Jimmy wrote: Second Call out the asterisk console looks like this-----------------------------------------------------: -- Executing [92952...@internal:1] Dial("SIP/222-09ab3588", "SIP/Cisco1760/2952210") in new stack -- Called Cisco1760/2952210 [Apr 22 16:08:58] NOTICE[3450]: chan_sip.c:14489 handle_request_invite: Call from '222' to extension '2952210' rejected because extension not found. -- Got SIP response 486 "Busy here" back from 172.17.2.1 -- SIP/Cisco1760-09ab7cf8 is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing [92952...@internal:2] Congestion("SIP/222-09ab3588", "") in new stack == Spawn extension (internal, 92952210, 2) exited non-zero on 'SIP/222-09ab3588' localhost*CLI>
--------------sip.conf --------- [general] bindaddr=0.0.0.0 [Cisco1760] context=incoming_calls type=friend host=172.17.2.1 dtmfmode=rfc2833 disallow=all allow=ulaw insecure=very ----------extensions.conf------------ [globals] OUTBOUNDTRUNK=SIP/Cisco1760 [outbound-local] exten => _9NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9NXXXXXX,n,Congestion() exten => _9NXXXXXX,n,Hangup() -----------Cisco 1760 config ---------- dial-peer voice 100 pots (This line that is set to preference 2 does not work) huntstop preference 2 destination-pattern .T port 0/0 forward-digits all ! dial-peer voice 2212 pots (This line that is set to Preference 1 is the one that works) huntstop preference 1 destination-pattern .T port 0/1 forward-digits all ------------------------------------------------ You do not want to use huntstop on the dialpeers in this situation. The huntstop option tells the call routing function in the router to stop search for a call route if it encounters a failure. Call number 2 hits dialpeer 1, finds it busy and the huntstop causes the processing to stop. Dan _______________________________________________ -- 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