; for any other voicemail context, the following will produce the stutter tone:
; 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.
; 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.
; 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.
; 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.
; 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.
; You may also set the default receive and transmit gains (in dB)
; Gain Settings: increasing / decreasing the volume level on a channel.
; The values are in db (decibells). A positive number
; increases the volume level on a channel, and a
; negavive value decreases volume level.
; Dynamic Range Compression: you can also enable dynamic range compression
; on a channel. This will 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.
; There are several independent gain settings:
; rxgain: gain for the rx (receive - into Asterisk) channel. Default: 0.0
; txgain: gain for the tx (transmit - out of Asterisk Asterisk) channel.
; Default: 0.0
; cid_rxgain: set the gain just for the caller ID sounds Asterisk
; emits. Default: 5.0 .
; rxdrc: dynamic range compression for the rx channel. Default: 0.0
; txdrc: dynamic range compression for the tx channel. Default: 0.0
; 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.
; 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.
; 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.
; Channel variable to be set for all calls from this channel
;setvar=ATTENDED_TRANSFER_COMPLETE_SOUND=beep ; This channel variable will
; cause the given audio file to
; be played upon completion of
; an attended 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!
; Specify whether flash-hook transfers to ‘busy’ channels should complete or
; return to the caller performing the transfer (default is yes).
; Calls will have the party id user tag set to this string value.
; 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
; 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’.
; Channels may be associated with an account code to ease
; ADSI (Analog Display Services Interface) can be enabled on a per-channel
; basis if you have (or may have) ADSI compatible CPE equipment
; 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.
; 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.
; 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.
; 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.
; 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.
; 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.
; 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 buzydetect (unless
; busypattern is set explicitly). The possible values are:
; us (default)
; ca (alias for ‘us’)
; cr (Costa Rica)
; br (Brazil, alias for ‘cr’)
; This feature can also easily detect false hangups. The symptoms of this is
; being disconnected in the middle of a call for no reason.
; 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
; The default is -1: not to set anything.
;tonezone = 0 ; 0 is US
; 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.
; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
; Pulse digits from phones (FXS devices, FXO signalling) are always
; For fax detection, uncomment one of the following lines. The default is OFF
; 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.
; 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
; 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.
; 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.
; 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.
; 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.
; 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.
; ignore_failed_channels: Continue even if some channels failed to configure.
; False by default, as if even a single channel failed to configure, it might
; mean other channels are misplaced and having them work may not be a good
; idea. If enabled (set to true), chan_dahdi will nevertheless attempt to
; configure other channels rather than giving up. This normally makes sense
; only if you use names (!) for DAHDI channels.
;ignore_failed_channels = true
; Configure jitter buffers in DAHDI (each one is 20ms, default is 4)
; This is set globally, rather than per-channel.
;------------------------------ 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.
; This setting is global, rather than per-channel. It will not update on
; a reload.
; Syntax is: cadence=ring,silence[,ring,silence[…]]
; These are the default cadences:
; 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>
;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.
;channel => 15
;channel => 16
; All those in group 0 I’ll use for outgoing calls
; Strip most significant digit (9) before sending
;channel => 45
;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
; Alternatively, the number of the channel may be replaced with a relative
; path to a device file under /dev/dahdi . The final element of that file
; must be a number, though. The directory separator is ‘!’, as we can’t
; use ‘/’ in a dial string. So if we have
; we could use:
;channel => span-name!pstn!00!1-4
;channel => span-name!pstn!00!1,2,3,4
; See also ignore_failed_channels above.
; 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.
; If no pattern is matched here is where we go.
;channel => 1
; AMI alarm event reporting
;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 Calling Nature of Address Indicator
; unknown: Unknown
; subscriber: Subscriber
; national: National
; international: International
; dynamic: Dynamically selects the appropriate dialplan
; sample 1 for Germany
;ss7_internationalprefix = 00
;ss7_nationalprefix = 0
; 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.
; All settings apply to linkset 1
;linkset = 1
; 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
; 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
; For more information on setting up SS7, see the README file in libss7 or
; wiki.asterisk.org/wiki/display/ … m+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)
; Max amount of ANI to ask for
; Max amount of DNIS to ask for
; 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
; Caller Category to send
; 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
; 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
; whether or not to drop call files into mfcr2_logdir
; 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
; 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
; 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’.
; 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’)
; 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’)
; This feature allows to skip the use of Group B/II signals and go directly
; to the accepted state for incoming calls
; 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.
; Skip request of calling party category and ANI
; you need openr2 >= 1.2.0 to use this feature
; 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
; 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
; 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
; ---------------- 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:
;echocancel = 64
;dahdichan = 1-8
;group = 1
; Is somewhat equivalent to the following snippet in the section
;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.
;echocancel = yes
;group = 0,4
;callgroup = 3
;pickupgroup = 3
;threewaycalling = yes
;transfer = yes
;context = phones
;faxdetect = incoming
;dahdichan = 1
;callerid = My Name <501>
;mailbox = 501@mailboxes
;dahdichan = 2
;faxdetect = no
;context = fax
;dahdichan = 3
;pickupgroup = 3,4