[asterisk-users] GSM gateway and FXO ATA

[Paul Hales]

What you need is something like:

exten => _456.,1,Dial(SIP/${EXTEN}@IP.OF.2N.UNIT,30,tTD(${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: tomo at pbx.lama.hr
> e-mail: tparcina#lama.hr
> http://www.lama.hr
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