Hi,

I'm trying to set up a basic FXO <> SIP gateway. That is, I want calls from my SIP phone to simply be dumped onto the POTS line. My (entire) extensions.conf is:

        [from-sip]
        exten => _9NXXXXXX,1,Dial(ZAP/1/${EXTEN})

and my zaptel.conf is:

        fxsks=1
        loadzone=us
        defaultzone=us

and my zapata.conf is:

        context=incoming
        signalling=fxs_ks
        echocancel=yes
        echocancelwhenbridged=yes
        relaxdtmf=yes
        rxgain=1.5
        txgain=1.5
        immediate=no
        busydetect=no
        callprogress=no
        musiconhold=default
        usecallerid=yes
        callerid=asreceived
        channel => 1

I am using an SNOM200 SIP phone and a TDM01B (1-port FXO bundle).

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

*CLI> -- Executing Dial("SIP/555-83ee", "ZAP/1/92262802") in new stack
Jul 23 13:50:24 WARNING[-267056208]: channel.c:1860 ast_request: No channel type registered for 'ZAP'
Jul 23 13:50:24 NOTICE[-267056208]: app_dial.c:696 dial_exec: Unable to create channel of type 'ZAP'
== Everyone is busy/congested at this time



Here's my channel map:

        [EMAIL PROTECTED] asterisk]# /sbin/ztcfg -vv

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


Channel map:

        Channel 01: FXS Kewlstart (Default) (Slaves: 01)

        1 channels configured.


What have I done wrong?


- Mike _______________________________________________ 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

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