Hello,
I've got an Asterisk system with 3 SIP trunks configured. Each SIP
trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound
call routing (via AMP 1.10.007a) uses the 3 trunks in descending order,
all set with max channels to 4. Unfortunately, when the first trunk
reports a "480 Service Unavailable" (all ports in use), Asterisk
reports
congestion without rolling to the next available trunk.
I've looked at the AMP dialplan on this system, as well as a more
recent system (1.10.009beta1), and although the dialplan has been
improved, it still doesn't seem to account for this condition. As you
can see below at step 7, if the max channels have been used on the
current trunk, the call fails.
What is the correct way to do fail over between trunks, and in an AMP
friendly way that won't get clobbered during the next config change?
Regards,
Chris
[macro-dialout-trunk]
exten => s,1,GotoIf($[foo${ARG3} = foo]?3:2)) ; arg3 is pattern password
exten => s,2,Authenticate(${ARG3})
exten => s,3,Macro(user-callerid)
exten => s,4,Macro(record-enable,${CALLERIDNUM},OUT)
exten => s,5,Macro(outbound-callerid,${ARG1})
exten => s,6,SetGroup(OUT_${ARG1})
exten => s,7,CheckGroup(${OUTMAXCHANS_${ARG1}})
; if we've used up the max channels, continue at 110 (n+101)
exten => s,8,SetVar(DIAL_NUMBER=${ARG2})
exten => s,9,SetVar(DIAL_TRUNK=${ARG1})
exten => s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper
dial string for this trunk
exten => s,11,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ;
OUTNUM is the final dial number
exten => s,12,Cut(custom=OUT_${ARG1},:,1) ; Custom trunks are prefixed
with "AMP:"
exten => s,13,GotoIf($[${custom} = AMP]?16)
exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM}) ; Regular Trunk Dial
exten => s,15,Goto(s-${DIALSTATUS},1)
exten => s,110,Noop(max channels used up)
exten => s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten => s-BUSY,2,Busy()
exten => s-BUSY,3,Wait(60)
exten => s-BUSY,4,NoOp()
exten => _s-.,1,NoOp(Dial failed due to ${DIALSTATUS})
