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