8 sep 2006 kl. 17.18 skrev Michel Zenone:
> Hi!
> I try to make my Asterisk contact a SIP user thanks to a redirect
> server. In fact Asterisk try to reach a SIP address that is redirected
> to the good one.
>
> The error response is:
>
>
> *CLI> -- Executing Dial("OSS/dsp",
"sip/352000000@192.168.0.102|
> 30|
> H|g") in new stack
> -- Called 352000000@192.168.0.102
> -- Got SIP response 300 "Redirect" back from 192.168.0.102
> -- Now forwarding OSS/dsp to 'Local/testeur@sipside' (thanks to
> SIP/192.168.0.102-a4df)
> Sep 8 17:12:11 NOTICE[22263]: chan_local.c:479 local_alloc: No such
> extension/context testeur@sipside creating local channel
> Sep 8 17:12:11 NOTICE[22263]: app_dial.c:467 wait_for_answer:
> Unable to
> create local channel for call forward to 'Local/
> testeur@sipside' (cause
> = 0)
> == Everyone is busy/congested at this time (1:0/0/1)
> == Auto fallthrough, channel 'OSS/dsp' status is
'CHANUNAVAIL'
> == Console is full duplex
> << Hangup on console >>
>
>
> Does anybody know how to make Asterisk work with this?
Well, like always, reading the messages from Asterisk gives you a
hint. When Asterisk receives
the redirect, it goes back to the dialplan using the local channel.
In this case it looks for
testeur@sipside
/Olle