0 to LOCAL_7 facilities are traditionally reserved for administrator and application use. The facility can be very helpful to define rules that split messages for example to different log files based on the facility level. 
Syslog_facility=[16-23] correlate to local0-local7.

CIFS enabled

CFIS share

CFIS user

CFIS password

Domain

Max. Log-Size (in MB)


Security

Preferences security.PNG

Old password

If you want to change your password, you must first enter the current one here as authentication.

New password

If you want to change your password, you can enter the new one here.

Confirm new password

If you want to change your password, you need to enter it again for verification.

Disable API

Disable API Sessions

Enable SSH-Access

Safe guard


ACL

Preferences ACL.PNG


Access control list:

SIP
Enter an IP-address or an IP-address range, to allow access to the beroFix SIP listening port.
SIP-TLS
Enter an IP-address or an IP-address range, to allow access to the beroFix SIPS listening port.
HTTP 
Enter an IP-address or an IP-address range, to allow access to the HTTP interface.
HTTPS
Enter an IP-address or an IP-address range, to allow access to the HTTPS interface.
Telnet
Enter an IP-address or an IP-address range, to allow access to the telnet interface.
SSH
Enter an IP-address or an IP-address range, to allow access via SSH.
SNMP
Enter an IP-address or an IP-address range, to allow access via SNMP.
SNMP-Trap
Enter an IP-address or an IP-address range, to allow access via SNMP-Trap.
bfdetect 
Enter an IP-address or an IP-address range, to allow responses to the bfdetect requests
All
Manual

Note: By default the Network connected to eth0 will be added as an allowed Network i.e. all access via the lan connected to eth0 will be allowed.

Preferences ACL1.PNG

Some Examples:

ACL 127.0.0.1 / 32 will limit the access to the IP-address 127.0.0.1 (localhost)
ACL 192.168.1.0 / 24 will limit the access to the network 192.168.1.1-192.168.1.254
ACL 172.20.0.0 / 16 will limit the access to the network 172.20.0.1-172.20.254.254


Causes Map

With the causes map you can modify the sended causes.

Preferences causes map1.PNG

The example in the picture converts the SIP status code [486 Busy Here] to the ISDN causes code [17 User busy].


Call direction

to choose the direction of a call has to be from this cause to be relevant.

To cause (ISDN)

to choose a cause to which the original cause should be translated to.

From cause (SIP)

 to choose which cause is to be converted to the different cause set above.

Please find more detailed info here: SIP response map.


Call Progress

Preferences call progress.PNG


Preferences call progress1.PNG


Miscellaneous

Preferences misc.PNG

Management

In this menu point you can reach the settings, tools and information to manage the gateway.

State

This will give you an overview about the PSTN, SIP and PCM interconnection state. It shows the state of the PSTN-Ports.


ISDN status

All ISDN Ports , which are in a port group, with port number, the type of connection [PTP] or [PMP] and the state for layer 1 and layer 2. If the LED are green the port is up. You can get more information regarding the state of these ports by hovering over the L1 link and L2 link field.

Management state1.PNG

Analog status

It shows a list of all analog ports which are in an analog-port-group. On FXO and FXS ports you can also see the line voltage by hovering over the Line voltage LED.

Management state2.PNG

SIP Registration Status


Management state3.PNG

Using SIP-to-SIP and SBC Functionality

By adding a SIP to SIP license on the gateway, we can use it to connect two SIP peers.

How to order and install a SIP2SIP licence on the gateway

Each SIP2SIP licence is linked to a single gateway. We therefore need the serial number of the gateway in order to create a licence for it. To find out the serial number of your gateway, go to "info" under "management+".

Include the serial number of your gateway in the license order you place at your local distributor.

You will then receive a file with the name "isgw_SerialNumberOfTheGateway.licence". 

To install the license in your gateway, go to "Miscellaneous" under "Preferences" and upload the licence under "SIP to SIP License". 

How to use your SIP2SIP licence

Once your license has been uploaded on the gateway, you can connect two SIP peers in the dialplan. 

  1. Create two SIP peers or accounts under "SIP"
  2. Create dialplan rules to link both SIP peers:

Available licences and upgrades

Different SIP2SIP licenses are available:

Article nameDescriptionGateway needed
BN2SIP2No license needed for two SIP-to-SIP channels (2 simultaneous calls without transcoding)BF400box or more
BN2SIP4License for four SIP-to-SIP channels (4 simultaneous calls without transcoding)BF400box or more
BN2SIP8License for eight SIP-to-SIP channels (8 simultaneous calls without transcoding)BF400box or more
BN2SIP16License for 16 SIP-to-SIP channels (16 simultaneous calls without transcoding)BF1600box or more
BN2SIP32License for 32 SIP-to-SIP channels (32 simultaneous calls without transcoding)BF1600box or more
BN2SIP64License for 64 SIP-to-SIP channels (64simultaneous calls without transcoding)BF6400box

Upgrades are available from license 4 to 8 and from 16 to 32.


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