We are using Freeswitch as the core of our UK-based SIP platform, which connects with main carriers in the UK, and have a couple of issues which we have been unable to properly resolve ourselves. We are now seeking a Freeswitch expert who can assist in achieving resolution, on a consultancy/contract basis. Where is the most appropriate place to find suitably qualified experts, with proven track-records and endorsements? Layout is basically PSTN_CARRIER<->FS<-->CUSTOMER_PBX (there are also proxies and FS-based SBCs (Kamailio)) At high-level, the issues are: - Loss of RTP when a call is transferred or diverted by the CUSTOMER_PBX - Attempting to transfer a call from the CUSTOMER_PBX can result in ringback continuing, after the transfer-to endpoint answers (again, no RTP). - Behaviour varies depending on which PSTN_CARRIER calls are routed through (we have multiple interconnects) - We can see nothing obviously wrong in the call traces. Thank you Kind regards, Andy