[Asterisk-Users] Voipbuster blocking Asterisk/IAX connections?

[Eric Wieling aka ManxPower]

Don Fanning wrote:
> Taking in everyone's suggestions (added a username line also) here is
> what I got.
> Still no joy
> ---
> 
> *CLI>
> *CLI>
> *CLI>
>     -- Executing SetCallerID("SIP/100-b225", ""xxxx"") in new stack
>     -- Executing Dial("SIP/100-b225",
> "IAX2/xxxxx at voipbuster/0015163011118") in new stack
>     -- Called xxxxx at voipbuster/0015163011118
>     -- Hungup 'IAX2/voipbuster/6'
>   == No one is available to answer at this time
>     -- Executing Congestion("SIP/100-b225", "") in new stack
>   == Spawn extension (internalselections, 90015163011118, 3) exited
> non-zero on 'SIP/100-b225'

Put a Noop(HANGUPCAUSE=${HANGUPCAUSE}) and a 
Noop(DIALSTATUS=${DIALSTATUS}) as the two priorities after your Dial to 
see WHY the call was hungup.
-- 
Eric Wieling * BTEL Consulting * 504-210-3699 x2120

r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user. Really, you almost certainly do not
want to use this. Asterisk will generate ring tones automatically where
it is appropriate to do so. "r" makes it go the next step and
additionally generate ring tones where it is probably not appropriate to
do so.