[Asterisk-Users] Re: How can i call to a cellphone here in Mexico?

Alchaemist info at lnet.com.ar
Wed Sep 21 15:05:59 MST 2005


Hi again...

        How are you dialling this?
90446612345678 ? or 0446612345678 ?

        Also another possibility is that the card is sending the DTMF when it haven't yet get the tone from your PSTN? just thinking... in that case you can use the 'w' in the dialstring to get a wait delay of 0.5 secs. (I don't think it might be this one because your problem is speciffic to cells, but just thinking aloud)

        Good luck!
Alchaemist
  "Claudio Canseco" <claudio.canseco at gmail.com> wrote in message news:8c1b1bde05092113458b6621 at mail.gmail.com...
  Hi, thanks for your replay Alex:


  Right now a have an Asterisk server on a Dell Optiplex GX110 (PIII 666MHz, 320 RAM) with no soundcard.
  With an X100P clone card (an ambient modem).

  Everything looks good, I've been able to make local calls trough PSTN, IAX, SIP.
  I only have 1 POTS line, and 4 SIP softphones (X-lite) running all right.
  The only problem so far I have noticed (or realized of  :P), it is that i can make calls
  to cellularphone numbers, * tries to connect but i get redirected to the emergency service number 066.

  I don't think it is because of my dialplan, eventhough I tried several configurations. Anyways here is part of the dialplan
  where my softphones make calls:


    ;########################################
    ;#  Llamadas salientes  [outgoing]      #
    ;########################################

    [outgoing]
    include => toPSTN
    include => iaxtel
    include => fwd-iax

    ;########   -> PSTN ########

    [toPSTN]                         ; Permite hacer llamadas locales (7-digitos sin contar 9)
    ignorepat => 9

    exten => _92XXXXXX,1,NoOp("Call for "${EXTEN:1})
    exten => _92XXXXXX,2,Dial(Zap/1/${EXTEN:1})

    exten => _904466XXXXXXXX,1,NoOp("Call for "(${EXTEN:1})  ;Llamadas a Celular
    exten => _904466XXXXXXXX,2,Dial(Zap/1/ww${EXTEN:1})

    ;######## -> IAXTEL ########

    [iaxtel]
    exten => _1700XXXXXXX,1,Dial(IAX2/${cuenta}@iaxtel.com/${EXTEN}@iaxtel )
    exten => _1888NXXXXXX,1,Dial( IAX2/${cuenta}@iaxtel.com/${EXTEN}@iaxtel )
    exten => _1877NXXXXXX,1,Dial(IAX2/${cuenta}@iaxtel.com/${EXTEN}@iaxtel )
    exten => _1866NXXXXXX,1,Dial( IAX2/${cuenta}@iaxtel.com/${EXTEN}@iaxtel )
    exten => _1800NXXXXXX,1,Dial(IAX2/${cuenta}@iaxtel.com/${EXTEN}@iaxtel )

    ;########   -> FWD  ########

    [fwd-iax]
    exten => _3.,1,SetCallerId,${FWDCIDNAME}
    exten => _3.,2,Dial( IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:1},60,tr) 
    exten => _3.,3,Congestion



    ;#########################################
    ;#  Softphone x-lite                     #
    ;#########################################

    [x-lite]  ; Note: SIP extensions are defined here as "66" followed by any two digits
    include => default
    include => servicios
    include => outgoing

    exten => 6600,1,NoOp(Llamada saliente maneja IAX2)
    exten => 6600,2,Macro(dial,kano00,IAX2/kano00,20,tr)

    exten => _X,1,NoOp(Llamada saliente maneja SIP)
    exten => _X,2,Macro(dial,667${EXTEN},SIP/667${EXTEN},20,tr)


  All softphones working are SIP, and are directed to the [x-lite] context.

  This is my zapata.conf:

    [channels]
    language=es
    context=incoming
    signalling=fxs_ks.
    usecallerid=yes
    hidecallerid=no
    callwaiting=yes
    callwaitingcallerid=yes
    threewaycalling=yes
    transfer=yes
    cancallforward=yes
    callreturn=yes 

    echocancel=yes
    echocancelwhenbridged=yes
    echotraining=800
    rxgain=0.0
    txgain=25.0

    group=1
    pickupgroup=1
    immediate=yes
    musiconhold=default
    relaxdtmf=yes   ; Relajar el DTMF, poner si asterisk salta o duplica algún DTMF, 
      ; dando lugar a un número incorrecto.
    channel => 1

  And my simple configuration file, 
  zaptel.conf:
    loadzone=mx
    defaultzone=mx
    fxsks=1

  As you can see this aren't complicated configurations because i only have 1 X100P card, and I am currently using little extensions.

  Also, I am not using AMP but I'm thinking to installing it over my current installation. I installed asterisk and zaptel from instructions i got from several documentations sites (voip-info wiki, digium, etc). 

  Well, I hope this info can help to look down the problem. Thanks again,

  Regards,

  Claudio



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