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