<DIV>Thanks for the information.</DIV>
<DIV>But still we are facing the same problem.</DIV>
<DIV>We tried upgrading the firmware to latest available on sipura website and still the result is same.</DIV>
<DIV> </DIV>
<DIV>Does any specific DTMF setting required? we have tried all the 3 options in asterisk (inband, rfc2833 and info) but no luck</DIV>
<DIV><BR><BR><B><I>Joseph <syscon@interbaun.com></I></B> wrote:</DIV>
<BLOCKQUOTE class=replbq style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">On PSTN-Line tab<BR><BR>Subscriber Information<BR>User ID: 99<BR>Password: 99<BR><BR>Dial Plans<BR>Dial Plan 1: S0<:99><BR><BR>PSTN-To-VoIP Gateway Setup<BR>PSTN-To-VoIP Gateway Enable: Yes<BR>PSTN Ring Thru Line 1: Yes<BR>PSTN Caller Default DP: 1<BR><BR>That should be it I think.<BR><BR>-- <BR>#Joseph<BR><BR><BR>On Tue, 2005-03-01 at 04:34 -0800, dhananjay sarnaik wrote:<BR>> Dear All<BR>> <BR>> <BR>> <BR>> Im facing wearied problem with Sipura 3000 and asterisk .<BR>> <BR>> <BR>> <BR>> Im trying to configure Asterisk with Sipura 3000 . I have configured<BR>> asterisk with FSX port which is working fine.<BR>> <BR>> I want to configure Asterisk FXO port for my outgoing and incoming<BR>> calls.<BR>> <BR>> Once Sipura received call from outside it will deliver to Asterisk and<BR>> asterisk will play IVR user dial any
extension<BR>> <BR>> Here is my configuration<BR>> <BR>> <BR>> <BR>> sip.conf<BR>> <BR>> <BR>> <BR>> [99]<BR>> <BR>> type = friend<BR>> <BR>> secret = 99<BR>> <BR>> host = dynamic<BR>> <BR>> insecure = very<BR>> <BR>> context = pstn-in<BR>> <BR>> dtmfmode = inband<BR>> <BR>> nat = no<BR>> <BR>> qualify = 1000<BR>> <BR>> disallow = all<BR>> <BR>> allow = ulaw<BR>> <BR>> allow = alaw<BR>> <BR>> allow = gsm<BR>> <BR>> <BR>> <BR>> extension.conf<BR>> <BR>> <BR>> <BR>> [pstn-in]<BR>> <BR>> exten => 99,1,Answer()<BR>> <BR>> exten => 99,2,Goto,pstn|s|1<BR>> <BR>> <BR>> <BR>> [pstn]<BR>> <BR>> include => test-set<BR>> <BR>> exten => s,1,Answer()<BR>> <BR>> exten => s,2,Background(ext-or-zero)<BR>> <BR>> exten => s,3,Wait(2)<BR>> <BR>> exten => 0,1,Answer()<BR>> <BR>> exten =>
0,2,Background(one-moment-please)<BR>> <BR>> exten => 0,3,Dial(SIP/2210,10)<BR>> <BR>> <BR>> <BR>> <BR>> <BR>> it is working for my outbound dialing but for incoming when user press<BR>> extension call is not forwarded to the right extension. log of<BR>> asterisk (/var/log/asterisk/full) shows incorrect DTMF values.<BR>> <BR>> <BR>> <BR>> Thanks in advance <BR>> <BR>> <BR>> <BR>> Regards<BR>> <BR>> Dhananjay S<BR><BR><BR>_______________________________________________<BR>Asterisk-Users mailing list<BR>Asterisk-Users@lists.digium.com<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR>To UNSUBSCRIBE or update options visit:<BR>http://lists.digium.com/mailman/listinfo/asterisk-users<BR></BLOCKQUOTE><p>
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