After deploying many PBXs, I have never experienced such issues with a provider as I have with this one. I am hoping someone can explain to me if they know the issue or if it is something we can fix PBX end.
Basically, normal calls inbound and outbound work fine with audio, however there is one number (Germany non geographic) that when we call, there is no audio, so after 30s the call is dropped.
Having checked a wireshark capture, I see that the provider do not send any stream over the RTP session:
PBX IP: 192.168.222.3
SIP Trunk IP: 192.168.222.2
They have came back to me and explained:
If the Re-INVITE comes from the B-side without SDP, your equipment sets up an SDP offer, which is correct, but the version in the o-line would have to be incremented (at least 313269269 in this case).
From the capture above, there does not look to be a Re-INVITE sent from the B side. Even if there was, is there any way to change the way that our SDP sends the o-line mid call? I think they are just trying to put the issue down to our system, but the number we need to dial is critical to the business.
Thanks in advance