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 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
