Adil Mafzool Zaka
adil_mafzool at yahoo.com
Mon Jun 29 22:42:10 UTC 2020
You would need to do wireshark captures trace with signalling and RTP which component FS or PBX have the issue. For FS whether you are using proxy Media or by pass media on sip profile / dial plan. Which version of FS you using, is it release version or custom build. There are so many factors to see. You can raise the issues in github as well on Freeswitch as there is a large active community.
At my work we use custom build FS as SBC and its being quite stable overtime.
RegardsMuhammad Adil M Zaka+44(0)7776114280
On Monday, June 29, 2020, 06:59:06 PM GMT+1, Andy Newlands <andynewlands at 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.
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