I am using asterisk 1.2.13 with multiple SIP service providers to handle
my out going call. I have my dialplan list below.
It works fine with one exception -
when my primary SIP provider off air - not responding to the SIP INVITE-
it took 60 seconds for sip.channel to return with DIALSTATUS=NOANSWER.
I would expect the SIP channel failed the call immediately after T2
timeout. I check the sip.conf and can not find any configuration that
will affect this, any suggestion - that will make SIP channel return a
failure status after the T2 timeout.
[out-trunk]
exten => s,1,Dial(SIP/a.a.a.a/${EXTEN},180)
exten => s,n,NoOp(Call complete DIALSTATUS=${DIALSTATUS})
exten => s,n,Goto(SecondaryRoute,s-${DIALSTATUS},1)
[SecondaryRoute]
exten => s-NOANSWER,1,NoOp(Primary Route - NOANSWER)
exten => s-NOANSWER,n,Dial(SIP/b.b.b.b/${EXTEN},180)
exten => s-BUSY,1,NoOp(Primary Route - BUSY)
exten => s-BUSY,n,Dial(SIP/b.b.b.b/${EXTEN},180)
exten => s-CHANUNAVAIL,1,NoOp(Primary Route - CHANUNAVAIL)
exten => s-CHANUNAVAIL,n,Dial(SIP/b.b.b.b/${EXTEN},180)
exten => s-CONGESTION,1,NoOp(Primary Route - CONGESTION)
exten => s-CONGESTION,n,Dial(SIP/b.b.b.b/${EXTEN},180)
exten => _s-.,1,Congestion
exten => s-,1,Congestion
begin:vcard
fn:KeOu Chao
n:Chao;KeOu
email;internet:[EMAIL PROTECTED]
tel;work:703-648-0066
x-mozilla-html:FALSE
version:2.1
end:vcard
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
