asterisk users - Mar 2005 - Sipura 3000 Inbound Dialing Problem

Dear All

 

I’m facing wearied problem with Sipura 3000 and asterisk .

 

I’m 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


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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