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

[Tomasz Zieleniewski]

Thanks it helped, I had the noload in modules.conf.
But now I have another problem:

When 302 response is received by asterisk it falls in to some context.
according to rfc 3261 uac which receives 302 should retry the request at the
address given by the contact header filed.
I am not able to make the same routing decision because  conditions  are
different.
What can I do here.
I have for instance such problem that my asterisk works as a gateway.
When there is an external call this call is forwarded to some internal sip
address.
After this my sip client responses with 302 which point to his voicemail
(sip uri in the contact).
What can be done in such situation to make is work??


On Jan 4, 2008 1:06 PM, Johansson Olle E <oej at edvina.net> wrote:

>
> 4 jan 2008 kl. 11.50 skrev 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.
>
> You have to enable chan_local in menuselect (1.4) and make sure it's
> not disabled
> in modules.conf.
>
> This is not a developer question, so please take this kind of
> questions to
> asterisk-users in the future. Thank you!
>
> Best regards,
> /Olle
>
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