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 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:
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Config String Name GUI Label Values ------------------------------------------------------------------------------------- ea Early Audio yes/no Turns Audioaudio on in the pre-Connectedconnected Statestate. Most users want to enable this, to hear Alertingalerting Soundssounds and other inband Audioaudio Messagesmessages. Default is yes. t38 T.38 Support yes/no Check for Fax Tonestones and try a T.38 reinvite to make a reliable Fax-Over-IP Connectionconnection. The SIP Devicedevice which is connected to beroFixthe beroNet Gateway must support T.38. Most ATAs and some SIP SoftpbxessoftPBXes support T.38. Default is yes. force_t38_reinvite Force T.38 reinvite x in ms, 0=off Some SIP Devicesdevices take very long to reinvite the berofixberoNet Gateway into a T38 session and some Faxtones are hard to detect. Here you can define a time in milliseconds after which berofixthe beroNet Gateway simply forces a T.38 re-invite regardless if it has received a FaxtonFaxtone or not. This setting can be used in the dialplanDialplan 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 Sideside. If set to inband the DTMF Tonestones are left unchanged. If set to rfc2833 the Tonestones 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 Tonestones should be removed from the PSTN side (tdm), the IP Sideside (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 CalleridCallerID 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 berofixGateway will enable CLIR for this call (CalleridCallerID will be hidden). Default is empty. ie_on_sip IE on SIP yes/no If set to 1, the beroNet beroFixGateway will encode ISDN Informationinformation Elementselements like the Bearer Capability or the Release Cause as X-BF SIP Headers. beroFixThe beroNet Gateway will also look for X-BF Headers in incoming SIP Messagesmessages to encode them into ISDN Informationinformation Elementselements. See Howto to use X-BF Headers for more details. Default=0. codecs Codecs pcma, pcmu, gsm, g729, g723, g726-32 This Settingsetting defines which codecs are offered and accepteraccepted by the berofixberoNet 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 berofixgateway encodes the ISDN oad into the from_user, if this sip peer is configured as a Proxy. If on the other hand the Peerpeer is configured as a Registrarregistrar, then use the account-name, so that the registrar can authenticate us. NOTE: some SIP Serversservers including Asterisk use the from_user Partpart of the FROM header as the CalleridCallerID-Number. So when the beroFixgateway registers at such SIP Serversservers, it must sent it's calleridcallerID 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 berofixberoNet Gateway will encode into the SIP displayname Partpart 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 berofixthe gateway should sent out a 183 Messages without SDP, if a Proceeding or a Progress ISDN Messagemessage is received. In general it is a good idea to tell the other SIP Sideside that we received a Proceeding or Progress. But some Asterisk Versionsversions (<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 Networknetwork Releasesreleases the call with a proper Reasonreason and with inband Informationinformation, this setting will be reviewed. If set to 1, berofixthe 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 Networknetwork via RTP, so that the user can hear it. The user will then after a while hangup the call by himself Soso that it is fully released. If set to 0 berofixthe gateway will immediately finish the call by sending back a proper SIP Responseresponse that is mapped for the PSTN Releaserelease Reasonreason. See beroFix ISDN Cause/SIP Response map for details of this mapping. Default is 1.
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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 Tailtail length in 8ms steps. This means how many Transmittransmit samples the Echocanceler will save and compare against it's receive samples. The higher this value is, the longer it takes for the echocancelerEchocanceler to adapt to the echo. But if it is choosenset to small, it may not cope with the echo at all. In digital Networksnetworks 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 Formatformat, mostly in the "native" dialed Formatformat with a 0 prefix for national and a 00 prefix for international numbers. 1=International Number is in international Formatformat. This means that the number has no 0 as prefix, but the international and the national prefix. Let's say it is a Numbernumber from Berlin/Germany, then the prefix for Germany is 49 and for Berlin is 030. So the Numbernumber must start with 4930XXX. 2=National Number is in national Formatformat. 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 (CalleridCallerID) 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 CalleridCallerID 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 Typetype of number the redirected Numbernumber has. Default is 0. cpnnumplan Type Of Number (Connected Party) 0,1,2,4 Like onumplan, to indicat what Typetype 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 Prefixprefix will be used. Default: none internationalprefix International Prefix x - prefix When an incoming call has an international Calling Party Number, the configured Prefixprefix will be used. Default: 00 nationalprefix National Prefix x - prefix When an incoming call has a national Calling Party Number, the configured Prefixprefix will be used. Default: 0 localprefix Local Prefix x - prefix When an incoming call has a local Calling Party Number, the configured Prefixprefix will be used. Default: none privateprefix Private Prefix x - prefix When an incoming call has a Private Calling Party Number, the configured Prefixprefix 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 bearerBearer Capability (Type of Data) will be transmitted in the B-Channel. For normal Speechspeech 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, berofixthe beroNet Gateway will try to deflect calls on reception of a "302 Moved Temporary" to the given Destination. On PMP PMPlines Linesthe beroFixgateway will send a Calldeflect on PP Lineslines 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 Thethe Telco only allows sending of audio in the Connectedconnected Statestate. 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 Ringingringing tone by ourself. Default: no. oad_setting OAD Setting fromuser,displayname Defines if the fromuser Partpart or the displayname Partpart of the FROM Header should be transmitted as CalleridCallerID (oad). Default: fromuser. ignorep8 Ignore Progress Indicator 8 yes/no If set to yes, berofixthe 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 BchannelB-Channel after they've send a Progress Indicator (8), otherwise a disturbing noise is hearedheard. On the normal telephone line the BchannelB-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, berofixthe beroNet Gateway will send any aschiiASCII character as ISDN Numbersnumbers. Otherwise it will only send numeric numbers and discard others. Default: no. featurecodes DTMF Feature Codes codes It is possible, that the beroNet berofixGateway takes some actions during a call when the user has entered a predefined DTMF Tone Sequence. These Actionsactions include sending of specific ISDN Supplementary Services, that are not yet map-able to SIP Methodsmethods. Currently only the MCID (Malicious Caller Identification) Featurefeature is supported. The "featurecodes" string has 3 Parametersparameters, 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 Digitsdigits, 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 beroFixgateway will send out the ISDN Facility MCID.