What version of Asterisk? If this is not [EMAIL PROTECTED] you may want to install it and start over. It eases many of the problems experienced by newbs when learning *.
Otherwise, make sure you use the ztcfg -vvvv so you can see some error verbosity. You may need to recompile your zaptel stuff. Just make sure you follow the instructions and recompile asterisk after. Regards, Wliey -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Sent: Thursday, March 17, 2005 10:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Shane Dalgleish Subject: Re: [Asterisk-Users] Newbie can't dial out to pstn I have just run ztcfg and got these errors: # ztcfg Notice: Configuration file is /etc/zaptel.conf line 209: Cannot get number of tones chanel 1 line 209: Cannot init tones chanel 1 line 209: Cannot get number of tones chanel 2 line 209: Cannot init tones chanel 2 line 209: Cannot get number of tones chanel 3 line 209: Cannot init tones chanel 3 line 209: Cannot get number of tones chanel 4 line 209: Cannot init tones chanel 4 What would these mean. I searched the archives and couldn't find these errors. Greg On 18/03/2005, at 1:24 PM, Greg wrote: > I was just copy an example from somewhere. I made the change but the > mobile still doesn't ring. The line the card is attached to works > fine. here is the new output > > Executing Goto("SIP/2002-4385", "mobile|0400039953|1") in new stack > -- Goto (mobile,0400039953,1) > -- Executing Goto("SIP/2002-4385", "localcall|0400039953|1") in > new stack > -- Goto (localcall,0400039953,1) > -- Executing Dial("SIP/2002-4385", "ZAP/1/0400039953|60|r") in new > stack > -- Called 1/0400039953 > -- Zap/1-1 answered SIP/2002-4385 > -- Hungup 'Zap/1-1' > == Spawn extension (localcall, 0400039953, 1) exited non-zero on > 'SIP/2002-4385' > > is this line -- Zap/1-1 answered SIP/2002-4385 displayed when the card > tries to make the call or when the card thinks it has established the > call? > > Regards, > Greg > > By the way, I'm on the Gold Coast. > > On 18/03/2005, at 12:32 PM, Shane Dalgleish wrote: > >> Greg, >> >> Any reason why you are putting the country code on the front for a >> mobile call through pstn? >> (Unless you have something like an Ericsson F220M Fixed Cellular >> Terminal connected to it?) >> >> And you said the tdm400p never tries to pick up the phone? >> Have you connected a normal phone on the line and had a listen? >> >> >> Where is Aus are you? :o) >> >> Cheers >> Shane >> >>> -----Original Message----- >>> From: [EMAIL PROTECTED] >>> [mailto:[EMAIL PROTECTED] On Behalf Of Greg >>> Sent: Friday, 18 March 2005 1:01 PM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: [Asterisk-Users] Newbie can't dial out to pstn >>> >>> Hi, >>> 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 >>> >> >> > > _______________________________________________ > 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 > _______________________________________________ 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 _______________________________________________ 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
