Freeswitch has very large user community accessible through its mailing lists as well as official consultancy services provided by its original developers at signalwire.com. I, myself have been working in various Freeswitch based products and services for over 12 years and can assist if you prefer paid consultancy services. Generally speaking, if time & confidentiality permits, you can share your technical problems in user mailing list and get almost instant response from multiple experts for free which offers much more flexible solutions then hiring a single paid expert. For example, for RTP issue you mentioned, can you share sip trace at problematic endpoint (customer PABX in this case)? Thank you. -- Muhammad Shahzad Shafi Burraq Technologies UG Frankfurt, Germany Tel: +49 176 24 73 85 20 On Mon, 29 Jun 2020, 19:59 Andy Newlands, <andynewlands@gmail.com> wrote:
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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