[OpenSIPS-Users] convert 180 to 183 after the fact

[Laszlo]

What if you simply drop the 180 in the onreply_route?

-Laszlo


2013/9/23 Jeff Pyle <jpyle at fidelityvoice.com>

> Hello,
>
> I have one particular PSTN call flow that causes a 183 with SDP, then a
> 180 without SDP prior to 200 OK.  Some of my customer endpoints don't
> handle the 180 properly after a 183 and they cease to hear ringback.
>
> I'm thinking through how intercept the 180 and convert it to a 183 with
> SDP.  I have a media server available to generate the 183 and the media.
>  I'm struggling with how to relay the INVITE to the media server when the
> 180 arrives in the middle of the call setup.
>
> Any recommendations are appreciated.
>
>
>
> Regards,
> Jeff
>
>
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-- 

--
Kind regards,
Laszlo Bekesi
http://voipfreak.net
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