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

[Steve Davies]

2008/9/18 Michael Neuhauser <mike at firmix.at>:
> 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


Mike,

You are officially my hero :) I was indeed missing that patch, and
everything suddenly works fine.

Regards,
Steve