Product: Yeastar S50
HW Version: V2.20 0000-0000
Firmware 30.10.0.28
We have problem with Linkus.
Every outgoing call is ending itself after 30 seconds.
Beside Linkus we have of course other physical phones registered to PBX and here everything works just fine.
Any ideas ?
Please advise.
PBX is behind UBNT ER-Lite-3
Did you ever try LCS(Linkus Cloud Service)? If not, I would suggest you to try and compare if you would have the same issue.
This is typically a NAT issue. The T1 timer expires after 32 seconds if an 200OK is not seen in the interim after the INVITE. If the OK is not seen, the call will drop.
You should do a capture - Maintenance, troubleshooting and Ethernet capture. Using wireshark you can examine the packets captured during a test call and see if the IP being sent and received is correct in the Contact and Connection headers for both the PBX and Linkus.
There are a number of questions that might help further as you did not indicate the version of Linkus - Desktop or cell, how Linkus is connected - LAN, Cellular network or Wi-Fi and if local or remote from the PBX?
Nevertheless, a capture will show the issue.
I have been using the Linkus cloud service for about a month with good success. Its pretty cost effect as well they give you a 30 day trial with the chat function as well. I would try to up sell the customer its a great cost effect tool
Yes, I agree with Larry, this is probably a NAT issue.
Could you pelase post screenshot your NAT settings on the S PBX and a pcap log would be prefer.
Linkus Cloud Service is perfect without worrying NAT issues per my understanding that is why I suggest you to try and compare.
You were right. NAT and Port Forwarding did a trick.
Now calls are not dropping after 30 seconds.
But ... can't be ideal. First 4-5 second of the call is very distorted. After that period of time, all is good.
Any advise ?
Hard to say without seeing the capture file and knowing how connected. Given the reference to the router, it appears that you must have been using the service externally, but still unknown if Wi-Fi or cellular and what kind. Try both connection methods if possible and see if a difference.
So here is our network structure:
Modem from ISP ( In Bridge mode ) -> Our Main router (UBNT ERLite-3 with Public IP address -> LAN ( Included PBX )
Inside our LAN, we have many VOIP physical phones which works like a charm.
Few weeks ago we decided to use Linkus for our employees who works out side of the company ( from home or other places ). They are using mainly a cell phones with installed Linkus app. Some of them are using WiFi on the, some normal cellular (LTE ) -> both have same issue, so distorted sound quality during first couple seconds of the call. After it, all is fine. So generally, they are connecting to PBX via WAN IP address of our main router.
For test purposes I have installed Linkus Desktop on my Windows laptop and I did a test from my home. Same issue.
I'm adding some screen shots from our PBX and Router.
Is it possible for you to provide a pcap logs for the call so that we can understand the distorted voice issue.
By the way, I would suggest you submit a ticket at the same time so the support team could help also.
A couple of things-
1. The RTP range in the PBX is set to 10000-12000, but in the router the forwarding appears to be much larger. This is not a cause of the issue, but rather an observation of the expanse of forwarded ports that does not appear to be needed.
2. Why the port translation on the Linkus external IP? Why not use 8111 and eliminate the translation? I do not think this to be a cause either, but just odd by not maintaining a straight NAT.
3. Does the router have SIP ALG enabled? If so, try disabling it although I have been told by others that the UBNT ALG is really good and works as one would hope. I do not have UBNT, so am unable to confirm.
A capture is the only way.
It would help to know the number or extension called from and to in your testing so we don't have to look at each line to see which is a Linkus extension. Nevertheless, the capture shows no audio.
I see a Linkus call from extension 30 to an external number. It is coming from a private IP in the 172 subnet and then to the S-PBX in the 192.168.12.X subnet. I cannot see a reply from the S20, which makes me suspect that you are have the network setting on the PBX to something other than LAN.
Can you advise how the S20 is set? Also, why do you have local SIP ports and TCP enabled? Are you using TCP?
Please advise on above before we go forward. I need to understand the scenario and the setup.
" I see a Linkus call from extension 30 to an external number. It is coming from a private IP in the 172 subnet and then to the S-PBX in the 192.168.12.X subnet. "
Larry, I agree this part, but it seems like it is S20 gets no ACK from 172 subnet that cause the SIP section not establish correctly. I see from the 200 OK that sent from S20 to Linkus, it is using public IP of S20 in the contact header and in SDP, which would caused Linkus sent the ACK to the public address.
Try add 172 subnet as Local network Identification into your NAT settings on S20 and see if that helps.
Oscar, you are correct. I had not gotten that far last night as I only looked at the flow. As Maciej had earlier indicated he had fixed the NAT problem and all was good with the calls with the exception of the first few seconds of audio, I never looked at the final OK from the PBX and did not notice the plethora of other OKs.
So, yes add the 172 subnet into the NAT section.
Oscar, good catch, thanks.