chan_dahdi.conf.sample

;
; 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 <busycount> 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 buzydetect (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.
;
; 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:
;
;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/<whatever you put here>
; 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