[asterisk-users] Asterisk and SIP Redirect message

[Johansson Olle E]

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 at 192.168.0.102| 
> 30|
> H|g") in new stack
>     -- Called 352000000 at 192.168.0.102
>     -- Got SIP response 300 "Redirect" back from 192.168.0.102
>     -- Now forwarding OSS/dsp to 'Local/testeur at 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 at 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 at 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 at sipside

/Olle