los canales y lineas no se cierran inmediatamente.

Discussion in 'Elastix 2.x' started by luiszg, Jun 18, 2009.

  1. luiszg

    Joined:
    Jan 12, 2009
    Messages:
    117
    Likes Received:
    0
    bueno otra cosa curiosa. y es que estoy decidido a entender todo lo que pasa con asterisk.

    Muy amenudo sucede en mi centralida, que cuando voy a marcar un numero externo me dice que todas las lineas se encuentran ocupadas.

    Sin embargo cuando reviso fisicamente no hay nadie hablando por tel, o solo hay una persona.

    Como ya les he comentado antes, mi central se compone de 2 lineas PSTN configuradas en cuatro canales diferentes y dos grupos diferentes.

    tengo 4 canales para poder poner diferentes patrones de marcado:
    por ejemplo marcando con el 2 ( antepuesto al numero externo ) usan la linea 2 y marcando con el 1 ( antepuesto al numero externo) entocnes usen la linea 1.
    Tengo otro canal para que cuando marquen con el nueve ( antepuesto al numero externo ) salga con cualquiera de las dos que este desocupada. es decir ponga ambos grupos ( go y g1 ) en la secuencia. ( ultimo campo caudno configur mi ruta saliente.

    Entendida mi configuración. tengo varias dudas:
    ¿porque cuando hago una llamada externa se usan dos canales?
    ¿porque a veces me sale el mensaje de todas las lineas estan ocupadas y solo hay una ocupada o ninguna.?

    alguna forma de solcuionar esto.

    PD: porque cuando entra una llamada por la PSTN y quien llama cuelga. Mi softh phone no se cuelga de inmediato. En cambio cuando me llaman de un extension SIP en la misma red y quien llama cuelga. mi softh phone si se cuelga.

    :) de antemano muchas gracias!!
     
  2. jcastellanos

    Joined:
    Feb 10, 2009
    Messages:
    2,404
    Likes Received:
    0
    por la linea analoga, no se esta colgando pon esto en tu chan_dahdi.conf :

    busydetect=yes
    busycount=4
    callprogress=no

    con eso basta
     
  3. luiszg

    Joined:
    Jan 12, 2009
    Messages:
    117
    Likes Received:
    0
    Hola.

    No em ha funcionado. Ahora he sumado otra cosa, les cuento mis pruebas:

    Ya he puesto el IVR y tambien en el chan_dahdi.conf lo que jcastellanos me ha dicho. sin embargo si alguien llama, y le comienza a sonar el IVR y cuelga ( sin marcar ninguna opcion) la linea queda ocupada. ( linea telefonica pstn y troncal ).

    de igual manera si yo llamo con la linea telefonica ( pstn ) y cuelgo la troncal queda ocupada por un minuto ( mas o menos ) entonces al intentar usar esa misma troncal me dice que todas las lineas están ocupadas.

    ¿que puede estar pasando?


    AGREGO: acabo de hacer una prueba. llame de mi celular a la linea PSTN y antes de que terminara el IVR colgue. luego volvi a intentar llamar desde mi celular y la linea pstn sonaba ocupada. Luego cogi un telefono fijo ( no es telefono IP ) que esta conectada a la linea pstn y se seguia escuchando el IVR.


    chao!
     
  4. jcastellanos

    Joined:
    Feb 10, 2009
    Messages:
    2,404
    Likes Received:
    0
    ok, pasame tu chan_dahdi.conf
     
  5. luiszg

    Joined:
    Jan 12, 2009
    Messages:
    117
    Likes Received:
    0
    Mi chan_dahdi.conf :

    Code:
    ;
    ; DAHDI telephony interface
    ;
    ; Configuration file
    ;
    ; You need to restart Asterisk to re-configure the DAHDI channels
    ; CLI> reload chan_dahdi.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.  
    ;        trunkgroup => <trunkgroup>,<dchannel>[,<backup1>...]
    ;
    ;        trunkgroup  is the numerical trunk group to create
    ;        dchannel    is the DAHDI channel which will have the 
    ;                    d-channel for the trunk.
    ;        backup1     is an optional list of backup d-channels.
    ;
    ;trunkgroup => 1,24,48
    ;trunkgroup => 1,24
    ;
    ; Spanmap: Associates a span with a trunk group
    ;        spanmap => <dahdispan>,<trunkgroup>[,<logicalspan>]
    ;
    ;        dahdispan   is the DAHDI span number to associate
    ;        trunkgroup  is the trunkgroup (specified above) for the mapping
    ;        logicalspan is the logical span number within the trunk group to use.
    ;                    if unspecified, no logical span number is used.
    ;
    ;spanmap => 1,1,1
    ;spanmap => 2,1,2
    ;spanmap => 3,1,3
    ;spanmap => 4,1,4
    
    [channels]
    ;
    ; Default language
    ;
    ;language=en
    ;
    ; Default context
    ;
    context=default
    ;
    ; Switchtype:  Only used for PRI.
    ;
    ; national:	  National ISDN 2 (default)
    ; dms100:	  Nortel DMS100
    ; 4ess:           AT&T 4ESS
    ; 5ess:	          Lucent 5ESS
    ; euroisdn:       EuroISDN (also known as ETSI NET/5; Cisco calls this "primary-net5")
    ; ni1:            Old National ISDN 1
    ; qsig:           Q.SIG
    ;
    switchtype=national
    ;
    ; Some switches (AT&T especially) require network specific facility IE
    ; supported values are currently 'none', 'sdn', 'megacom', 'tollfreemegacom', 'accunet'
    ;
    ;nsf=none
    ;
    ; PRI Dialplan:  Only RARELY used for PRI.
    ;
    ; unknown:        Unknown
    ; private:        Private ISDN
    ; local:          Local ISDN
    ; national:	  National ISDN
    ; international:  International ISDN
    ; dynamic:	  Dynamically selects the appropriate dialplan
    ;
    ;pridialplan=national
    ;
    ; 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
    ; dynamic:	  Dynamically selects the appropriate dialplan
    ;
    ;prilocaldialplan=national
    ;
    ; 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
    ;
    ; Allow inband audio (progress) when a call is RELEASEd by the far end of a PRI
    ;
    ;inbanddisconnect=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).
    ; K:    Layer 2 max number of outstanding unacknowledged I frames (default 7)
    ; N200: Layer 2 max number of retransmissions of a frame (default 3)
    ; T200: Layer 2 max time before retransmission of a frame (default 1000 ms)
    ; T203: Layer 2 max time without frames being exchanged (default 10000 ms)
    ; T305: Wait for DISCONNECT acknowledge (default 30000 ms)
    ; T308: Wait for RELEASE acknowledge (default 4000 ms)
    ; T309: Maintain active calls on Layer 2 disconnection (default -1, Asterisk clears calls)
    ;       EuroISDN: 6000 to 12000 ms, according to (N200 + 1) x T200 + 2s
    ;       May vary in other ISDN standards (Q.931 1993 : 90000 ms)
    ; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
    ;
    ; 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))
    ; fgccama	  Feature Group C-CAMA (DP DNIS, MF ANI)
    ; fgccamamf	  Feature Group C-CAMA MF (MF DNIS, MF ANI)
    ; fxs_ls:         FXS (Loop Start)
    ; fxs_gs:         FXS (Ground Start)
    ; fxs_ks:         FXS (Kewl Start)
    ; fxo_ls:         FXO (Loop Start)
    ; fxo_gs:         FXO (Ground Start)
    ; fxo_ks:         FXO (Kewl Start)
    ; pri_cpe:        PRI signalling, CPE side
    ; pri_net:        PRI signalling, Network side
    ; 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=fxo_ls
    ;
    ; If you have an outbound signalling format that is different from format
    ; specified above (but compatible), you can specify outbound signalling format,
    ; (see below). The 'signalling' format specified will be the inbound signalling
    ; format. If you only specify 'signalling', then it will be the format for
    ; both inbound and outbound.
    ; 
    ; signalling=featdmf
    ; outsignalling=featb
    ;
    ; 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 milliseconds)
    ;toneduration=100
    ;
    ; Whether or not to do distinctive ring detection on FXO lines
    ;
    ;usedistinctiveringdetection=yes
    ;distinctiveringaftercid=yes	; enable dring detection after callerid for those countries like Australia
    				; where the ring cadence is changed *after* the callerid spill.
    ;
    ; 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
    ;     v23_jp   = v23 as used in Japan
    ;     dtmf     = DTMF as used in Denmark, Sweden and Netherlands
    ;     smdi     = Use SMDI for callerid.  Requires SMDI to be enabled (usesmdi).
    ;
    ;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 internal extensions
    ; With this set to 'yes', busy extensions will hear the call-waiting
    ; tone, and can use hook-flash to switch between callers. The Dial()
    ; app will not return the "BUSY" result for extensions.
    ;
    callwaiting=yes
    ;
    ; 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.
    ; See doc/callingpres.txt
    ;
    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
    ;
    ; For FXS ports (either direct analog or over T1/E1):
    ;   Support flash-hook call transfer (requires three way calling)
    ;   Also enables call parking (overrides the 'canpark' parameter)
    ;
    ; For digital ports using ISDN PRI protocols:
    ;   Support switch-side transfer (called 2BCT, RLT or other names)
    ;   This setting must be enabled on both ports involved, and the
    ;   'facilityenable' setting must also be enabled to allow sending
    ;   the transfer to the ISDN switch, since it sent in a FACILITY
    ;   message.
    ;
    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 received 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@context 
    ;
    ; Enable echo cancellation 
    ; Use either "yes", "no", or a power of two from 32 to 256 if you wish to
    ; actually set the number of taps of cancellation.
    ;
    ; Note that when setting the number of taps, the number 256 does not translate
    ; to 256 ms of echo cancellation.  echocancel=256 means 256 / 8 = 32 ms.
    ;
    ; Note that if any of your DAHDI cards have hardware echo cancellers,
    ; then this setting only turns them on and off; numeric settings will
    ; be treated as "yes". There are no special settings required for
    ; hardware echo cancellers; when present and enabled in their kernel
    ; modules, they take precedence over the software echo canceller compiled
    ; into DAHDI automatically.
    ;
    echocancel=yes
    ;
    ; 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 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)
    ;
    ; WARNING:  In some cases this option can make echo worse!  If you are
    ; trying to debug an echo problem, it is worth checking to see if your echo
    ; is better with the option set to yes or no.  Use whatever setting gives
    ; the best results.
    ;
    ; Note that these parameters do not apply to hardware echo cancellers.
    ;
    ;echotraining=yes
    ;echotraining=800
    ;
    ; If you are having trouble with DTMF detection, you can relax the DTMF
    ; detection parameters.  Relaxing them may make the DTMF detector more likely
    ; to have "talkoff" where DTMF is detected when it shouldn't be.
    ;
    ;relaxdtmf=yes
    ;
    ; 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 dialling *8#.  For simple offices, just
    ; make these both the same.  Groups range from 0 to 63.
    ;
    callgroup=1
    pickupgroup=1
    
    ;
    ; Specify whether the channel should be answered immediately or if the simple
    ; switch should provide dialtone, read digits, etc.
    ; Note: If immediate=yes the dialplan execution will always start at extension
    ; 's' priority 1 regardless of the dialed number!
    ;
    immediate=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
    ;
    ; 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 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 DAHDI 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
    ;
    ; 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.
    ;
    ; If this option is set to "passthrough", then the hold message will always be
    ; passed through as signalling instead of generating hold music locally. This
    ; setting is only valid when used on a channel that uses digital signalling.
    ;
    ; This option may be specified 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. It may be specified globally or on
    ; a per-channel.
    ;
    ;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.
    ;
    ;idledial=6999
    ;idleext=6999@dialout
    ;minunused=2
    ;minidle=1
    ;
    ; Configure jitter buffers in DAHDI (each one is 20ms, default is 4)
    ;
    ;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.
    
    ; 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 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
    ;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 = 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.
    ;
    ;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
    
    busydetect=yes
    busycount=4
    callprogress=no
    
     
  6. luiszg

    Joined:
    Jan 12, 2009
    Messages:
    117
    Likes Received:
    0
    Lo puse en etiqutas code ¿o prefieres que lo adjujnte?

    CHAO y gracias!
     
  7. jcastellanos

    Joined:
    Feb 10, 2009
    Messages:
    2,404
    Likes Received:
    0
    pero que es eso? jajajaj que vercion de elastix tienes?? mi .conf es asi

    [trunkgroups]

    [channels]
    context=from-pstn
    signalling=fxs_ks
    rxwink=300 ; Atlas seems to use long (250ms) winks
    usecallerid=yes
    hidecallerid=no
    callwaiting=yes
    usecallingpres=yes
    callwaitingcallerid=yes
    threewaycalling=yes
    transfer=yes
    canpark=yes
    cancallforward=yes
    callreturn=yes
    echocancel=yes
    echocancelwhenbridged=no
    faxdetect=incoming
    echotraining=800
    rxgain=0.0
    txgain=0.0
    callgroup=1
    pickupgroup=1

    ;Uncomment these lines if you have problems with the disconection of your analog lines

    busydetect=yes
    busycount=4
    callprogress=no



    immediate=no

    #include dahdi-channels.conf
    #include chan_dahdi_additional.conf

    basicamente es lo mismo pero sin tnta basura, te recomiendo que quite todos los comentarios que estan despues e un ";" para mirar que tiene en si el .conf, seguro algo no estamos viendo
     
  8. luiszg

    Joined:
    Jan 12, 2009
    Messages:
    117
    Likes Received:
    0
    mi version es la 1.3-2

    te he pasado el contenido de chan_dahdi.conf.

    lo que he hecho es loguearme en la interfaz web de elastix, entrar a pbx luego a herramientas y luego a editor de archivos. buscar chan_dahdi.conf y abrirlo copiar y pegar el contenido.

    no es asi? :S :S:unsure:
     
  9. luiszg

    Joined:
    Jan 12, 2009
    Messages:
    117
    Likes Received:
    0
    Eso que me has pasado parece el contenido de mi archivo:
    zapata.conf
     
  10. jcastellanos

    Joined:
    Feb 10, 2009
    Messages:
    2,404
    Likes Received:
    0
    si, en efecto, es que omitimos la vercion del elastix, aun usas zaptel por eso la diferencia, pon eso que te pase en ese archivo (zapata.conf) y listo :) (quitalo del otro claro)

    pruebalo y me cuentas ok?

    saludos
     
  11. luiszg

    Joined:
    Jan 12, 2009
    Messages:
    117
    Likes Received:
    0
    Gracias jcastellanos.

    Por cierto que mejoras trae la nueva version ( es la 1.5? ) vale la pena cambiar de version?
     
  12. jcastellanos

    Joined:
    Feb 10, 2009
    Messages:
    2,404
    Likes Received:
    0
    pues la primera y mas importante es el DAHDI, y las actualizaciones que tiene sonmejores, es mas estable y aparte trae el freepbx mas actualizado, ahora, hay que pensar si realmente quieres actualizar, si esta en produccion, mejor instalalo en otro servidor, para que lo pruebes y veas si te combiene o no :)

    funciono lo de la llamada??

    saludos
     
  13. luiszg

    Joined:
    Jan 12, 2009
    Messages:
    117
    Likes Received:
    0
    Hola. No. No em ha funcionado. Si pongo eso en mi zapata.conf. y luego llamo a mi PSTN la llamada no entra al elaastix ( pero si entra a la linea porque escucho el tono en micelular) sin embargo nunca suenan los telefono IP ni los soft phone que deberian de sonar.

    que crees que sea?

    CHAO y gracias!!
     
  14. jcastellanos

    Joined:
    Feb 10, 2009
    Messages:
    2,404
    Likes Received:
    0
    chizpas, si hay una linea que hay que ponerle, pero no recuerdo bien cual, tengo un zapata.conf guardado cuando implemente un asterisk puro, dejame ver que es lo que tiene y te digo.

    creo que es

    answeronpolarityswitch=yes
    hanguponpolarityswitch=yes

    intenta con estos
     
  15. luiszg

    Joined:
    Jan 12, 2009
    Messages:
    117
    Likes Received:
    0
    Hola entonces el archvio quedaría así:


    busydetect=yes
    busycount=4
    callprogress=no
    answeronpolarityswitch=yes
    hanguponpolarityswitch=yes
     
  16. jcastellanos

    Joined:
    Feb 10, 2009
    Messages:
    2,404
    Likes Received:
    0
    no, solo las dos ultimasm asi funciona?
     
  17. luiszg

    Joined:
    Jan 12, 2009
    Messages:
    117
    Likes Received:
    0
    No, de ninguna de las dos maneras funciona. Mi elastix no detecta cuando la linea PSTN es colgada.

    agradecería cualquei ayuda!! ¿alguna idea?

    CHAO!
     
  18. jcastellanos

    Joined:
    Feb 10, 2009
    Messages:
    2,404
    Likes Received:
    0
    pasa tu zapata.conf
     
  19. lemuelgv

    Joined:
    Sep 9, 2008
    Messages:
    73
    Likes Received:
    0
    yo habia solucionado esto activando el bussycount para lineas de telmex


    por otra parte tambien tengo instalados telulares qu etiene la capacidad de hacer polarity reversal

    y estaban funcionando con el comando hanguponpolarityswitch?yes pero hoy al parecer no detecta el colgado

    ya revise mi zapata.conf (uso la version 1.3) y sigue activado el polarityswitch, reinicie el servicio "service zaptel restart" y sigue igual

    tal ves tenga que reiniciar el servidor completo pero como esta en produccion no me es tan sencillo

    algun consejo que me puedan dar

    saludos!!
     
  20. jcastellanos

    Joined:
    Feb 10, 2009
    Messages:
    2,404
    Likes Received:
    0
    quitale esa linea, si antes te funciono y ahora no, quitala y mira que pasa
     

Share This Page