[Asterisk-Users] Sipura 3000 Inbound Dialing Problem

[Joseph]

On PSTN-Line tab

Subscriber Information
User ID: 99
Password: 99

Dial Plans
Dial Plan 1: S0<:99>

PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: Yes
PSTN Ring Thru Line 1: Yes
PSTN Caller Default DP: 1

That should be it I think.

-- 
#Joseph


On Tue, 2005-03-01 at 04:34 -0800, dhananjay sarnaik wrote:
> Dear All
> 
>  
> 
> Im facing wearied problem with Sipura 3000 and asterisk .
> 
>  
> 
> Im trying to configure Asterisk with Sipura 3000 . I have configured
> asterisk with FSX port  which is working fine.
> 
> I want to configure Asterisk FXO port for my outgoing and incoming
> calls.
> 
> Once Sipura received call from outside it will deliver to Asterisk and
> asterisk will play IVR user dial any extension
> 
> Here is my configuration
> 
>  
> 
> sip.conf
> 
>  
> 
> [99]
> 
> type = friend
> 
> secret = 99
> 
> host = dynamic
> 
> insecure = very
> 
> context = pstn-in
> 
> dtmfmode = inband
> 
> nat = no
> 
> qualify = 1000
> 
> disallow = all
> 
> allow = ulaw
> 
> allow = alaw
> 
> allow = gsm
> 
>  
> 
> extension.conf
> 
>  
> 
> [pstn-in]
> 
> exten => 99,1,Answer()
> 
> exten => 99,2,Goto,pstn|s|1
> 
>  
> 
> [pstn]
> 
> include => test-set
> 
> exten => s,1,Answer()
> 
> exten => s,2,Background(ext-or-zero)
> 
> exten => s,3,Wait(2)
> 
> exten => 0,1,Answer()
> 
> exten => 0,2,Background(one-moment-please)
> 
> exten => 0,3,Dial(SIP/2210,10)
> 
>  
> 
>  
> 
> it is working for my outbound dialing but for incoming when user press
> extension call is not forwarded to the right extension. log of
> asterisk (/var/log/asterisk/full) shows incorrect DTMF values.
> 
>  
> 
> Thanks in advance 
> 
>  
> 
> Regards
> 
> Dhananjay S