; ; DAHDI Telephony Configuration file ; ; You need to restart Asterisk to re-configure the DAHDI channel ; CLI> module reload chan_dahdi.so ; will reload the configuration file, but not all configuration options ; are re-configured during a reload (signalling, as well as PRI and ; SS7-related settings cannot be changed on a reload). ; ; This file documents many configuration variables. Normally unless you know ; what a variable means or that it should be changed, there's no reason to ; un-comment those lines. ; ; Examples below that are commented out (those lines that begin with a ';' but ; no space afterwards) typically show a value that is not the default value, ; but would make sense under certain circumstances. The default values are ; usually sane. Thus you should typically not touch them unless you know what ; they mean or you know you should change them. [trunkgroups] ; ; Trunk groups are used for NFAS connections. ; ; Group: Defines a trunk group. ; trunkgroup => ,[,...] ; ; trunkgroup is the numerical trunk group to create ; dchannel is the DAHDI channel which will have the ; d-channel for the trunk. ; backup1 is an optional list of backup d-channels. ; ;trunkgroup => 1,24,48 ;trunkgroup => 1,24 ; ; Spanmap: Associates a span with a trunk group ; spanmap => ,[,] ; ; dahdispan is the DAHDI span number to associate ; trunkgroup is the trunkgroup (specified above) for the mapping ; logicalspan is the logical span number within the trunk group to use. ; if unspecified, no logical span number is used. ; ;spanmap => 1,1,1 ;spanmap => 2,1,2 ;spanmap => 3,1,3 ;spanmap => 4,1,4 [channels] ; ; Default language ; ;language=en ; ; Context for incoming calls. Defaults to 'default' ; context=public ; ; Switchtype: Only used for PRI. ; ; national: National ISDN 2 (default) ; dms100: Nortel DMS100 ; 4ess: AT&T 4ESS ; 5ess: Lucent 5ESS ; euroisdn: EuroISDN (common in Europe) ; ni1: Old National ISDN 1 ; qsig: Q.SIG ; ;switchtype=euroisdn ; ; MSNs for ISDN spans. Asterisk will listen for the listed numbers on ; incoming calls and ignore any calls not listed. ; Here you can give a comma separated list of numbers or dialplan extension ; patterns. An empty list disables MSN matching to allow any incoming call. ; Only set on PTMP CPE side of ISDN span if needed. ; The default is an empty list. ;msn= ; ; Some switches (AT&T especially) require network specific facility IE. ; Supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet' ; ; nsf cannot be changed on a reload. ; ;nsf=none ; ;service_message_support=yes ; Enable service message support for channel. Must be set after switchtype. ; ; Dialing options for ISDN (i.e., Dial(DAHDI/g1/exten/options)): ; R Reverse Charge Indication ; Indicate to the called party that the call will be reverse charged. ; K(n) Keypad digits n ; Send out the specified digits as keypad digits. ; ; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for ; the dialed number. Leaving this as 'unknown' (the default) works for most ; cases. In some very unusual circumstances, you may need to set this to ; 'dynamic' or 'redundant'. ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; dynamic: Dynamically selects the appropriate dialplan using the ; prefix settings. ; redundant: Same as dynamic, except that the underlying number is not ; changed (not common) ; ; pridialplan cannot be changed on reload. ;pridialplan=unknown ; ; PRI Local Dialplan: Only RARELY used for PRI (sets the calling number's ; numbering plan). In North America, the typical use is sending the 10 digit ; callerID number and setting the prilocaldialplan to 'national' (the default). ; Only VERY rarely will you need to change this. ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; from_channel: Use the CALLERID(ton) value from the channel. ; dynamic: Dynamically selects the appropriate dialplan using the ; prefix settings. ; redundant: Same as dynamic, except that the underlying number is not ; changed (not common) ; ; prilocaldialplan cannot be changed on reload. ;prilocaldialplan=national ; ; PRI Connected Line Dialplan: Sets the connected party number's numbering plan. ; ; unknown: Unknown ; private: Private ISDN ; local: Local ISDN ; national: National ISDN ; international: International ISDN ; from_channel: Use the CONNECTEDLINE(ton) value from the channel. ; dynamic: Dynamically selects the appropriate dialplan using the ; prefix settings. ; redundant: Same as dynamic, except that the underlying number is not ; changed (not common) ; ; pricpndialplan cannot be changed on reload. ;pricpndialplan=from_channel ; ; pridialplan may be also set at dialtime, by prefixing the dialed number with ; one of the following letters: ; U - Unknown ; I - International ; N - National ; L - Local (Net Specific) ; S - Subscriber ; V - Abbreviated ; R - Reserved (should probably never be used but is included for completeness) ; ; Additionally, you may also set the following NPI bits (also by prefixing the ; dialed string with one of the following letters): ; u - Unknown ; e - E.163/E.164 (ISDN/telephony) ; x - X.121 (Data) ; f - F.69 (Telex) ; n - National ; p - Private ; r - Reserved (should probably never be used but is included for completeness) ; ; You may also set the prilocaldialplan in the same way, but by prefixing the ; Caller*ID Number rather than the dialed number. ; Please note that telcos which require this kind of additional manipulation ; of the TON/NPI are *rare*. Most telco PRIs will work fine simply by ; setting pridialplan to unknown or dynamic. ; ; ; PRI caller ID prefixes based on the given TON/NPI (dialplan) ; This is especially needed for EuroISDN E1-PRIs ; ; None of the prefix settings can be changed on reload. ; ; sample 1 for Germany ;internationalprefix = 00 ;nationalprefix = 0 ;localprefix = 0711 ;privateprefix = 07115678 ;unknownprefix = ; ; sample 2 for Germany ;internationalprefix = + ;nationalprefix = +49 ;localprefix = +49711 ;privateprefix = +497115678 ;unknownprefix = ; ; PRI resetinterval: sets the time in seconds between restart of unused ; B channels; defaults to 'never'. ; ;resetinterval = 3600 ; ; Enable per ISDN span to force a RESTART on a channel that returns a cause ; code of PRI_CAUSE_REQUESTED_CHAN_UNAVAIL(44). If this option is enabled ; and the reason the peer rejected the call with cause 44 was that the ; channel is stuck in an unavailable state on the peer, then this might ; help release the channel. It is worth noting that the next outgoing call ; Asterisk makes will likely try the same channel again. ; ; NOTE: Sending a RESTART in response to a cause 44 is not required ; (nor prohibited) by the standards and is likely a primitive chan_dahdi ; response to call collisions (glare) and buggy peers. However, there ; are telco switches out there that ignore the RESTART and continue to ; send calls to the channel in the restarting state. ; Default no. ; ;force_restart_unavailable_chans=yes ; ; Assume inband audio may be present when a SETUP ACK message is received. ; Q.931 Section 5.1.3 says that in scenarios with overlap dialing, when a ; dialtone is sent from the network side, progress indicator 8 "Inband info ; now available" MAY be sent to the CPE if no digits were received with ; the SETUP. It is thus implied that the ie is mandatory if digits came ; with the SETUP and dialtone is needed. ; This option should be enabled, when the network sends dialtone and you ; want to hear it, but the network doesn't send the progress indicator when ; needed. ; ; NOTE: For Q.SIG setups this option should be enabled when outgoing overlap ; dialing is also enabled because Q.SIG does not send the progress indicator ; with the SETUP ACK. ; Default no. ; ;inband_on_setup_ack=yes ; ; Assume inband audio may be present when a PROCEEDING message is received. ; Q.931 Section 5.1.2 says the network cannot assume that the CPE side has ; attached to the B channel at this time without explicitly sending the ; progress indicator ie informing the CPE side to attach to the B channel ; for audio. However, some non-compliant ISDN switches send a PROCEEDING ; without the progress indicator ie indicating inband audio is available and ; assume that the CPE device has connected the media path for listening to ; ringback and other messages. ; Default no. ; ;inband_on_proceeding=yes ; ; Overlap dialing mode (sending overlap digits) ; Cannot be changed on a reload. ; ; incoming: incoming direction only ; outgoing: outgoing direction only ; no: neither direction ; yes or both: both directions ; ;overlapdial=yes ; Send/receive ISDN display IE options. The display options are a comma separated ; list of the following options: ; ; block: Do not pass display text data. ; Q.SIG: Default for send/receive. ; ETSI CPE: Default for send. ; name_initial: Use display text in SETUP/CONNECT messages as the party name. ; Default for all other modes. ; name_update: Use display text in other messages (NOTIFY/FACILITY) for COLP name ; update. ; name: Combined name_initial and name_update options. ; text: Pass any unused display text data as an arbitrary display message ; during a call. Sent text goes out in an INFORMATION message. ; ; * Default is an empty string for legacy behavior. ; * The name options are not recommended for Q.SIG since Q.SIG already ; supports names. ; * The send block is the only recommended setting for CPE mode since Q.931 uses ; the display IE only in the network to user direction. ; ; display_send and display_receive cannot be changed on reload. ; ;display_send= ;display_receive= ; Allow sending an ISDN Malicious Caller ID (MCID) request on this span. ; Default disabled ; ;mcid_send=yes ; Send ISDN date/time IE in CONNECT message option. Only valid on NT spans. ; ; no: Do not send date/time IE in CONNECT message. ; date: Send date only. ; date_hh Send date and hour. ; date_hhmm Send date, hour, and minute. ; date_hhmmss Send date, hour, minute, and second. ; ; Default is an empty string which lets libpri pick the default ; date/time IE send policy. ; ;datetime_send= ; Send ISDN conected line information. ; ; block: Do not send any connected line information. ; connect: Send connected line information on initial connect. ; update: Same as connect but also send any updates during a call. ; Updates happen if the call is transferred. (Default) ; ;colp_send=update ; Allow inband audio (progress) when a call is DISCONNECTed by the far end of a PRI ; ;inbanddisconnect=yes ; ; Allow a held call to be transferred to the active call on disconnect. ; This is useful on BRI PTMP NT lines where an ISDN phone can simulate the ; transfer feature of an analog phone. ; The default is no. ;hold_disconnect_transfer=yes ; BRI PTMP layer 1 presence. ; You should normally not need to set this option. ; You may need to set this option if your telco brings layer 1 down when ; the line is idle. ; required: Layer 1 presence required for outgoing calls. (default) ; ignore: Ignore alarms from DAHDI about this span. ; (Layer 1 and 2 will be brought back up for an outgoing call.) ; NOTE: You will not be able to detect physical line problems ; until an outgoing call is attempted and fails. ; ;layer1_presence=ignore ; BRI PTMP layer 2 persistence. ; You should normally not need to set this option. ; You may need to set this option if your telco brings layer 1 down when ; the line is idle. ; : Use libpri default. ; keep_up: Bring layer 2 back up if peer takes it down. ; leave_down: Leave layer 2 down if peer takes it down. (Libpri default) ; (Layer 2 will be brought back up for an outgoing call.) ; ;layer2_persistence=leave_down ; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; with all telcos. ; ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones (default) ; ; priindication cannot be changed on a reload. ; ;priindication = outofband ; ; If you need to override the existing channels selection routine and force all ; PRI channels to be marked as exclusively selected, set this to yes. ; ; priexclusive cannot be changed on a reload. ; ;priexclusive = yes ; ; ; If you need to use the logical channel mapping with your Q.SIG PRI instead ; of the physical mapping you must use the qsigchannelmapping option. ; ; logical: Use the logical channel mapping ; physical: Use physical channel mapping (default) ; ;qsigchannelmapping=logical ; ; If you wish to ignore remote hold indications (and use MOH that is supplied over ; the B channel) enable this option. ; ;discardremoteholdretrieval=yes ; ; ISDN Timers ; All of the ISDN timers and counters that are used are configurable. Specify ; the timer name, and its value (in ms for timers). ; K: Layer 2 max number of outstanding unacknowledged I frames (default 7) ; N200: Layer 2 max number of retransmissions of a frame (default 3) ; T200: Layer 2 max time before retransmission of a frame (default 1000 ms) ; T203: Layer 2 max time without frames being exchanged (default 10000 ms) ; T305: Wait for DISCONNECT acknowledge (default 30000 ms) ; T308: Wait for RELEASE acknowledge (default 4000 ms) ; T309: Maintain active calls on Layer 2 disconnection (default 6000 ms) ; EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s ; May vary in other ISDN standards (Q.931 1993 : 90000 ms) ; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms) ; ; T-RESPONSE: Maximum time to wait for a typical APDU response. (default 4000 ms) ; This is an implementation timer when the standard does not specify one. ; T-ACTIVATE: Request supervision timeout. (default 10000 ms) ; T-RETENTION: Maximum time to wait for user A to activate call-completion. (default 30000 ms) ; Used by ETSI PTP, ETSI PTMP, and Q.SIG as the cc_offer_timer. ; T-CCBS1: T-STATUS timer equivalent for CC user A status. (default 4000 ms) ; T-CCBS2: Maximum time the CCBS service will be active (default 45 min in ms) ; T-CCBS3: Maximum time to wait for user A to respond to user B availability. (default 20000 ms) ; T-CCBS5: Network B CCBS supervision timeout. (default 60 min in ms) ; T-CCBS6: Network A CCBS supervision timeout. (default 60 min in ms) ; T-CCNR2: Maximum time the CCNR service will be active (default 180 min in ms) ; T-CCNR5: Network B CCNR supervision timeout. (default 195 min in ms) ; T-CCNR6: Network A CCNR supervision timeout. (default 195 min in ms) ; CC-T1: Q.SIG CC request supervision timeout. (default 30000 ms) ; CCBS-T2: Q.SIG CCBS supervision timeout. (default 60 min in ms) ; CCNR-T2: Q.SIG CCNR supervision timeout. (default 195 min in ms) ; CC-T3: Q.SIG CC Maximum time to wait for user A to respond to user B availability. (default 30000 ms) ; ;pritimer => t200,1000 ;pritimer => t313,4000 ; ; CC PTMP recall mode: ; specific - Only the CC original party A can participate in the CC callback ; global - Other compatible endpoints on the PTMP line can be party A in the CC callback ; ; cc_ptmp_recall_mode cannot be changed on a reload. ; ;cc_ptmp_recall_mode = specific ; ; CC Q.SIG Party A (requester) retain signaling link option ; retain Require that the signaling link be retained. ; release Request that the signaling link be released. ; do_not_care The responder is free to choose if the signaling link will be retained. ; ;cc_qsig_signaling_link_req = retain ; ; CC Q.SIG Party B (responder) retain signaling link option ; retain Prefer that the signaling link be retained. ; release Prefer that the signaling link be released. ; ;cc_qsig_signaling_link_rsp = retain ; ; See ccss.conf.sample for more options. The timers described by ccss.conf.sample ; are not used by ISDN for the native protocol since they are defined by the ; standards and set by pritimer above. ; ; To enable transmission of facility-based ISDN supplementary services (such ; as caller name from CPE over facility), enable this option. ; Cannot be changed on a reload. ; ;facilityenable = yes ; ; This option enables Advice of Charge pass-through between the ISDN PRI and ; Asterisk. This option can be set to any combination of 's', 'd', and 'e' which ; represent the different variants of Advice of Charge, AOC-S, AOC-D, and AOC-E. ; Advice of Charge pass-through is currently only supported for ETSI. Since most ; AOC messages are sent on facility messages, the 'facilityenable' option must ; also be enabled to fully support AOC pass-through. ; ;aoc_enable=s,d,e ; ; When this option is enabled, a hangup initiated by the ISDN PRI side of the ; asterisk channel will result in the channel delaying its hangup in an ; attempt to receive the final AOC-E message from its bridge. The delay ; period is configured as one half the T305 timer length. If the channel ; is not bridged the hangup will occur immediatly without delay. ; ;aoce_delayhangup=yes ; pritimer cannot be changed on a reload. ; ; Signalling method. The default is "auto". Valid values: ; auto: Use the current value from DAHDI. ; em: E & M ; em_e1: E & M E1 ; em_w: E & M Wink ; featd: Feature Group D (The fake, Adtran style, DTMF) ; featdmf: Feature Group D (The real thing, MF (domestic, US)) ; featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through ; a Tandem Access point ; featb: Feature Group B (MF (domestic, US)) ; fgccama: Feature Group C-CAMA (DP DNIS, MF ANI) ; fgccamamf: Feature Group C-CAMA MF (MF DNIS, MF ANI) ; fxs_ls: FXS (Loop Start) ; fxs_gs: FXS (Ground Start) ; fxs_ks: FXS (Kewl Start) ; fxo_ls: FXO (Loop Start) ; fxo_gs: FXO (Ground Start) ; fxo_ks: FXO (Kewl Start) ; pri_cpe: PRI signalling, CPE side ; pri_net: PRI signalling, Network side ; bri_cpe: BRI PTP signalling, CPE side ; bri_net: BRI PTP signalling, Network side ; bri_cpe_ptmp: BRI PTMP signalling, CPE side ; bri_net_ptmp: BRI PTMP signalling, Network side ; sf: SF (Inband Tone) Signalling ; sf_w: SF Wink ; sf_featd: SF Feature Group D (The fake, Adtran style, DTMF) ; sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US)) ; sf_featb: SF Feature Group B (MF (domestic, US)) ; e911: E911 (MF) style signalling ; ss7: Signalling System 7 ; mfcr2: MFC/R2 Signalling. To specify the country variant see 'mfcr2_variant' ; ; The following are used for Radio interfaces: ; fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the ; channel bank) ; fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the ; channel bank) ; fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the ; channel bank) ; fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at ; the channel bank) ; em_rx: Receive audio/COR on an E&M interface (1-way) ; em_tx: Transmit audio/PTT on an E&M interface (1-way) ; em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface ; (2-way) ; em_rxtx: Same as em_txrx (for our dyslexic friends) ; sf_rx: Receive audio/COR on an SF interface (1-way) ; sf_tx: Transmit audio/PTT on an SF interface (1-way) ; sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface ; (2-way) ; sf_rxtx: Same as sf_txrx (for our dyslexic friends) ; ss7: Signalling System 7 ; ; signalling of a channel can not be changed on a reload. ; ;signalling=fxo_ls ; ; If you have an outbound signalling format that is different from format ; specified above (but compatible), you can specify outbound signalling format, ; (see below). The 'signalling' format specified will be the inbound signalling ; format. If you only specify 'signalling', then it will be the format for ; both inbound and outbound. ; ; outsignalling can only be one of: ; em, em_e1, em_w, sf, sf_w, sf_featd, sf_featdmf, sf_featb, featd, ; featdmf, featdmf_ta, e911, fgccama, fgccamamf ; ; outsignalling cannot be changed on a reload. ; ;signalling=featdmf ; ;outsignalling=featb ; ; For Feature Group D Tandem access, to set the default CIC and OZZ use these ; parameters (Will not be updated on reload): ; ;defaultozz=0000 ;defaultcic=303 ; ; A variety of timing parameters can be specified as well ; The default values for those are "-1", which is to use the ; compile-time defaults of the DAHDI kernel modules. The timing ; parameters, (with the standard default from DAHDI): ; ; prewink: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ; rxflash: Receiver flashtime (default 1250ms) ; debounce: Debounce timing (default 600ms) ; ; None of them will update on a reload. ; ; How long generated tones (DTMF and MF) will be played on the channel ; (in milliseconds). ; ; This is a global, rather than a per-channel setting. It will not be ; updated on a reload. ; ;toneduration=100 ; ; Whether or not to do distinctive ring detection on FXO lines: ; ;usedistinctiveringdetection=yes ; ; enable dring detection after caller ID for those countries like Australia ; where the ring cadence is changed *after* the caller ID spill: ; ;distinctiveringaftercid=yes ; ; Whether or not to use caller ID: ; usecallerid=yes ; ; Type of caller ID signalling in use ; bell = bell202 as used in US (default) ; v23 = v23 as used in the UK ; v23_jp = v23 as used in Japan ; dtmf = DTMF as used in Denmark, Sweden and Netherlands ; smdi = Use SMDI for caller ID. Requires SMDI to be enabled (usesmdi). ; ;cidsignalling=v23 ; ; What signals the start of caller ID ; ring = a ring signals the start (default) ; polarity = polarity reversal signals the start ; polarity_IN = polarity reversal signals the start, for India, ; for dtmf dialtone detection; using DTMF. ; (see https://wiki.asterisk.org/wiki/display/AST/Caller+ID+in+India) ; dtmf = causes monitor loop to look for dtmf energy on the ; incoming channel to initate cid acquisition ; ;cidstart=polarity ; ; When cidstart=dtmf, the energy level on the line used to trigger dtmf cid ; acquisition. This number is compared to the average over a packet of audio ; of the absolute values of 16 bit signed linear samples. The default is set ; to 256. The choice of 256 is arbitrary. The value you should select should ; be high enough to prevent false detections while low enough to insure that ; no dtmf spills are missed. ; ;dtmfcidlevel=256 ; ; Whether or not to hide outgoing caller ID (Override with *67 or *82) ; (If your dialplan doesn't catch it) ; ;hidecallerid=yes ; ; Enable if you need to hide just the name and not the number for legacy PBX use. ; Only applies to PRI channels. ;hidecalleridname=yes ; ; On UK analog lines, the caller hanging up determines the end of calls. So ; Asterisk hanging up the line may or may not end a call (DAHDI could just as ; easily be re-attaching to a prior incoming call that was not yet hung up). ; This option changes the hangup to wait for a dialtone on the line, before ; marking the line as once again available for use with outgoing calls. ; Specified in milliseconds, not set by default. ;waitfordialtone=1000 ; ; For analog lines, enables Asterisk to use dialtone detection per channel ; if an incoming call was hung up before it was answered. If dialtone is ; detected, the call is hung up. ; no: Disabled. (Default) ; yes: Look for dialtone for 10000 ms after answer. ; : Look for dialtone for the specified number of ms after answer. ; always: Look for dialtone for the entire call. Dialtone may return ; if the far end hangs up first. ; ;dialtone_detect=no ; ; The following option enables receiving MWI on FXO lines. The default ; value is no. ; The mwimonitor can take the following values ; no - No mwimonitoring occurs. (default) ; yes - The same as specifying fsk ; fsk - the FXO line is monitored for MWI FSK spills ; fsk,rpas - the FXO line is monitored for MWI FSK spills preceded ; by a ring pulse alert signal. ; neon - The fxo line is monitored for the presence of NEON pulses ; indicating MWI. ; When detected, an internal Asterisk MWI event is generated so that any other ; part of Asterisk that cares about MWI state changes is notified, just as if ; the state change came from app_voicemail. ; For FSK MWI Spills, the energy level that must be seen before starting the ; MWI detection process can be set with 'mwilevel'. ; ;mwimonitor=no ;mwilevel=512 ; ; This option is used in conjunction with mwimonitor. This will get executed ; when incoming MWI state changes. The script is passed 2 arguments. The ; first is the corresponding configured mailbox, and the second is 1 or 0, ; indicating if there are messages waiting or not. ; Note: app_voicemail mailboxes are in the form of mailbox@context. ; ; /usr/local/bin/dahdinotify.sh 501@mailboxes 1 ; ;mwimonitornotify=/usr/local/bin/dahdinotify.sh ; ; The following keyword 'mwisendtype' enables various VMWI methods on FXS lines (if supported). ; The default is to send FSK only. ; The following options are available; ; 'rpas' Ring Pulse Alert Signal, alerts intelligent phones that a FSK message is about to be sent. ; 'lrev' Line reversed to indicate messages waiting. ; 'hvdc' 90Vdc OnHook DC voltage to indicate messages waiting. ; 'hvac' or 'neon' 90Vac OnHook AC voltage to light Neon bulb. ; 'nofsk' Disables FSK MWI spills from being sent out. ; It is feasible that multiple options can be enabled. ;mwisendtype=rpas,lrev ; ; Whether or not to enable call waiting on internal extensions ; With this set to 'yes', busy extensions will hear the call-waiting ; tone, and can use hook-flash to switch between callers. The Dial() ; app will not return the "BUSY" result for extensions. ; callwaiting=yes ; ; Configure the number of outstanding call waiting calls for internal ISDN ; endpoints before bouncing the calls as busy. This option is equivalent to ; the callwaiting option for analog ports. ; A call waiting call is a SETUP message with no B channel selected. ; The default is zero to disable call waiting for ISDN endpoints. ;max_call_waiting_calls=0 ; ; Allow incoming ISDN call waiting calls. ; A call waiting call is a SETUP message with no B channel selected. ;allow_call_waiting_calls=no ; Configure the ISDN span to indicate MWI for the list of mailboxes. ; You can give a comma separated list of up to 8 mailboxes per span. ; An empty list disables MWI. ; ; The default is an empty list. ;mwi_mailboxes=vm-mailbox{,vm-mailbox} ; vm-mailbox = Internal voicemail mailbox identifier. ; Note: app_voicemail mailboxes must be in the form of mailbox@context. ;mwi_mailboxes=501@mailboxes,502@mailboxes ; Configure the ISDN mailbox number sent over the span for MWI mailboxes. ; The position of the number in the list corresponds to the position in ; mwi_mailboxes. If either position in mwi_mailboxes or mwi_vm_boxes is ; empty then that position is disabled. ; ; The default is an empty list. ;mwi_vm_boxes=mailbox_number{,mailbox_number} ;mwi_vm_boxes=501,502 ; Configure the ISDN span voicemail controlling numbers for MWI mailboxes. ; What number to call for a user to retrieve voicemail messages. ; ; You can give a comma separated list of numbers. The position of the number ; corresponds to the position in mwi_mailboxes. If a position is empty then ; the last number is reused. ; ; For example: ; mwi_vm_numbers=700,,800,,900 ; is equivalent to: ; mwi_vm_numbers=700,700,800,800,900,900,900,900 ; ; The default is no number. ;mwi_vm_numbers= ; Whether or not restrict outgoing caller ID (will be sent as ANI only, not ; available for the user) ; Mostly use with FXS ports ; Does nothing. Use hidecallerid instead. ; ;restrictcid=no ; ; Whether or not to use the caller ID presentation from the Asterisk channel ; for outgoing calls. ; See dialplan function CALLERID(pres) for more information. ; Only applies to PRI and SS7 channels. ; usecallingpres=yes ; ; Some countries (UK) have ring tones with different ring tones (ring-ring), ; which means the caller ID needs to be set later on, and not just after ; the first ring, as per the default (1). ; ;sendcalleridafter = 2 ; ; ; Support caller ID on Call Waiting ; callwaitingcallerid=yes ; ; Support three-way calling ; threewaycalling=yes ; ; For FXS ports (either direct analog or over T1/E1): ; Support flash-hook call transfer (requires three way calling) ; Also enables call parking (overrides the 'canpark' parameter) ; ; For digital ports using ISDN PRI protocols: ; Support switch-side transfer (called 2BCT, RLT or other names) ; This setting must be enabled on both ports involved, and the ; 'facilityenable' setting must also be enabled to allow sending ; the transfer to the ISDN switch, since it sent in a FACILITY ; message. ; NOTE: This should be disabled for NT PTMP mode. Phones cannot ; have tromboned calls pushed down to them. ; transfer=yes ; ; Allow call parking ; ('canpark=no' is overridden by 'transfer=yes') ; canpark=yes ; Sets the default parking lot for call parking. ; This is setable per channel. ; Parkinglots are configured in features.conf ; ;parkinglot=plaza ; ; Support call forward variable ; cancallforward=yes ; ; Whether or not to support Call Return (*69, if your dialplan doesn't ; catch this first) ; callreturn=yes ; ; Stutter dialtone support: If voicemail is received in the mailbox then ; taking the phone off hook will cause a stutter dialtone instead of a ; normal one. ; ; Note: app_voicemail mailboxes must be in the form of mailbox@context. ; ;mailbox=1234@context ; ; Enable echo cancellation ; Use either "yes", "no", or a power of two from 32 to 256 if you wish to ; actually set the number of taps of cancellation. ; ; Note that when setting the number of taps, the number 256 does not translate ; to 256 ms of echo cancellation. echocancel=256 means 256 / 8 = 32 ms. ; ; Note that if any of your DAHDI cards have hardware echo cancellers, ; then this setting only turns them on and off; numeric settings will ; be treated as "yes". There are no special settings required for ; hardware echo cancellers; when present and enabled in their kernel ; modules, they take precedence over the software echo canceller compiled ; into DAHDI automatically. ; ; echocancel=yes ; ; Some DAHDI echo cancellers (software and hardware) support adjustable ; parameters; these parameters can be supplied as additional options to ; the 'echocancel' setting. Note that Asterisk does not attempt to ; validate the parameters or their values, so if you supply an invalid ; parameter you will not know the specific reason it failed without ; checking the kernel message log for the error(s) put there by DAHDI. ; ;echocancel=128,param1=32,param2=0,param3=14 ; ; Generally, it is not necessary (and in fact undesirable) to echo cancel when ; the circuit path is entirely TDM. You may, however, change this behavior ; by enabling the echo canceller during pure TDM bridging below. ; echocancelwhenbridged=yes ; ; In some cases, the echo canceller doesn't train quickly enough and there ; is echo at the beginning of the call. Enabling echo training will cause ; DAHDI to briefly mute the channel, send an impulse, and use the impulse ; response to pre-train the echo canceller so it can start out with a much ; closer idea of the actual echo. Value may be "yes", "no", or a number of ; milliseconds to delay before training (default = 400) ; ; WARNING: In some cases this option can make echo worse! If you are ; trying to debug an echo problem, it is worth checking to see if your echo ; is better with the option set to yes or no. Use whatever setting gives ; the best results. ; ; Note that these parameters do not apply to hardware echo cancellers. ; ;echotraining=yes ;echotraining=800 ; ; If you are having trouble with DTMF detection, you can relax the DTMF ; detection parameters. Relaxing them may make the DTMF detector more likely ; to have "talkoff" where DTMF is detected when it shouldn't be. ; ;relaxdtmf=yes ; ; Hardware gain settings increase/decrease the analog volume level on a channel. ; The values are in db (decibels) and can be adjusted in 0.1 dB increments. ; A positive number increases the volume level on a channel, and a negavive ; value decreases volume level. ; ; Hardware gain settings are only possible on hardware with analog ports ; because the gain is done on the analog side of the analog/digital conversion. ; ; When hardware gains are disabled, Asterisk will NOT touch the gain setting ; already configured in hardware. ; ; hwrxgain: Hardware receive gain for the channel (into Asterisk). ; Default: disabled ; hwtxgain: Hardware transmit gain for the channel (out of Asterisk). ; Default: disabled ; ;hwrxgain=disabled ;hwtxgain=disabled ;hwrxgain=2.0 ;hwtxgain=3.0 ; ; Software gain settings digitally increase/decrease the volume level on a channel. ; The values are in db (decibels). A positive number increases the volume ; level on a channel, and a negavive value decreases volume level. ; ; Software gains work on the digital side of the analog/digital conversion ; and thus can also work with T1/E1 cards. ; ; rxgain: Software receive gain for the channel (into Asterisk). Default: 0.0 ; txgain: Software transmit gain for the channel (out of Asterisk). ; Default: 0.0 ; ; cid_rxgain: Add this gain to rxgain when Asterisk expects to receive ; a Caller ID stream. ; Default: 5.0 . ; ;rxgain=2.0 ;txgain=3.0 ; ; Dynamic Range Compression: You can also enable dynamic range compression ; on a channel. This will digitally amplify quiet sounds while leaving louder ; sounds untouched. This is useful in situations where a linear gain setting ; would cause clipping. Acceptable values are in the range of 0.0 to around ; 6.0 with higher values causing more compression to be done. ; ; rxdrc: dynamic range compression for the rx channel. Default: 0.0 ; txdrc: dynamic range compression for the tx channel. Default: 0.0 ; ;rxdrc=1.0 ;txdrc=4.0 ; ; Logical groups can be assigned to allow outgoing roll-over. Groups range ; from 0 to 63, and multiple groups can be specified. By default the ; channel is not a member of any group. ; ; Note that an explicit empty value for 'group' is invalid, and will not ; override a previous non-empty one. The same applies to callgroup and ; pickupgroup as well. ; group=1 ; ; Ring groups (a.k.a. call groups) and pickup groups. If a phone is ringing ; and it is a member of a group which is one of your pickup groups, then ; you can answer it by picking up and dialing *8#. For simple offices, just ; make these both the same. Groups range from 0 to 63. ; callgroup=1 pickupgroup=1 ; ; Named ring groups (a.k.a. named call groups) and named pickup groups. ; If a phone is ringing and it is a member of a group which is one of your ; named pickup groups, then you can answer it by picking up and dialing *8#. ; For simple offices, just make these both the same. ; The number of named groups is not limited. ; ;namedcallgroup=engineering,sales,netgroup,protgroup ;namedpickupgroup=sales ; Channel variables to be set for all calls from this channel ;setvar=CHANNEL=42 ;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will ; cause the given audio file to ; be played upon completion of ; an attended transfer to the ; target of the transfer. ; ; Specify whether the channel should be answered immediately or if the simple ; switch should provide dialtone, read digits, etc. ; Note: If immediate=yes the dialplan execution will always start at extension ; 's' priority 1 regardless of the dialed number! ; ;immediate=yes ; ; Specify whether flash-hook transfers to 'busy' channels should complete or ; return to the caller performing the transfer (default is yes). ; ;transfertobusy=no ; Calls will have the party id user tag set to this string value. ; ;cid_tag= ; With this set, you can automatically append the MSN of a party ; to the cid_tag. An '_' is used to separate the tag from the MSN. ; Applies to ISDN spans. ; Default is no. ; ; Table of what number is appended: ; outgoing incoming ; net dialed caller ; cpe caller dialed ; ;append_msn_to_cid_tag=no ; caller ID can be set to "asreceived" or a specific number if you want to ; override it. Note that "asreceived" only applies to trunk interfaces. ; fullname sets just the ; ; fullname: sets just the name part. ; cid_number: sets just the number part: ; ;callerid = 123456 ; ;callerid = My Name <2564286000> ; Which can also be written as: ;cid_number = 2564286000 ;fullname = My Name ; ;callerid = asreceived ; ; should we use the caller ID from incoming call on DAHDI transfer? ; ;useincomingcalleridondahditransfer = yes ; ; Add a description for the channel which can be shown through the Asterisk ; console when executing the 'dahdi show channels' command is run. ; ;description=Phone located in lobby ; ; AMA flags affects the recording of Call Detail Records. If specified ; it may be 'default', 'omit', 'billing', or 'documentation'. ; ;amaflags=default ; ; Channels may be associated with an account code to ease ; billing ; ;accountcode=lss0101 ; ; ADSI (Analog Display Services Interface) can be enabled on a per-channel ; basis if you have (or may have) ADSI compatible CPE equipment ; ;adsi=yes ; ; SMDI (Simplified Message Desk Interface) can be enabled on a per-channel ; basis if you would like that channel to behave like an SMDI message desk. ; The SMDI port specified should have already been defined in smdi.conf. The ; default port is /dev/ttyS0. ; ;usesmdi=yes ;smdiport=/dev/ttyS0 ; ; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D ; etc, it can be useful to perform busy detection either in an effort to ; detect hangup or for detecting busies. This enables listening for ; the beep-beep busy pattern. ; ;busydetect=yes ; ; If busydetect is enabled, it is also possible to specify how many busy tones ; to wait for before hanging up. The default is 3, but it might be ; safer to set to 6 or even 8. Mind that the higher the number, the more ; time that will be needed to hangup a channel, but lowers the probability ; that you will get random hangups. ; ;busycount=6 ; ; If busydetect is enabled, it is also possible to specify the cadence of your ; busy signal. In many countries, it is 500msec on, 500msec off. Without ; busypattern specified, we'll accept any regular sound-silence pattern that ; repeats times as a busy signal. If you specify busypattern, ; then we'll further check the length of the sound (tone) and silence, which ; will further reduce the chance of a false positive. ; ;busypattern=500,500 ; ; NOTE: In make menuselect, you'll find further options to tweak the busy ; detector. If your country has a busy tone with the same length tone and ; silence (as many countries do), consider enabling the ; BUSYDETECT_COMPARE_TONE_AND_SILENCE option. ; ; To further detect which hangup tone your telco provider is sending, it is ; useful to use the dahdi_monitor utility to record the audio that main/dsp.c ; is receiving after the caller hangs up. ; ; For FXS (FXO signalled) ports ; switch the line polarity to signal the connected PBX that an outgoing ; call was answered by the remote party. ; For FXO (FXS signalled) ports ; watch for a polarity reversal to mark when a outgoing call is ; answered by the remote party. ; ;answeronpolarityswitch=yes ; ; For FXS (FXO signalled) ports ; switch the line polarity to signal the connected PBX that the current ; call was "hung up" by the remote party ; For FXO (FXS signalled) ports ; In some countries, a polarity reversal is used to signal the disconnect of a ; phone line. If the hanguponpolarityswitch option is selected, the call will ; be considered "hung up" on a polarity reversal. ; ;hanguponpolarityswitch=yes ; ; polarityonanswerdelay: minimal time period (ms) between the answer ; polarity switch and hangup polarity switch. ; (default: 600ms) ; ; On trunk interfaces (FXS) it can be useful to attempt to follow the progress ; of a call through RINGING, BUSY, and ANSWERING. If turned on, call ; progress attempts to determine answer, busy, and ringing on phone lines. ; This feature is HIGHLY EXPERIMENTAL and can easily detect false answers, ; so don't count on it being very accurate. ; ; Few zones are supported at the time of this writing, but may be selected ; with "progzone". ; ; progzone also affects the pattern used for busydetect (unless ; busypattern is set explicitly). The possible values are: ; us (default) ; ca (alias for 'us') ; cr (Costa Rica) ; br (Brazil, alias for 'cr') ; uk ; ; This feature can also easily detect false hangups. The symptoms of this is ; being disconnected in the middle of a call for no reason. ; ;callprogress=yes ;progzone=uk ; ; Set the tonezone. Equivalent of the defaultzone settings in ; /etc/dahdi/system.conf. This sets the tone zone by number. ; Note that you'd still need to load tonezones (loadzone in ; /etc/dahdi/system.conf). ; The default is -1: not to set anything. ;tonezone = 0 ; 0 is US ; ; The number of ANI info digits to expect before the main ANI spill. ; Switches using ANI-B, -C, and -D will usually send 1 digit. Modern digital ; systems will send 2, following NANPA ANI II requirements. ; ;ani_info_digits=2 ; ; Time in ms to wait before asterisk sends wink to start ANI spill. Can be ; shortened if your switch supports it. ; ;ani_wink_time=1000 ; ; Time in ms to wait for each digit in the spill including the ST pulse. ; This value can affect how long it takes to recognize ANI failures that do ; not send a ST pulse. If ANI failures take too long to recognize, you can ; lower this value. ; ;ani_timeout=10000 ; ; FXO (FXS signalled) devices must have a timeout to determine if there was a ; hangup before the line was answered. This value can be tweaked to shorten ; how long it takes before DAHDI considers a non-ringing line to have hungup. ; ; ringtimeout will not update on a reload. ; ;ringtimeout=8000 ; ; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF ; Pulse digits from phones (FXS devices, FXO signalling) are always ; detected. ; ;pulsedial=yes ; ; For fax detection, uncomment one of the following lines. The default is *OFF* ; ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ; ; When 'faxdetect' is enabled, one could use 'faxdetect_timeout' to disable fax ; detection after the specified number of seconds into a call. Be aware that ; outgoing analog channels may consider the channel is answered immediately ; when dialing completes. Analog does not have a reliable method of detecting ; when the far end answers. Zero disables the timeout. ; Default is 0 to disable the timeout. ; ;faxdetect_timeout=30 ; ; When 'faxdetect' is used, one could use 'faxbuffers' to configure the DAHDI ; transmit buffer policy. The default is *OFF*. When this configuration ; option is used, the faxbuffer policy will be used for the life of the call ; after a fax tone is detected. The faxbuffer policy is reverted after the ; call is torn down. The sample below will result in 6 buffers and a full ; buffer policy. ; ;faxbuffers=>6,full ; ; When FXO signalling (FXS device, e.g. analog phone) is used, overlap dialing ; is typically used. Asterisk has several configurable (per-channel) timeouts ; to know how long to wait for the next digit. All the values are in ; milliseconds. ; * firstdigit_timeout: a longer timeout before any digit is dialed. ; By default: 16 seconds. ; * interdigit_timeout: timeout for next digits, if the current number dialed ; does not match a number in the current context. Default: 8 seconds. ; * matchdigit_timeout: timeout for next digits, if the current number dialed ; matches a number in the current context. Default: 3 seconds. ; ;firstdigit_timeout=16000 ;interdigit_timeout=8000 ;matchdigit_timeout=3000 ; ; Configure the default number of DAHDI buffers and the transmit policy to use. ; This can be used to eliminate data drops when scheduling jitter prevents ; Asterisk from writing to a DAHDI channel regularly. Most users will probably ; want "faxbuffers" instead of "buffers". ; ; The policies are: ; immediate - DAHDI will immediately start sending the data to the hardware after ; Asterisk writes to the channel. This is the default mode. It ; introduces the least amount of latency but has an increased chance for ; hardware under runs if Asterisk is not able to keep the DAHDI write ; queue from going empty. ; half - DAHDI will wait until half of the configured buffers are full before ; starting to transmit. This adds latency to the audio but reduces ; the chance of under runs. Essentially, this is like an in-kernel jitter ; buffer. ; full - DAHDI will not start transmitting until all buffers are full. ; Introduces the most amount of latency and is susceptible to over ; runs from the Asterisk process. ; ; The receive policy is never changed. DAHDI will always pass up audio as soon ; as possible. ; ; The default number of buffers is 4 (from jitterbuffers) and the default policy ; is immediate. ; ;buffers=4,immediate ; ; This option specifies what to do when the channel's bridged peer puts the ; ISDN channel on hold. Settable per logical ISDN span. ; moh: Generate music-on-hold to the remote party. ; notify: Send hold notification signaling to the remote party. ; For ETSI PTP and ETSI PTMP NT links. ; (The notify setting deprecates the mohinterpret=passthrough setting.) ; hold: Use HOLD/RETRIEVE signaling to release the B channel while on hold. ; For ETSI PTMP TE links. ; ;moh_signaling=moh ; ; This option specifies a preference for which music on hold class this channel ; should listen to when put on hold if the music class has not been set on the ; channel with Set(CHANNEL(musicclass)=whatever) in the dialplan, and the peer ; channel putting this one on hold did not suggest a music class. ; ; This option may be set globally or on a per-channel basis. ; ;mohinterpret=default ; ; This option specifies which music on hold class to suggest to the peer channel ; when this channel places the peer on hold. This option may be set globally, ; or on a per-channel basis. ; ;mohsuggest=default ; ; PRI channels can have an idle extension and a minunused number. So long as ; at least "minunused" channels are idle, chan_dahdi will try to call "idledial" ; on them, and then dump them into the PBX in the "idleext" extension (which ; is of the form exten@context). When channels are needed the "idle" calls ; are disconnected (so long as there are at least "minidle" calls still ; running, of course) to make more channels available. The primary use of ; this is to create a dynamic service, where idle channels are bundled through ; multilink PPP, thus more efficiently utilizing combined voice/data services ; than conventional fixed mappings/muxings. ; ; Those settings cannot be changed on reload. ; ;idledial=6999 ;idleext=6999@dialout ;minunused=2 ;minidle=1 ; ; ; ignore_failed_channels: Continue even if some channels failed to configure. ; True by default. Disable this if you can guarantee that DAHDI starts before ; Asterisk and want to be sure chan_dahdi will not start with broken ; configuration. ; ;ignore_failed_channels = false ; ; Configure jitter buffers in DAHDI (each one is 20ms, default is 4) ; This is set globally, rather than per-channel. ; ;jitterbuffers=4 ; ; ----------------------------- JITTER BUFFER CONFIGURATION -------------------------- ; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; DAHDI channel. Defaults to "no". An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The DAHDI channel can't accept jitter, ; thus an enabled jitterbuffer on the receive DAHDI side will always ; be used if the sending side can create jitter. ; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds. ; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is ; resynchronized. Useful to improve the quality of the voice, with ; big jumps in/broken timestamps, usually sent from exotic devices ; and programs. Defaults to 1000. ; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a DAHDI ; channel. Two implementations are currently available - "fixed" ; (with size always equals to jbmax-size) and "adaptive" (with ; variable size, actually the new jb of IAX2). Defaults to fixed. ; jbtargetextra = 40 ; This option only affects the jb when 'jbimpl = adaptive' is set. ; The option represents the number of milliseconds by which the new ; jitter buffer will pad its size. the default is 40, so without ; modification, the new jitter buffer will set its size to the jitter ; value plus 40 milliseconds. increasing this value may help if your ; network normally has low jitter, but occasionally has spikes. ; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no". ; ---------------------------------------------------------------------------------- ; ; You can define your own custom ring cadences here. You can define up to 8 ; pairs. If the silence is negative, it indicates where the caller ID spill is ; to be placed. Also, if you define any custom cadences, the default cadences ; will be turned off (overwritten). ; ; This setting is global, rather than per-channel. It will not update on ; a reload, but new and modified cadences will update on dahdi restart. ; A maximum of 24 cadences may be specified. ; ; Syntax is: cadence=ring,silence[,ring,silence[...]] ; ; These are the default cadences: ; ;cadence=125,125,2000,-4000 ;cadence=250,250,500,1000,250,250,500,-4000 ;cadence=125,125,125,125,125,-4000 ;cadence=1000,500,2500,-5000 ; ; Each channel consists of the channel number or range. It inherits the ; parameters that were specified above its declaration. ; ; ;callerid="Green Phone"<(256) 428-6121> ;description=Reception Phone ; add a description for 'dahdi show channels' ;channel => 1 ;callerid="Black Phone"<(256) 428-6122> ;description=Courtesy Phone ;channel => 2 ;callerid="CallerID Phone" <(630) 372-1564> ;description= ; reset the description for following channels ;channel => 3 ;callerid="Pac Tel Phone" <(256) 428-6124> ;channel => 4 ;callerid="Uniden Dead" <(256) 428-6125> ;channel => 5 ;callerid="Cortelco 2500" <(256) 428-6126> ;channel => 6 ;callerid="Main TA 750" <(256) 428-6127> ;channel => 44 ; ; For example, maybe we have some other channels which start out in a ; different context and use E & M signalling instead. ; ;context=remote ;signaling=em ;channel => 15 ;channel => 16 ;signalling=em_w ; ; All those in group 0 I'll use for outgoing calls ; ; Strip most significant digit (9) before sending ; ;stripmsd=1 ;callerid=asreceived ;group=0 ;signalling=fxs_ls ;channel => 45 ;signalling=fxo_ls ;group=1 ;callerid="Joe Schmoe" <(256) 428-6131> ;channel => 25 ;callerid="Megan May" <(256) 428-6132> ;channel => 26 ;callerid="Suzy Queue" <(256) 428-6233> ;channel => 27 ;callerid="Larry Moe" <(256) 428-6234> ;channel => 28 ; ; Sample PRI (CPE) config: Specify the switchtype, the signalling as either ; pri_cpe or pri_net for CPE or Network termination, and generally you will ; want to create a single "group" for all channels of the PRI. ; ; switchtype cannot be changed on a reload. ; ; switchtype = national ; signalling = pri_cpe ; group = 2 ; channel => 1-23 ; Used for distinctive ring support for x100p. ; You can see the dringX patterns is to set any one of the dringXcontext fields ; and they will be printed on the console when an inbound call comes in. ; ; dringXrange is used to change the acceptable ranges for "tone offsets". Defaults to 10. ; Note: a range of 0 is NOT what you might expect - it instead forces it to the default. ; A range of -1 will force it to always match. ; Anything lower than -1 would presumably cause it to never match. ; ;dring1=95,0,0 ;dring1context=internal1 ;dring1range=10 ;dring2=325,95,0 ;dring2context=internal2 ;dring2range=10 ; If no pattern is matched here is where we go. ;context=default ;channel => 1 ; AMI alarm event reporting ;reportalarms=channels ;Possible values are: ;channels - report each channel alarms (current behavior, default for backward compatibility) ;spans - report an "SpanAlarm" event when the span of any configured channel is alarmed ;all - report channel and span alarms (aggregated behavior) ;none - do not report any alarms. ; ---------------- Options for use with signalling=ss7 ----------------- ; None of them can be changed by a reload. ; ; Variant of SS7 signalling: ; Options are itu and ansi ;ss7type = itu ; SS7 Called Nature of Address Indicator ; ; unknown: Unknown ; subscriber: Subscriber ; national: National ; international: International ; dynamic: Dynamically selects the appropriate dialplan ; ;ss7_called_nai=dynamic ; ; SS7 Calling Nature of Address Indicator ; ; unknown: Unknown ; subscriber: Subscriber ; national: National ; international: International ; dynamic: Dynamically selects the appropriate dialplan ; ;ss7_calling_nai=dynamic ; ; ; sample 1 for Germany ;ss7_internationalprefix = 00 ;ss7_nationalprefix = 0 ;ss7_subscriberprefix = ;ss7_unknownprefix = ; ; This option is used to disable automatic sending of ACM when the call is started ; in the dialplan. If you do use this option, you will need to use the Proceeding() ; application in the dialplan to send ACM or enable ss7_autoacm below. ;ss7_explicitacm=yes ; Use this option to automatically send ACM when the call rings or is answered and ; has not seen proceeding yet. If you use this option, you should disable ss7_explicitacm. ; You may still use Proceeding() to explicitly send an ACM from the dialplan. ;ss7_autoacm=yes ; Create the linkset with all CICs in hardware remotely blocked state. ;ss7_initialhwblo=yes ; This option is whether or not to trust the remote echo control indication. This means ; that in cases where echo control is reported by the remote end, we will trust them and ; not enable echo cancellation on the call. ;ss7_use_echocontrol=yes ; This option is to set what our echo control indication is to the other end. Set to ; yes to indicate that we are using echo cancellation or no if we are not. ;ss7_default_echocontrol=yes ; All settings apply to linkset 1 ;linkset = 1 ; Set the Signaling Link Code (SLC) for each sigchan. ; If you manually set any you need to manually set all. ; Should be defined before sigchan. ; The default SLC starts with zero and increases for each defined sigchan. ;slc= ; Point code of the linkset. For ITU, this is the decimal number ; format of the point code. For ANSI, this can either be in decimal ; number format or in the xxx-xxx-xxx format ;pointcode = 1 ; Point code of node adjacent to this signalling link (Possibly the STP between you and ; your destination). Point code format follows the same rules as above. ;adjpointcode = 2 ; Default point code that you would like to assign to outgoing messages (in case of ; routing through STPs, or using A links). Point code format follows the same rules ; as above. ;defaultdpc = 3 ; Begin CIC (Circuit indication codes) count with this number ;cicbeginswith = 1 ; What the MTP3 network indicator bits should be set to. Choices are ; national, national_spare, international, international_spare ;networkindicator=international ; First signalling channel ;sigchan = 48 ; Additional signalling channel for this linkset (So you can have a linkset ; with two signalling links in it). It seems like a silly way to do it, but ; for linksets with multiple signalling links, you add an additional sigchan ; line for every additional signalling link on the linkset. ;sigchan = 96 ; Channels to associate with CICs on this linkset ;channel = 25-47 ; ; Set this option if you wish to send an Information Request Message (INR) request ; if no calling party number is specified. This will attempt to tell the other end ; to send it anyways. Should be defined after sigchan. ;inr_if_no_calling=yes ; Set this to set whether or not the originating access is (non) ISDN in the forward and ; backward call indicators. Should be defined after sigchan ;non_isdn_access=yes ; This sets the number of binary places to shift the CIC when doing load balancing between ; sigchans on a linkset. Should be defined after sigchan. Default 0 ;sls_shift = 0 ; Send custom cause_location value ; Should be defined after sigchan. Default 1 (private local) ;cause_location=1 ; SS7 timers (ISUP and MTP3) should be explicitly defined for each linkset to be used. ; For a full list of supported timers and their default values (applicable for both ITU ; and ANSI) see ss7.timers ; Should be defined after sigchan ;#include ss7.timers ; For more information on setting up SS7, see the README file in libss7 or ; https://wiki.asterisk.org/wiki/display/AST/Signaling+System+Number+7 ; ----------------- SS7 Options ---------------------------------------- ; ---------------- Options for use with signalling=mfcr2 -------------- ; MFC-R2 signaling has lots of variants from country to country and even sometimes ; minor variants inside the same country. The only mandatory parameters here are: ; mfcr2_variant, mfcr2_max_ani and mfcr2_max_dnis. ; IT IS RECOMMENDED that you leave the default values (leaving it commented) for the ; other parameters unless you have problems or you have been instructed to change some ; parameter. OpenR2 library uses the mfcr2_variant parameter to try to determine the ; best defaults for your country, also refer to the OpenR2 package directory ; doc/asterisk/ where you can find sample configurations for some countries. If you ; want to contribute your configs for a particular country send them to the e-mail ; of the primary OpenR2 developer that you can find in the AUTHORS file of the OpenR2 package ; MFC/R2 variant. This depends on the OpenR2 supported variants ; A list of values can be found by executing the openr2 command r2test -l ; some valid values are: ; ar (Argentina) ; br (Brazil) ; mx (Mexico) ; ph (Philippines) ; itu (per ITU spec) ; mfcr2_variant=mx ; Max amount of ANI to ask for ; mfcr2_max_ani=10 ; Max amount of DNIS to ask for ; mfcr2_max_dnis=4 ; whether or not to get the ANI before getting DNIS. ; some telcos require ANI first some others do not care ; if this go wrong, change this value ; mfcr2_get_ani_first=no ; Caller Category to send ; national_subscriber ; national_priority_subscriber ; international_subscriber ; international_priority_subscriber ; collect_call ; usually national_subscriber works just fine ; you can change this setting from the dialplan ; by setting the variable MFCR2_CATEGORY ; (remember to set _MFCR2_CATEGORY from originating channels) ; MFCR2_CATEGORY will also be a variable available in your context ; on incoming calls set to the value received from the far end ; mfcr2_category=national_subscriber ; Call logging is stored at the Asterisk ; logging directory specified in asterisk.conf ; plus mfcr2/ ; if you specify 'span1' here and asterisk.conf has ; as logging directory /var/log/asterisk then the full ; path to your MFC/R2 call logs will be /var/log/asterisk/mfcr2/span1 ; (the directory will be automatically created if not present already) ; remember to set mfcr2_call_files=yes ; mfcr2_logdir=span1 ; whether or not to drop call files into mfcr2_logdir ; mfcr2_call_files=yes|no ; MFC/R2 valid logging values are: all,error,warning,debug,notice,cas,mf,stack,nothing ; error,warning,debug and notice are self-descriptive ; 'cas' is for logging ABCD CAS tx and rx ; 'mf' is for logging of the Multi Frequency tones ; 'stack' is for very verbose output of the channel and context call stack, only useful ; if you are debugging a crash or want to learn how the library works. The stack logging ; will be only enabled if the openr2 library was compiled with -DOR2_TRACE_STACKS ; You can mix up values, like: loglevel=error,debug,mf to log just error, debug and ; multi frequency messages ; 'all' is a special value to log all the activity ; 'nothing' is a clean-up value, in case you want to not log any activity for ; a channel or group of channels ; BE AWARE that the level of output logged will ALSO depend on ; the value you have in logger.conf, if you disable output in logger.conf ; then it does not matter you specify 'all' here, nothing will be logged ; so logger.conf has the last word on what is going to be logged ; mfcr2_logging=all ; MFC/R2 value in milliseconds for the MF timeout. Any negative value ; means 'default', smaller values than 500ms are not recommended ; and can cause malfunctioning. If you experience protocol error ; due to MF timeout try incrementing this value in 500ms steps ; mfcr2_mfback_timeout=-1 ; MFC/R2 value in milliseconds for the metering pulse timeout. ; Metering pulses are sent by some telcos for some R2 variants ; during a call presumably for billing purposes to indicate costs, ; however this pulses use the same signal that is used to indicate ; call hangup, therefore a timeout is sometimes required to distinguish ; between a *real* hangup and a billing pulse that should not ; last more than 500ms, If you experience call drops after some ; minutes of being stablished try setting a value of some ms here, ; values greater than 500ms are not recommended. ; BE AWARE that choosing the proper protocol mfcr2_variant parameter ; implicitly sets a good recommended value for this timer, use this ; parameter only when you *really* want to override the default, otherwise ; just comment out this value or put a -1 ; Any negative value means 'default'. ; mfcr2_metering_pulse_timeout=-1 ; Brazil uses a special calling party category for collect calls (llamadas por cobrar) ; instead of using the operator (as in Mexico). The R2 spec in Brazil says a special GB tone ; should be used to reject collect calls. If you want to ALLOW collect calls specify 'yes', ; if you want to BLOCK collect calls then say 'no'. Default is to block collect calls. ; (see also 'mfcr2_double_answer') ; mfcr2_allow_collect_calls=no ; This feature is related but independent of mfcr2_allow_collect_calls ; Some PBX's require a double-answer process to block collect calls, if ; you ever have problems blocking collect calls using Group B signals (mfcr2_allow_collect_calls=no) ; then you may want to try with mfcr2_double_answer=yes, this will cause that every answer signal ; is changed by answer->clear back->answer (sort of a flash) ; (see also 'mfcr2_allow_collect_calls') ; mfcr2_double_answer=no ; This feature allows to skip the use of Group B/II signals and go directly ; to the accepted state for incoming calls ; mfcr2_immediate_accept=no ; You most likely dont need this feature. Default is yes. ; When this is set to yes, all calls that are offered (incoming calls) which ; DNIS is valid (exists in extensions.conf) and pass collect call validation ; will be accepted with a Group B tone (either call with charge or not, depending on mfcr2_charge_calls) ; with this set to 'no' then the call will NOT be accepted on offered, and the call will start its ; execution in extensions.conf without being accepted until the channel is answered (either with Answer() or ; any other application resulting in the channel being answered). ; This can be set to 'no' if your telco or PBX needs the hangup cause to be set accurately ; when this option is set to no you must explicitly accept the call with DAHDIAcceptR2Call ; or implicitly through the Answer() application. ; mfcr2_accept_on_offer=yes ; Skip request of calling party category and ANI ; you need openr2 >= 1.2.0 to use this feature ; mfcr2_skip_category=no ; WARNING: advanced users only! I really mean it ; this parameter is commented by default because ; YOU DON'T NEED IT UNLESS YOU REALLY GROK MFC/R2 ; READ COMMENTS on doc/r2proto.conf in openr2 package ; for more info ; mfcr2_advanced_protocol_file=/path/to/r2proto.conf ; Brazil use a special signal to force the release of the line (hangup) from the ; backward perspective. When mfcr2_forced_release=no, the normal clear back signal ; will be sent on hangup, which is OK for all mfcr2 variants I know of, except for ; Brazilian variant, where the central will leave the line up for several seconds (30, 60) ; which sometimes is not what people really want. When mfcr2_forced_release=yes, a different ; signal will be sent to hangup the call indicating that the line should be released immediately ; mfcr2_forced_release=no ; Whether or not report to the other end 'accept call with charge' ; This setting has no effect with most telecos, usually is safe ; leave the default (yes), but once in a while when interconnecting with ; old PBXs this may be useful. ; Concretely this affects the Group B signal used to accept calls ; The application DAHDIAcceptR2Call can also be used to decide this ; in the dial plan in a per-call basis instead of doing it here for all calls ; mfcr2_charge_calls=yes ; ---------------- END of options to be used with signalling=mfcr2 ; Configuration Sections ; ~~~~~~~~~~~~~~~~~~~~~~ ; You can also configure channels in a separate chan_dahdi.conf section. In ; this case the keyword 'channel' is not used. Instead the keyword ; 'dahdichan' is used (as in users.conf) - configuration is only processed ; in a section where the keyword dahdichan is used. It will only be ; processed in the end of the section. Thus the following section: ; ;[phones] ;echocancel = 64 ;dahdichan = 1-8 ;group = 1 ; ; Is somewhat equivalent to the following snippet in the section ; [channels]: ; ;echocancel = 64 ;group = 1 ;channel => 1-8 ; ; When starting a new section almost all of the configuration values are ; copied from their values at the end of the section [channels] in ; chan_dahdi.conf and [general] in users.conf - one section's configuration ; does not affect another one's. ; ; Instead of letting common configuration values "slide through" you can ; use configuration templates to easily keep the common part in one ; place and override where needed. ; ;[phones](!) ;echocancel = yes ;group = 0,4 ;callgroup = 3 ;pickupgroup = 3 ;threewaycalling = yes ;transfer = yes ;context = phones ;faxdetect = incoming ; ;[phone-1](phones) ;dahdichan = 1 ;callerid = My Name <501> ;mailbox = 501@mailboxes ; ; ;[fax](phones) ;dahdichan = 2 ;faxdetect = no ;context = fax ; ;[phone-3](phones) ;dahdichan = 3 ;pickupgroup = 3,4