[Asterisk-Users] can't dial some particular numbers (providers ?) with asterisk sip / zap channels

Simone Cittadini mymailforlists at gmail.com
Fri Feb 24 02:02:34 MST 2006


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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