What you need is something like: exten => _456.,1,Dial(SIP/[EMAIL PROTECTED],30,tTD(${EXTEN:3}))
regards, PaulH AsteriskIT www.asteriskit.com.au On Tue, 2006-08-22 at 10:59 +0200, Tomislav Parčina wrote: > Hi list! > > I'm trying to connect Analog GSM gateway (2N Ateus) with Asterisk 1.2.5 over > Grandstream HT488 ATA. I have plugged gateway to FXO port of ATA. Incoming > calls work the way I call GSM number and then I get DISA to call inside > company. Outgoing call work well when I call VoIP number of ATA which calls > gateway and then I dial number I wish to call over gateway. As I said, it > works fine. > > Now I would like to dial ATA_number+number_I_wish_to_call so that I don't > have to dial twice when I'm trying to establish outgoing call from company > thru gateway. > > I have tried this but it doesn't work well. > > ; to dial outside thru GSM gateway > exten => _456.,1,Dial(SIP/${EXTEN},30,tTD(${EXTEN:3})) > exten => _456.,n,Hangup > > This is what I see on CLI: > > -- Executing Dial("SIP/577-104c", "SIP/4560989970434|30|tTD(248)") in new > stack > == Everyone is busy/congested at this time (1:0/0/1) > -- Executing Hangup("SIP/577-104c", "") in new stack > == Spawn extension (sip, 4560989970434, 2) exited non-zero on 'SIP/577-104c' > > Why asterisk thinks that gateway is busy when it's not? > > > > -- > Tomislav Parčina > Lama Computers Split > Stinice 12, 21000 Split > Tel.: +385(21)495148 > Mob.: +385(91)1212148 > SIP: [EMAIL PROTECTED] > e-mail: tparcina#lama.hr > http://www.lama.hr > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users