asterisk users - Jan 2008 - Unable to forward call on SIP channel after SIP response 302 Moved Temporarily

Hi,

I have the following problem that when asterisk receives SIP response 302 it
cannot forward the call
I get such debug:
[Jan  4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel
type registered for 'Local'
[Jan  4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to
create local channel for call forward to 'Local/poczta at routing-sip'
(cause 66)

What could be the reason for this?

Thank You in advance
bests

-tomasz

<--- SIP read from 192.168.0.165:7060 --->
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 192.168.0.165:7070;branch=z9hG4bK274b7d50;rport=7070
Contact: <sip:poczta at voip.rd.touk.pl:7170>
To: <sip:hellboy at voip.rd.touk.pl:7060>;tag=12711d6c
From: "IPFon"<sip:0225761853 at
192.168.0.165:7070>;tag=as66773d49
Call-ID: 08027b6d45abc7a00c2c6a8630d2de47 at 192.168.0.165
CSeq: 102 INVITE
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 0


<------------->
[Jan  4 10:43:27] --- (9 headers 0 lines) ---
[Jan  4 10:43:27]     -- Got SIP response 302 "Moved Temporarily" back
from
192.168.0.165
[Jan  4 10:43:27] Transmitting (no NAT) to 192.168.0.165:7060:
ACK sip:hellboy at voip.rd.touk.pl:7060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.165:7070;branch=z9hG4bK274b7d50;rport
From: "IPFon" <sip:0225761853 at
192.168.0.165:7070>;tag=as66773d49
To: <sip:hellboy at voip.rd.touk.pl:7060>;tag=12711d6c
Contact: <sip:0225761853 at 192.168.0.165:7070>
Call-ID: 08027b6d45abc7a00c2c6a8630d2de47 at 192.168.0.165
Seq: 102 ACK
User-Agent: TouK S.K.A
Max-Forwards: 70
Content-Length: 0


---
[Jan  4 10:43:27]     -- Now forwarding SIP/213.218.117.72-00837b20 to '
Local/poczta at routing-sip' (thanks to SIP/voip.rd.touk.pl-00843c10)
[Jan  4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel
type registered for 'Local'
[Jan  4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to
create local channel for call forward to 'Local/poczta at routing-sip'
(cause 66)
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On Friday 04 January 2008 11:58:25 Tomasz Zieleniewski
wrote:
> Hi,
>
> I have the following problem that when asterisk receives SIP response 302
> it cannot forward the call
> I get such debug:
> [Jan  4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel
> type registered for 'Local'
> [Jan  4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to
> create local channel for call forward to 'Local/poczta at
routing-sip' (cause
> = 66)
Maybe this:
"Local channel
 Description: Local Proxy Channel Driver
 Syntax: Local/extension at context/n
 Configuration file: none

chan_local is a pseudo-channel. Use of this channel simply loops calls back 
into the dialplan in a different context. Useful for recursive routing.

Notes: Adding "/n" at the end of the string will make the Local
channel not
do a native transfer (the "n" stands for "n"o release) upon
the remote end
answering the line. This is an esoteric, but important feature if you
expect the Local channel to handle calls _exactly_ like a normal channel.
If you do not have the "no release" feature set, then as soon as the
destination (inside of the Local channel0 answers the line, the variables
and dial plan will revert back to that of the original call, and the Local
channel will become a zombie and be removed from the active channels list.
This is desirable in some circumstances, but can result in unexpected
dialplan behavior if you are doing fancy things with variables in your call
handling. "

Boyko