I use an Asterisk box as a switch between 2 SIP trunks (one trunk to the “PSTN”, one trunk to our IP-Centrex softswitch).
It does RTP proxy too.
I am having some issues with RTP media loss on the customers connected to our IP-Centrex, so I need to make a deep analysis of some calls AFTERWARDS (because the issue we are having is random).
My issue is that on the Asterisk side, I can dig and find a call in the logs and the CDRs, but I can’t find a way to log the RTP ports of both call legs which were used for a specific call going through the Asterisk.
From a debugging perspective, it’s a nightmare.
I really need a SIP channel to RTP ports (at least the local one) mapping.
Anyone has a solution for that ?
Perhaps a debugging option I missed somewhere ?
Thanks in advance