ON my SV8500 I have mostly DT700 IP phones,but have a few SP350 softphones and have many 3rd party analog gateways (Sangoma Vega 5000). I have plenty of proper licenses for each end-points.
Recently it was reported to me that when a user of an analog phone connected to the analog gateway calls an SP350 softphone, there is only 1-way audio. The analog-phone user can hear the softphone user, but the softphone user can't hear the analog phone. No matter who initiates the calls.
So I started testing this out under different circumstances, and it always boils down to just that pair: 3rd party SIP and SP350.

-3rd party SIP to DT700 GOOD both ways
-3rd party SIP to 3rd part SIP GOOD both ways
-3rd party SIP to SP350 FAIL
-DT700 to SP350 GOOD both ways

While we have many different subnets and VLANs, I ruled out access list problems and packet block by:
- for the sake of test, removed ALL blocking between all networks. The problem persisted.
- Tried to put the 3rd part SIP gateway on the same subnet as the SP350. Problem persisted.

Lastly, I installed Wireshark on the PC that hosts one of the SP350 phones. It captured voice packets going both ways. Wireshark has a feature to decode VOIP calls from its captured packets. I did that. It found that call and played it back to me nicely with 2-way audio!!! So really, SP350 is somehow just refusing to play the sound of the 3rd party SIP phone. But it can't be an audio setting on the PC or the SP350 itself since it plays nicely the sound of other SP350 callers or other DT700 callers.

So what on earth is going on??? I'm using SP350 version 5.2.0 (I think) with HP Thinclients running Win 7, but the problem existed on a Win XP machine and SP350 version 4.something.

I compared the voice packets coming to the SP350 from the DT700 and from the 3rd party SIP device, and found the below differences:
-DIFFSERV value (0x00 for the 3rd part SIP and 0x8b for the DT700)
-UDP port number is 3462 for the DT700 and 10000 for the 3rd party SIP device.
-the packet length is different
- They show the same codec: G711u

I then reconfigured the 3rd party device to use the same UDP port and same DIFFSERV value as the DT700, and STILL NO AUDIO!
I contacted my vendor's tech support who is pointing me towards playing with the LOC-ID settings in the AIVCL/N commands. NEC recommends a G711 with payload of 20ms of G729 with payload of 40ms setting. I tried those but still nothing.

The install manual of the SP350 says that Win 7 only supports 4 levels of TOS setting for the SP350, and the PBX has to be configured accordingly. However, it doesn't say how to find out what is the TOS setting in my Win 7 computer for the SP350, thus I have no idea how to match that in the PBX.

Anyways. This isn't going anywhere. Oh by the way, I then tried a Grandstream analog SIP gateway just for a test and the exact same audio problem exists.

Any great ideas?