Hi, I am using Asterisk 1.4.13
The call comes in from a Coppercom soft-switch with a private IP of 192.168.104.2. This gets forwarded to a SessionBorderController with public IP of x.x.x.x,. This then gets to our asterisk server with a public IP of 8.7.192.58 and a private IP of 192.168.5.0. This asterisk server is behind a Cisco PIX firewall where we have disabled the SIP and UDP fixups. We have forwarded all the ports 10000 through 20000 and 5060 tcp/udp ports to the private IP of the asterisk server. When I dial 4025901000 the call flows through to the asterisk server where an IVR is played. When the user chooses 6 the asterisk initiates a new outbound call to 4023399500 using the inbound trunk i_sip_trunk. When the user chooses 7 the asterisk initiates a new outbound call to 4023399500 using the outbound trunk o_sip_trunk. The difference between the i_sip_trunk and o_sip_trunk is that the host ip. The i_sip_trunk has the host ip of the public IP of SessionBorderController whereas the o_sip_trunk has the host ip as that of the private ip of the asterisk server. 4023399500 is a test number when I am expecting to hear another IVR "Thank you for calling Ai Software Solutions we apply technology to solutions blah blah blah". Lets find out what happens when the user presses 6 there is extended period of silence. When the user presses 7 there is a fast busy received upon pressing the extension. Any help rendered from your end would be greatly greatly appreciated. I have been breaking my head to resolve this problem for a long time now. Thx Ravi Sip.conf file ; SIP Configuration example for Asterisk [general] context=i_sip_trunk context=o_sip_trunk allowoverlap=no bindport=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0; private IP of the Cisco PIX 506 firewall externip=8.7.192.58; public IP of the asterisk server srvlookup=yes allow=ulaw allow=alaw [i_sip_trunk] type=peer nat=no canreinvite=no host=x.x.x.x; public IP of the SBC qualify=yes dtmfmode=rfc2833 context=default [o_sip_trunk] type=peer nat=no canreinvite=no host=192.168.5.10; private IP of the asterisk server qualify=yes dtmfmode=rfc2833 context=default extensions.conf [general] static=yes writeprotect=no clearglobalvars=no [default] include => agnosco exten => h,1,Hangup exten => i,1,Congestion exten => i,2,Hangup [agnosco] include => agnosco_mainmenu include => i_sip_trunk include => o_sip_trunk [agnosco_mainmenu] exten => s,1,Answer exten => s,n,Background(agnosco_intro) exten => s,n,WaitExten ;Dial said extensions exten => 6,1,Dial(SIP/[EMAIL PROTECTED],30) exten => 7,1,Dial(SIP/[EMAIL PROTECTED],30) [i_sip_trunk] exten => 4025901000,1,Goto(1000,1) exten => 1000,1,Goto(agnosco_mainmenu,s,1) [o_sip_trunk] exten => _NXXNXXXXXX, 1,Dial(SIP/[EMAIL PROTECTED],30) exten => _NXXNXXXXXX, 2, congestion() _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users