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