I tried this http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html
But I am NOT getting call in asterisk. SIP.conf file : _________________ [general] port = 5060 bindaddr = 0.0.0.0 context = default externhost=59.160.44.21 localnet=192.168.0.2/255.255.255.0 ; register SIP account on remote machine if using SIP trunks ; register => testSIPtrunk:[email protected]:5060 ; [sip] type=peer username=fiducia_ag fromuser=fiducia_ag authuser=fiducia_ag secret=password host=64.56.64.64 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833 [ipkall.com] host=voiper.ipkall.com context=from-ipkall dtmfmode=rfc2833 insecure=invite type=friend canreinvite=no disallow=all allow=ulaw Extension.conf: _________________ [from-ipkall] exten => 901835,1,NoOp(from-ipkall) exten => 901835,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM}) exten => 901835,3,Dial(Local/200 at internal)
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