I have just put in a tdm400p with 4 fxo modules and am trying to dial out from x-lite to dial my mobile phone just to test.
The output in the asterisk console is like this
Executing Goto("SIP/2002-239b", "mobile|61400039953|1") in new stack
-- Goto (mobile,61400039953,1)
-- Executing Goto("SIP/2002-239b", "localcall|61400039953|1") in new stack
-- Goto (localcall,61400039953,1)
-- Executing Dial("SIP/2002-239b", "ZAP/1/61400039953|60|r") in new stack
-- Called 1/61400039953
-- Zap/1-1 answered SIP/2002-239b
-- Hungup 'Zap/1-1'
== Spawn extension (localcall, 61400039953, 1) exited non-zero on 'SIP/2002-239b'
It never tries to pick up the phone and dial out. I'm not sure if the config is correct, but I can easily dial between x-lite clients, just not get the pstn.
Can anyone see any glaring mistakes?
Any help is grealty appreciated.
Regards, Greg
My extensions.conf part is this:
exten => _04XXXXXXXX,1,GoTo(mobile,61${EXTEN:1},1)[localcall] ; local calls by PSTN ?is a fixed charge, voip is per minute
exten => _X.,1,Dial(ZAP/1/${EXTEN},60,r)
exten => _X.,2,Congestion
exten => _X.,3,Hangup
exten => _X.,103,Hangup
exten => _X.,104,Hangup
exten => _X.,105,Hangup[mobile] ; Maybe be cheaper to route mobile calls differently to STD in some cases
exten => _X.,1,Goto(localcall,${EXTEN},1)
zaptel.conf fxsks=1-4 loadzone=au defaultzone=au channels=1-4
zapata.conf [channels] � busydetect=1 busycount=7 � relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes � usecallerid=yes � echocancel=yes echocancelwhenbridged=yes � rxgain=0.0 txgain=0.0 � group=1 pickupgroup=1-4 � immediate=no � context=incomingcall � signalling=fxs_ks callerid=asreceived channel=1-4
_______________________________________________ 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
