[Freeswitch-users] PBX Call-Transfer Issue - timing of RTP?

[Andy Newlands]

We have Freeswitch acting as a SIP platform, with internects with various
UK carriers (including BT) on one side, with customer PBX connected from
the other side.

We have an intermittent problem where a call comes into FS from the PSTN
and is routed to the customer PBX (and Avaya), with the call then
transferred back, via Freeswitch, then the PSTN to another phone number.

Sometimes the original caller continues to hear ringtone after the
transfer-destination has answered (and no conversation is possible).

I have attached Wireshark SIP flows for the scenario where this works and
the one where transfer does not complete properly (labelled in red, above
each ladder diagram, "BROKEN", on the left and "WORKING" on the right).

In the working example, we see RTP BEFORE the REFER.  In the non-working
example, RTP is sent AFTER the BYE.

1.  We don't understand why the Avaya sends messages in a different order,
on different occasions, for the exact same operation?
2. Should Freeswitch be able to handle both the scenarios shown in the
attached image?
3. Are there any config or dialplan settings which can be used to help fix
this?

NOTE: We also have other issues with call transfers in general, whereby the
transfer succeeds, but there is no audio heard by the transferee.

Thanks
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