I'm trying to make a dialplan: <extension name="Long Dial"> <condition field="destination_number" expression="^(^([0-9]{10})$|^([0-9]{11})$|^([0-9]{12})$|^\+([0-9]{11})$)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="failure_causes=TIMEOUT,NO_ROUTE_DESTINATION"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/> <action application="set" data="auto_hunt=true"/> <action application="set" data="real_dialed_number=7777777"/> </condition> <condition field="destination_number" expression="^(79|89)(\d{9})$" break="on-true"> <action application="set" data="dtmf=WWWWWWWWWWWWWWWWWWWWW111222#WWWWWW89$2#@100"/> <action application="bridge" data="sofia/gateway/gw1/89$2|sofia/gateway/gw2/89$2|sofia/gateway/gw3/89$2"/> <action application="transfer" data="to_card"/> </condition> </extension> <extension name="to_card"> <condition field="destination_number" expression="^to_card$"> <action application="bridge" data="{execute_on_answer=send_dtmf\s${dtmf}}sofia/internal/${real_dialed_number}@192.168.50.53:5061"/> </condition> </extension> The logic of his work: if unavailable ilti gateway to the next. if not available all the gateway to go to the extension transfer. A problem arises when user_busy on the phone: ------------------------------------------------------------------------ SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 192.168.50.11:5080;rport;branch=z9hG4bK1X4r00t93KmFg From: "Vladimir Elizarov" <sip:0000xxxx@xx.xx.xx.xx;transport=udp>;tag=yj15KpSQvK6eF To: <sip:8931237xxxx@xx.xx.xx.xx>;tag=2a5549004a431d10ff00001a646aff45 Call-ID: 1317949e-f694-122d-8fb6-00163efcbed2 CSeq: 132374757 INVITE Server: MERA MSIP v.1.0.2 Reason: Q.850;cause=17;text="User busy" Content-Length: 0 instead of making hangup, dialplan is going to transfer. Constant despite continue_on_fail = NORMAL_TEMPORARY_FAILURE, TIMEOUT, NO_ROUTE_DESTINATION" What's the problem? freeswitch 1.0.6 (git 19 june 2010) -- Best regards, Vladimir Elizarov