[AsteriskBrasil] ISDN-PRI

Josué Conti josueconti em gmail.com
Sexta Junho 9 20:44:21 BRT 2006


Fernando, boa noite.
Acredito até que possa ser problemas em sua configuração, uma dica.
Delete esse seu zapata.conf e configure um limpo, com apenas aquilo que
realmente você for utilizar, acho que será melhor para você administrar.
Poste na lista novamente, sem o debug, acho que dá para vermos alguma coisa.

Abraço e boa sorte

Josué


2006/6/9, Fernando Lujan <fernando.lujan em mandic.com.br>:
>
> 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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