Why not keep things simple and have the extension registered to site A in the first place? - would remove several layers of potential issues and areas to investigate.

Stuck in a never ending cycle of file copying.

if he phone is registered at B then when it makes a call the original traffic to site B is simply routed data & not passing through the IPO at site A
this means when site B connects the call back to Site A Neither IP Office knows that this is a trombone call for direct media path to be appropriate.

as sizbut has said the correct solution is to register the handset @site A & if necessary hot desk the Site B extn number.

Do things on the cheap & it will cost you dear

Why are you using H450 and not SCN?
SCN has the anti-tromboning that you want.

Hey Sizbut,
because we are converging 30 sites into one IPOSE. Centralized VM, directory, presence etc etc
Has to be this way as Site B has the PABX, site A has the phones and the ISDN gateway.
Voip Numpty commercial reasons but I would think a H450 could do this given there is a 'direct media' option.

Also noticed that handset at site A to handset at site A is also not using direct media.

Read my previous post.

Also
H450 is intended for connecting IPO to 3rd party systems - SCN is needed if you want anti tromboning to work.

If you are migrating to a Server edition the issue should go away as Server edition connects all the expansion servers using SCN anyway.
there seems to be little point in continuing to investigate a fault on a setup that is due to be replaced as long as you have a competent maintainer performing the upgrade




Do things on the cheap & it will cost you dear

OK, I have converted to an SCN trunk and I will test tomorrow. I have set networking level to none instead of SCN as I dont want the gateways to be part of the solution.
I could have sworn I tried manually creating an SCN trunk between IPOSE and IP500 a couple of years ago and it said it was an incompatible trunk (prob due to licensing) which is why i used H450

OK so I have tried websocket client/server, SCN proprietary and H450 as well as enabling 'allow direct media for devices behind NAT' and no combo of options have direct media between Site A handsets and Site A handset to Site A gateway
Logged ticket with Avaya

What version are on the phone systems ?
What is the link between the sites ?

10.1.1
MPLS with QoS

There are a few things which must be setup correctly to get anti-tromboning and direct media path working (in order of importance)

1st : all trunks MUST have a unique outgoing ID within the SCN, and I mean ALL trunks not only SCN trunks
2nd : Direct media patch MUST be enabled on all IP devices/trunks, direct media path is only possible if the routing within the network is properly setup
3th : all ip devices MUST use the same codec
4th : Always keep in mind there may be a software bug in IP Office which screw up your hard work, use sysmon to monitor the whole process.

Hey Intrigrant,
Forget the trunks, I have tested this between 2x IP Phones at site A registered directly to the IPOSE at site B, which match all requirements you have listed. Still no direct media.
Interestinly there is another site (site D) we have just setup and their phones are getting direct media so it must be a networking problem somewhere not a PABX problem.

1. Other than Media > VoiP Events > VoIP (Verbose) what do you recommend enabling in sysmon
2. I cannot find any RFC / spec on exactly how exactly direct media (aka shuffling) works with Avaya Phones ie is it part of H323, RTP, H245, SDP or proprietary. I would like to see in wireshark what is happening or not happening

In sysmonitor enable the filter System>Development tracing.
Then open Sysem>(S)RTP sessions, there you can see the ipadresses between IP devices for the audio stream.
Direct media path is embedded in the H.323 protocol which is used by all VoIP media devides.
I have a document somewhere about it with sysmon but I have a few gig documents and Avaya does not name them with the subject the document is about so i cannot find it.