[OpenSIPS-Users] convert 180 to 183 after the fact
jpyle at fidelityvoice.com
Mon Sep 23 14:41:16 CEST 2013
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.
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