[asterisk-users] Asterisk and SIP Redirect message
Michel Zenone
zenone at telip.com
Fri Sep 8 08:18:30 MST 2006
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?
Thanks a lot,
Michel
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