[asterisk-users] Unable to forward call on SIP channel after SIP response 302 Moved Temporarily

[Benchev]

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