[Freeswitch-dev] Consultancy?

[Andy Newlands]

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
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