Hi,

I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box
(customized kernel version 2.4.24). I want calls from my SIP soft-phones
to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap
HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc.
I've read everything I've found at www.voip-info.org, then I've downloaded
the latest bri-stuff.0.1.0-RC2g (released just today!) and started the installation.

My extensions.conf is:

        [default]
        ...cut...
        ignorepat => 9
        exten => 9,1,Dial(Zap/g1/)      ; direct outbound dialing
        exten => 9,2,Congestion

Here's my zaptel.conf:

        loadzone=it
        defaultzone=it

        span=1,1,3,ccs,ami
        bchan=1-2
        dchan=3

Here's my zapata.conf:

        [channels]
        ;
        ; Default language
        ;
        ;language=en
        ;
        ; Default context
        ;
        ;
        switchtype = euroisdn
        ; p2mp TE mode
        signalling = bri_cpe_ptmp

        pridialplan=local
        prilocaldialplan=local
        echocancel=yes
        immediate=yes
        group = 1
        context=default
        channel => 1-2

Here's my channels map:

        Zaptel Configuration
        ======================

        SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)

        Channel map:

        Channel 01: Individual Clear channel (Default) (Slaves: 01)
        Channel 02: Individual Clear channel (Default) (Slaves: 02)
        Channel 03: D-channel (Default) (Slaves: 03)

        3 channels configured.

Finally, here's my sip.conf:

        [general]
        context=default                 ; Default context for incoming calls
        ...cut...
        disallow=all                    ; First disallow all codecs
        allow=gsm                       ; Allow codecs in order of preference
        allow=ulaw
        allow=alaw

        ...cut...

        [alessandro]
        type=friend
        username=alessandro
        secret=bissoli
        host=dynamic
        dtmfmode=rfc2833
        disallow=all
        allow=gsm
        allow=alaw
        allow=ulaw



When I run asterisk and dial from the SIP phone I get this error:

    -- Executing Dial("SIP/alessandro-0f69", "Zap/g1/") in new stack Jul
26 14:27:31 NOTICE[311313]: app_dial.c:711 dial_exec: Unable to create channel
of type 'Zap'
  == Everyone is busy/congested at this time
    -- Executing Congestion("SIP/alessandro-0f69", "") in new stack
  == Spawn extension (default, 9, 2) exited non-zero on 'SIP/alessandro-0f69'

I really don't know what is wrong! Do you have any hints, please?

Alex

__________________________________________________________________
Tiscali ADSL Senza Canone, paga solo quello che consumi!
Non perdere la promozione valida fino al 27 luglio. Per te gratis il modem
in comodato e l'attivazione. In piu' navighi a soli 1,5 euro l'ora per i
primi tre mesi. Cosa aspetti? Attivala subito!
http://abbonati.tiscali.it/adsl/prodotti/640Kbps/



_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to