[asterisk-dev] SIP reinvite back to asterisk during async_goto

[Steve Davies]

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. "show channel
..." shows that the channel has been "un-bridged", but the rtp code
never seems to be called :( I also tried to send an AST_CONTROL_HOLD
to the channel before doing the async_goto, in the hope that this
would cause asterisk to grab the call before I do the goto, but it
seems to make no difference - Probably because I do not release the
channel lock between sending the HOLD, and doing the async_goto.

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.

Cheers,
Steve