Hi i use this into my extension :
exten => _00339xxxxxxxx,1,Set(foo=${SIP_HEADER(To)}) exten => _00339xxxxxxxx,2,Set(cut1=${CUT(foo,:,2)}) exten => _00339xxxxxxxx,3,Set(CLI=${CUT(cut1,>,1)}) exten => _00339xxxxxxxx,4,Set(toexten=${CUT(CLI,@,1)}) exten => _00339xxxxxxxx,5,Noop(ORIGINAL NUMBER : [ ${toexten} ]) exten => _00339xxxxxxxx,6,AGI(Ddi-Network.agi,${toexten}) exten => _00339xxxxxxxx,7,Set(CALLERPRES()=prohib_not_screened) exten => _00339xxxxxxxx,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt) exten => _00339xxxxxxxx,9,Hangup and i have in sip.conf: [MyOperator] type=peer host=host-of-my-operator qualify=yes dtmf=rfc2833 nat=no canreinvite=no canredirect=yes insecure=port,invite dtmfmode=rfc2833 disallow=all allow=g729 allow=alaw allow=g723 defaultuser=0033xxxxxx secret=xxxxx When i call directly from [MyOperator], no probleme i have sound/Voice but when a customer call to the "00339xxx..", the call are correct, asterisk call to my standard "SIP/MyOperator/${NUMAPPEL}" but no sound/voice (i receive the call without problems, only sound off) anyone have a idea of this problems ? bye Olivier -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users