[Freeswitch-users] Call progress not being passed

[David Witham]

Hi all,

We're running the latest FREESwitch 1.4 (1.4.26) on debian7.

We have a problem where we receive a SIP 183 from our upstream provider
(via a sofia gateway) but the 183 does not get forwarded to the internal
leg we bridge to. When we receive the 200 OK, that gets passed through and
the call is established with media.

We can add a ring_ready into the dialplan and force it to pass a 180 back
to the A party but that just masks the problem.

Has anyone else come across this behaviour? I haven't found any JIRAs that
match this although perhaps FS-7714 is related.

thanks,
David
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