I have a strange problem when calling some numbers with asterisk, I get
an hangup for busy condition even if the phone at the other end isn't busy.
I can route the calls via SIP to another carrier and then I have a SIP
code 486
or I can terminate them on digium cards (E1) and I have an Hangup code 17
I know for sure that one of the numbers is hosted by a different
provider than the one that has the de-facto monopoly here, so maybe is a
final-provider problem, even if I don't understand what kind of strange
signalling can reach that provider from my asterisk, I don't see nothing
unusual on the cli, is like any other call ended for a real busy condition.
More weird is that with the SIP route the called phone rings once, than
stops and I get the 486.
What have I've already tried :
Set(CALLERID(number)=[a real "traditional" phone number]) before the dial
SetTransferCapability(SPEECH)
as far as I know the route calls follow is :
linksys pap --sip--> asterisk (1.2 or 1.0) --iax--> asterisk server (1.2) --zap--> ..?..
<---- Hangup cause 17
linksys pap --sip--> asterisk (1.2 or 1.0) --iax--> asterisk server
(1.2) --sip--> ..?..
<----- 486 Busy here (but the end phone ringed once)
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