berofix has several Call related configuration options. Some of these are SIP specific others are PSTN specific. These options have default settings which can be overwritten by the SIP or PSTN group configurations. The final overwrite rule comes from the Dialplan. So the Priority order where 1 has the lowest priority and 3 the highest is:

1. Default Value
2. SIP/PSTN Group Value
3. Dialplan Setting

SIP Settings

Config String Name GUI Label Values
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ea Early Audio yes/no

Turns audio on in the pre-connected state. Most users want to enable this, to hear
alerting sounds and other inband audio messages.
Default is yes.

t38 T.38 Support yes/no

Check for Fax tones and try a T.38 reinvite to make a reliable Fax-Over-IP connection. 
The SIP device which is connected to the beroNet Gateway must support T.38. Most ATAs and some 
SIP softPBXes support T.38.
Default is yes.
       
force_t38_reinvite Force T.38 reinvite x in ms, 0=off

Some SIP devices take very long to reinvite the beroNet Gateway into a T38 session and some Faxtones
are hard to detect. Here you can define a time in milliseconds after which the beroNet Gateway simply 
forces a T.38 re-invite regardless if it has received a Faxtone or not. This setting can
be used in the Dialplan if it's clear that the call is going to a Fax extension.
Default is 0.

dtmfmode DTMF Mode rfc2833, info, inband

Defines what to do with DTMF Tones that where detected on the PSTN side. If set to inband
the DTMF tones are left unchanged. If set to rfc2833 the tones are sent via Special RTP
Packets, if set to info the Tones are sent via SIP Info Messages. 
Default is rfc2833.

dtmfremoval DTMF removal both, tdm, packet, none

Defines whether DTMF tones should be removed from the PSTN side (tdm), the IP side (packet), 
from both sides (both) or not at all. 
Default is none.

clir_on_sip CLIR on SIP the Matchname for CLIR

Here you can define a SIP CallerID which should be used to enable CLIR for this call. So if 
you define clir_on_sip="anonymous" and send calls with a SIP callerid="anonymous" 
(from_user/displayname), then beroNet Gateway will enable CLIR for this call (CallerID will be hidden).
Default is empty.
 
ie_on_sip IE on SIP yes/no

If set to 1, the beroNet Gateway will encode ISDN information elements like the Bearer Capability or
the Release Cause as X-BF SIP Headers. The beroNet Gateway will also look for X-BF Headers in incoming
SIP messages to encode them into ISDN information elements. See Howto to use X-BF Headers 
for more details.
Default=0.

codecs Codecs pcma, pcmu, gsm, g729, g723, g726-32

This setting defines which codecs are offered and accepted by the beroNet Gateway. The configured order
is also the offered order.
Default is empty and means pcma.

from_id_setting From id setting 0,1,2

Defines what should be coded into the SIP from_user Part of the FROM Header.
0 means, that the gateway encodes the ISDN oad into the from_user, if this 
sip peer is configured as a Proxy. If on the other hand the peer is configured
as a registrar, then use the account-name, so that the registrar can authenticate
us. NOTE: some SIP servers including Asterisk use the from_user part of the FROM 
header as the CallerID-Number. So when the gateway registers at such SIP servers, it
must sent it's callerID via the displayname part of the FROM Header.
1 means that always the accountname is encoded in the from_user. 
2 means that always the oad will be encoded in the from_user.
Default is 0.

display_name_setting Display name setting 0,1,2

This setting defines what the beroNet Gateway will encode into the SIP displayname part of the FROM Header. 
0 means, that the displayname will be the oad. But if there is a second oad, the displayname 
 will be the second oad. But if there is a qsigname, the displayname will be the qsigname. 
1 means, that the displayname will always be empty. 
2 means, that the displayname will always be the first oad. 
Default is 0.

allow_sip_183_without_sdp Allow SIP 183 without sdp yes/no

This setting defines whether the gateway should sent out a 183 Messages without SDP, if a Proceeding
or a Progress ISDN message is received. In general it is a good idea to tell the other SIP side
that we received a Proceeding or Progress. But some Asterisk versions (<1.4) don't handle this 
SIP event properly.
Default is 1

wait_for_cancel Wait for Cancel yes/no

This setting is important in the direction SIP->PSTN. When the PSTN network releases the call
with a proper reason and with inband information, this setting will be reviewed. 
If set to 1, the gateway will not send immediately a SIP response back to the originator of 
the call, instead it will playback the inband audio information from the PSTN network via 
RTP, so that the user can hear it. The user will then after a while hangup the call by himself
so that it is fully released. 
If set to 0 the gateway will immediately finish the call by sending back a proper SIP response 
that is mapped for the PSTN release reason. See beroFix ISDN Cause/SIP Response map for 
details of this mapping.
Default is 1.

ISDN Settings

Config String Name GUI Label Values
-------------------------------------------------------------------------------------
ec Echocancel yes/no
Set to yes if you want to enable the Echocanceler and to no if you want to disable it.

ectl EC tail length 0=8ms,1=16ms,...,15=128ms
Specifies the Echocancel tail length in 8ms steps. This means how many transmit samples the 
Echocanceler will save and compare against it's receive samples. The higher this value is, the
longer it takes for the Echocanceler to adapt to the echo. But if it is set to small, 
it may not cope with the echo at all. 
In digital networks like ISDN in Germany a value between 32ms and 64ms should be quite enough.
On long distance calls 128ms can be a better choice.

dnumplan Type Of Number (Called Party) 0,1,2,4
Destination Type of Number. Values are:

0=unknown                Number is in unknown format, mostly in the "native" dialed format
                         with a 0 prefix for national and a 00 prefix for international 
                         numbers.
1=International          Number is in international format. This means that the number
                         has no 0 as prefix, but the international and the national prefix. Let's
                         say it is a number from Berlin/Germany, then the prefix for Germany is 49
                         and for Berlin is 030. So the number must start with 4930XXX.
2=National               Number is in national format. The number has no 0 as prefix, but 
                         the local prefix of the city. So for Berlin (030) the number starts with 30XX.
4=Subscriber             ????
Default: 0 and should be mostly OK.

onumplan Type Of Number (Calling Party) 0,1,2,4
Origination (CallerID) Type of Number. The Values are exactly the same as for the Destination 
Type of number (dnumplan). When connected to some traditional PBXs, this must likely be changed to national
or international and the CallerID must be provided in such format (without 0, but with appropriate 
prefixes). In the case of CLIP/noScreening this must be changed to either subscriber, national 
or international, depending on the settings of the local switch.
Default is 0. 

rnumplan Type Of Number (Redirected Party) 0,1,2,4
Like onumplan, to indicate what type of number the redirected number has.
Default is 0.

cpnnumplan Type Of Number (Connected Party) 0,1,2,4
Like onumplan, to indicat what type of number the Connected Party Number has.
Default is 0.

unknownprefix Unknown Prefix x - prefix
When an incoming call has an unknown Calling Party Number, the configured prefix
will be used.
Default: none

internationalprefix International Prefix x - prefix
When an incoming call has an international Calling Party Number, the configured prefix
will be used.
Default: 00

nationalprefix National Prefix x - prefix
When an incoming call has a national Calling Party Number, the configured prefix
will be used.
Default: 0

localprefix Local Prefix x - prefix
When an incoming call has a local Calling Party Number, the configured prefix
will be used.
Default: none

privateprefix Private Prefix x - prefix
When an incoming call has a Private Calling Party Number, the configured prefix
will be used.
Default: none

screen Screening Indicator 0,1,2
0                        Calling Party Number is User-provided, not screened
1                        Calling Party Number is User-provided, verified and passed
2                        Calling Party Number is User-provided, verified and failed
Default: 0

pres Presentation Indicator 0,1,2
0                        Calling Party Number Presentation allowed
1                        Calling Party Number Presentation restricted
2                        Calling Party Number not available, due to interworking
Default: 0

bearer_cap Bearer Capability SPEECH,AUDIO_3_1_K,...,DIGITAL_UNRESTRICTED
Defines which Bearer Capability (Type of Data) will be transmitted in the B-Channel. For normal speech calls, 
set this to SPEECH, for faxes and modems set this to AUDIO_3_1K. For Digital Data Calls set this to 
DIGITAL_UNRESTRICTED.
Default: SPEECH.

cd Calldeflect yes/no 
When set to yes, the beroNet Gateway will try to deflect calls on reception of a "302 Moved Temporary" to the given 
Destination. On PMP lines the gateway will send a Calldeflect on PP lines it will use Partial Rerouting. 
Default: no.

eao Early Audio Outbound yes/no 
Play Early Tones for incoming Call Requests on a TE Line. Normally the Telco only allows sending of audio
in the connected state. But in some special cases it is possible to send audio already after sending an 
Progress or Alerting.
Default: no.

gen_ring_eao Generate Ringing on EAO yes/no
If Early Audio Outbound is set, and we receive a 180 Ringing, then we generate the ringing tone by ourself. 
Default: no.

oad_setting OAD Setting fromuser,displayname
Defines if the fromuser part or the displayname part of the FROM Header should be transmitted as 
CallerID (oad).
Default: fromuser.

ignorep8 Ignore Progress Indicator 8 yes/no
If set to yes, the beroNet Gateway will only enable the audio when a Progress Indicator 8 was receipt before. This 
is necessary for some nasty PBXs that do only start sending data on the B-Channel after they've send a 
Progress Indicator (8), otherwise a disturbing noise is heard. On the normal telephone line the B-Channel is 
very early connected, so this should be no in most cases to ensure the fastest B-Channel connection.
Default: no.

allow_all_chars_in_isdn_number yes/no
If set to yes, the beroNet Gateway will send any ASCII character as ISDN numbers. Otherwise it will only send 
numeric numbers and discard others.
Default: no.

featurecodes DTMF Feature Codes codes
It is possible, that the beroNet Gateway takes some actions during a call when the user has entered a predefined
DTMF Tone Sequence. These actions include sending of specific ISDN Supplementary Services, that are 
not yet map-able to SIP methods. Currently only the MCID (Malicious Caller Identification) feature 
is supported.
The "featurecodes" string has 3 parameters, the general structure of it is 
<direction>:<dtmfsequence>:<feature>
<direction>    can be "t" for to, "f" for from and "b" for both. This defines which call-leg can enable 
               the features.
<dtmfsequence> The Sequence of the DTMF digits, that need to be entered to enable the feature.
<feature>      The name of the feature that should be enabled.
An example looks like: "t:*700:mcid;".  When a call was made from ISDN->SIP, then the SIP Entity (which
is the to - direction) will be able to send "*700" via DTMF during the call and the gateway will send out the 
ISDN Facility MCID.