On Thu, 2008-09-18 at 16:37 +0100, Steve Davies wrote:
> Hi,
> 
> I am trying to put together an app to do essentially the same as the
> Manager "Action: Redirect" operation, so that 2 channels in a bridged
> call can be bounced off into the dialplan to do their own thing.
> 
> It is ALMOST working. The code is not complicated after-all, it runs
> ast_async_goto(...) on the 2 halves of the bridge once they've been
> identified. If I set canreinvite=no, or if the call is a SIP<->ZAP
> call, it works 100%.
> 
> The problem is if I have a reinvited SIP<->SIP call, then chan_sip/rtp
> never seems to reinvite the call back to Asterisk, so the audio paths
> which are subsequently set-up are all over the place.
> [...]
> 
> Any suggestions - I am running a Frankenstein version of asterisk, so
> it is possible this has already been discovered and fixed in a future
> version.

I had a similar problem and fixed it with a patch to chan_sip that went
into 1.4, trunk and 1.6.0 but if it's not included in your version it
might be worth to check if it solves your problem, see
http://bugs.digium.com/view.php?id=12513

Cheers,
	Mike
-- 
Dr. Michael Neuhauser                              mailto:mike@firmix.at
Firmix Software GmbH                                  sip:mike@firmix.at
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