[AsteriskBrasil] ISDN-PRI
Fernando Lujan
fernando.lujan em mandic.com.br
Sexta Junho 9 15:48:31 BRT 2006
Saudações,
Estava tentando configurar o Asterisk com o MFC/R2 e não tive sucesso.
Pedi para a operadora trocar a sinalização para ISDN.
Removi tudo referente a asterisk e reinstalei novamente.
O servidor que estou me ligando através de um T1 crossover está
configurado como slave.
Quando tento fazer uma ligação para o servidor no qual estou ligando o
asterisk recebo esta informação. (ha*CLI>
Creio que seja um problema de sinalização/protocolo.
Alguma idéia do que possa ser? Obrigado.
-- Accepting UNAUTHENTICATED call from 192.168.1.59:
> requested format = gsm,
> requested prefs = (),
> actual format = gsm,
> host prefs = (),
> priority = mine
-- Executing Answer("IAX2/123456-2", "") in new stack
-- Executing Dial("IAX2/123456-2", "Zap/g1/1144882120") in new stack
-- Making new call for cr 32770
-- Requested transfer capability: 0x00 - SPEECH
> Protocol Discriminator: Q.931 (8) len=49
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a3]
> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
> Ext: 1 Trans mode/rate: 64kbps,
circuit-mode (16)
> Ext: 1 User information layer 1: A-Law (35)
> [18 03 a1 83 81]
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Preferred Dchan: 0
> ChanSel: Reserved
> Ext: 1 Coding: 0 Number Specified Channel
Type: 3
> Ext: 1 Channel: 1 ]
> [28 08 4d 6f 7a 50 68 6f 6e 65]
> Display (len= 8) [ MozPhone ]
> [6c 08 00 80 39 39 39 39 39 39]
> Calling Number (len=10) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
> Presentation: Presentation permitted, user
number not screened (0) '999999' ]
> [70 0b 80 31 31 34 34 38 38 32 31 32 30]
> Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0) '1144882120' ]
> [a1]
> Sending Complete (len= 1)
-- Called g1/1144882120
== Primary D-Channel on span 1 down
Jun 9 14:41:22 WARNING[6430]: chan_zap.c:2290 pri_find_dchan: No
D-channels available! Using Primary channel 16 as D-channel anyway!
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated,
peerstate Overlap sending
> Protocol Discriminator: Q.931 (8) len=9
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: DISCONNECT (69)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the local user (1)
> Ext: 1 Cause: Unknown (16), class = Normal Event (1) ]
NEW_HANGUP DEBUG: Destroying the call, ourstate Disconnect Request,
peerstate Disconnect Indication
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'IAX2/123456-2' status is 'CHANUNAVAIL'
== Primary D-Channel on span 1 down
Jun 9 14:41:25 WARNING[6430]: chan_zap.c:2290 pri_find_dchan: No
D-channels available! Using Primary channel 16 as D-channel anyway!
> cdr_odbc: Query Successful!
-- Hungup 'IAX2/123456-2'
bilitei o debug do span 1)
*CLI> -- Accepting UNAUTHENTICATED call from 192.168.1.59:
> requested format = gsm,
> requested prefs = (),
> actual format = gsm,
> host prefs = (),
> priority = mine
-- Executing Answer("IAX2/123456-2", "") in new stack
-- Executing Dial("IAX2/123456-2", "Zap/g1/1144882120") in new stack
-- Making new call for cr 32770
-- Requested transfer capability: 0x00 - SPEECH
> Protocol Discriminator: Q.931 (8) len=49
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a3]
> Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer
capability: Speech (0)
> Ext: 1 Trans mode/rate: 64kbps,
circuit-mode (16)
> Ext: 1 User information layer 1: A-Law (35)
> [18 03 a1 83 81]
> Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0,
Preferred Dchan: 0
> ChanSel: Reserved
> Ext: 1 Coding: 0 Number Specified Channel
Type: 3
> Ext: 1 Channel: 1 ]
> [28 08 4d 6f 7a 50 68 6f 6e 65]
> Display (len= 8) [ MozPhone ]
> [6c 08 00 80 39 39 39 39 39 39]
> Calling Number (len=10) [ Ext: 0 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0)
> Presentation: Presentation permitted, user
number not screened (0) '999999' ]
> [70 0b 80 31 31 34 34 38 38 32 31 32 30]
> Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI:
Unknown Number Plan (0) '1144882120' ]
> [a1]
> Sending Complete (len= 1)
-- Called g1/1144882120
== Primary D-Channel on span 1 down
Jun 9 14:41:22 WARNING[6430]: chan_zap.c:2290 pri_find_dchan: No
D-channels available! Using Primary channel 16 as D-channel anyway!
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated,
peerstate Overlap sending
> Protocol Discriminator: Q.931 (8) len=9
> Call Ref: len= 2 (reference 2/0x2) (Originator)
> Message type: DISCONNECT (69)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the local user (1)
> Ext: 1 Cause: Unknown (16), class = Normal Event (1) ]
NEW_HANGUP DEBUG: Destroying the call, ourstate Disconnect Request,
peerstate Disconnect Indication
-- Hungup 'Zap/1-1'
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'IAX2/123456-2' status is 'CHANUNAVAIL'
== Primary D-Channel on span 1 down
Jun 9 14:41:25 WARNING[6430]: chan_zap.c:2290 pri_find_dchan: No
D-channels available! Using Primary channel 16 as D-channel anyway!
> cdr_odbc: Query Successful!
-- Hungup 'IAX2/123456-2'
zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone=us
defaultzone=us
zapata.conf
;
; Zapata telephony interface
;
; Configuration file
;
; You need to restart Asterisk to re-configure the Zap channel
; CLI> reload chan_zap.so
; will reload the configuration file,
; but not all configuration options are
; re-configured during a reload.
[trunkgroups]
;
; Trunk groups are used for NFAS or GR-303 connections.
;
; Group: Defines a trunk group.
; group => <trunkgroup>,<dchannel>[,<backup1>...]
;
; trunkgroup is the numerical trunk group to create
; dchannel is the zap 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 => <zapspan>,<trunkgroup>[,<logicalspan>]
;
; zapspan is the zap 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=us
;
; Default context
;
context=incoming
;
; Switchtype: Only used for PRI.
;
; national: National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess: AT&T 4ESS
; 5ess: Lucent 5ESS
; euroisdn: EuroISDN
; ni1: Old National ISDN 1
; qsig: Q.SIG
;
switchtype=euroisdn
;
; Some switches (AT&T especially) require network specific facility IE
; supported values are currently 'none', 'sdn', 'megacom', 'accunet'
;
;nsf=none
;
; PRI Dialplan: Only RARELY used for PRI.
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
;
pridialplan=unknown
;
; PRI Local Dialplan: Only RARELY used for PRI (sets the calling
number's numbering plan)
;
; unknown: Unknown
; private: Private ISDN
; local: Local ISDN
; national: National ISDN
; international: International ISDN
;
prilocaldialplan=unknown
;
; PRI callerid prefixes based on the given TON/NPI (dialplan)
; This is especially needed for euroisdn E1-PRIs
;
; 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
; channels, defaults to 3600; minimum 60 seconds. Some PBXs don't like
; channel restarts. so set the interval to a very long interval e.g.
100000000
; or 'never' to disable *entirely*.
;
;resetinterval = 3600
;
; Overlap dialing mode (sending overlap digits)
;
;overlapdial=yes
;
; 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
;
; 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 = 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).
;
; pritimer => t200,1000
; pritimer => t313,4000
;
; To enable transmission of facility-based ISDN supplementary services (such
; as caller name from CPE over facility), enable this option.
; facilityenable = yes
;
;
; Signalling method (default is fxs). Valid values:
; em: E & M
; 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))
; 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
; gr303fxoks_net: GR-303 Signalling, FXO Loopstart, Network side
; gr303fxsks_cpe: GR-303 Signalling, FXS Loopstart, CPE 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
;
; 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)
;
signalling=pri_net
;
; For Feature Group D Tandem access, to set the default CIC and OZZ use
these
; parameters:
;defaultozz=0000
;defaultcic=303
;
; A variety of timing parameters can be specified as well
; Including:
; 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)
;
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; How long generated tones (DTMF and MF) will be played on the channel
; (in miliseconds)
;toneduration=100
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes
;
; Whether or not to use caller ID
;
usecallerid=yes
;
; Type of caller ID signalling in use
; bell = bell202 as used in US
; v23 = v23 as used in the UK
; dtmf = DTMF as used in Denmark, Sweden and Netherlands
;
;cidsignalling=bell
;
; What signals the start of caller ID
; ring = a ring signals the start
; polarity = polarity reversal signals the start
;
;cidstart=ring
;
; Whether or not to hide outgoing caller ID (Override with *67 or *82)
;
hidecallerid=no
;
; Whether or not to enable call waiting on FXO lines
;
callwaiting=yes
;
; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
; available for the user)
; Mostly use with FXS ports
;
;restrictcid=no
;
; Whether or not use the caller ID presentation for the outgoing call
that the
; calling switch is sending.
;
usecallingpres=yes
;
; Some countries (UK) have ring tones with different ring tones (ring-ring),
; which means the callerid needs to be set later on, and not just after
; the first ring, as per the default.
;
;sendcalleridafter=1
;
;
; Support Caller*ID on Call Waiting
;
callwaitingcallerid=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; Support flash-hook call transfer (requires three way calling)
; Also enables call parking (overrides the 'canpark' parameter)
;
transfer=yes
;
; Allow call parking
; ('canpark=no' is overridden by 'transfer=yes')
;
canpark=yes
;
; Support call forward variable
;
cancallforward=yes
;
; Whether or not to support Call Return (*69)
;
callreturn=yes
;
; Stutter dialtone support: If a mailbox is specified without a voicemail
; context, then when voicemail is received in a mailbox in the default
; voicemail context in voicemail.conf, taking the phone off hook will
cause a
; stutter dialtone instead of a normal one.
;
; If a mailbox is specified *with* a voicemail context, the same will result
; if voicemail recieved in mailbox in the specified voicemail context.
;
; for default voicemail context, the example below is fine:
;
;mailbox=1234
;
; for any other voicemail context, the following will produce the
stutter tone:
;
;mailbox=1234 em 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.
;
echocancel=yes
;
; Generally, it is not necessary (and in fact undesirable) to echo
cancel when
; the circuit path is entirely TDM. You may, however, reverse this behavior
; by enabling the echo cancel 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
; asterisk 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)
;
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
;
; You may also set the default receive and transmit gains (in dB)
;
rxgain=0.0
txgain=0.0
;
; Logical groups can be assigned to allow outgoing rollover. Groups range
; from 0 to 63, and multiple groups can be specified.
;
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
;
callgroup=1
pickupgroup=1
;
; Specify whether the channel should be answered immediately or if the
simple
; switch should provide dialtone, read digits, etc.
;
immediate=no
;
; Specify whether flash-hook transfers to 'busy' channels should complete or
; return to the caller performing the transfer (default is yes).
;
;transfertobusy=no
;
; CallerID can be set to "asreceived" or a specific number if you want to
; override it. Note that "asreceived" only applies to trunk interfaces.
;
;callerid=2564286000
;
; 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
;
; 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 4, but better results
can be
; achieved if 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=4
;
; 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 the Asterisk Makefile 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 defining the
; -DBUSYDETECT_COMPARE_TONE_AND_SILENCE option.
;
; Use a polarity reversal to mark when a outgoing call is answered by the
; remote party.
;
;answeronpolarityswitch=yes
;
; 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
;
; 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"
;
; 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=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 Zap considers a non-ringing line to have hungup.
;
;ringtimeout=8000
;
; For FXO (FXS signalled) devices, whether to use pulse dial instead of DTMF
;
;pulsedial=yes
;
; For fax detection, uncomment one of the following lines. The default
is *OFF*
;
;faxdetect=both
;faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;
; Select which class of music to use for music on hold. If not specified
; then the default will be used.
;
musiconhold=default
;
; PRI channels can have an idle extension and a minunused number. So
long as
; at least "minunused" channels are idle, chan_zap will try to call
"idledial"
; on them, and then dump them into the PBX in the "idleext" extension (which
; is of the form exten em 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.
;
;idledial=6999
;idleext=6999 em dialout
;minunused=2
;minidle=1
;
; Configure jitter buffers in zapata (each one is 20ms, default is 4)
;
;jitterbuffers=4
;
; 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 callerid
spill is
; to be placed. Also, if you define any custom cadences, the default
cadences
; will be turned off.
;
; 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.
;
; For GR-303, CRV's are created like channels except they must start
with the
; trunk group followed by a colon, e.g.:
;
; crv => 1:1
; crv => 2:1-2,5-8
;
;
;callerid="Green Phone"<(256) 428-6121>
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
;channel => 2
;callerid="CallerID Phone" <(256) 428-6123>
;callerid="CallerID Phone" <(630) 372-1564>
;callerid="CallerID Phone" <(256) 704-4666>
;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
;sigalling=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 = national
; signalling = pri_cpe
; group = 2
; channel => 1-23
;
; Used for distintive 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.
;
;dring1=95,0,0
;dring1context=internal1
;dring2=325,95,0
;dring2context=internal2
; If no pattern is matched here is where we go.
;context=default
;channel => 1
channel => 1-15,17-31
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