[asterisk-users] Asterisk and SIP Redirect message
Johansson Olle E
olle at voop.com
Fri Sep 8 09:32:03 MST 2006
8 sep 2006 kl. 17.18 skrev Michel Zenone:
> 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|
> 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
> Sep 8 17:12:11 NOTICE: chan_local.c:479 local_alloc: No such
> extension/context testeur at sipside creating local channel
> Sep 8 17:12:11 NOTICE: 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
More information about the asterisk-users