michel.brabants at gmail.com
Mon Jun 29 20:35:57 UTC 2020
not interested in the consulting myself, but as I have a small bit of using
freeswitch as an sbc in the past, did you try to enable the option on your
call to rewrite the rtp-timestamps? This may fix the first issue, depending
on the exact sip-scenario.
<param name="*rtp*-*rewrite*-*timestamps*" value="true"/> ...
You can do a lot with transfers in freeswitch. If it's a refer, you'll have
to implement the refer-dialplan-part, but it has been a while and maybe I'm
Just a quick idea that help you along maybe.
On Mon, Jun 29, 2020 at 8:44 PM Andy Newlands <andynewlands at gmail.com>
> 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
> - 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,
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
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