<div dir="auto"><div>Freeswitch has very large user community accessible through its mailing lists as well as official consultancy services provided by its original developers at <a href="http://signalwire.com">signalwire.com</a>.</div><div dir="auto"><br></div><div dir="auto">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.</div><div dir="auto"><br></div><div dir="auto">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.<br><br>For example, for RTP issue you mentioned, can you share sip trace at problematic endpoint (customer PABX in this case)?</div><div dir="auto"><br></div><div dir="auto"><br></div><div dir="auto">Thank you.</div><div dir="auto"><br></div><div dir="auto">--</div><div dir="auto">Muhammad Shahzad Shafi</div><div dir="auto">Burraq Technologies UG</div><div dir="auto">Frankfurt, Germany</div><div dir="auto">Tel: +49 176 24 73 85 20</div><div dir="auto"><br><br><br><div class="gmail_quote" dir="auto"><div dir="ltr" class="gmail_attr">On Mon, 29 Jun 2020, 19:59 Andy Newlands, <<a href="mailto:andynewlands@gmail.com">andynewlands@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">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.<div><br></div><div>Where is the most appropriate place to find suitably qualified experts, with proven track-records and endorsements?</div><div><br></div><div>Layout is basically PSTN_CARRIER<->FS<-->CUSTOMER_PBX</div><div>(there are also proxies and FS-based SBCs (Kamailio))</div><div><br></div><div>At high-level, the issues are:</div><div><br></div><div><ul><li>Loss of RTP when a call is transferred or diverted by the CUSTOMER_PBX</li><li>Attempting to transfer a call from the CUSTOMER_PBX can result in ringback continuing, after the transfer-to endpoint answers (again, no RTP).</li><li>Behaviour varies depending on which PSTN_CARRIER calls are routed through (we have multiple interconnects)</li><li>We can see nothing obviously wrong in the call traces.</li></ul></div><div><div><div><div dir="ltr" data-smartmail="gmail_signature"><div dir="ltr"><div>Thank you</div><div><br></div>Kind regards,<div><br></div><div>Andy</div></div></div></div></div></div></div>
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