I was wondering if anyone could possibly help me with this issue I’m facing. It may seem like a pretty straightforward case, but unfortunately, I introduced something that complicates it. Currently, I am using the latest version of free PBX. It is hosted on a dedicated VM with a static IP behind NAT. If that were the only factor, it would explain some of the one-way audio complications I am experiencing. However, it is not the case. Instead of using a VPN to secure the connection and open the port, I am using Twingate, which I have been using for a few years and find to be much more reliable than a traditional VPN. This is where the problem begins: the phone that is connecting to the PBX is receiving audio from extensions in the system and can receive calls back from the system, but there is no audio when the calls are on hold from the softphone. The extensions in the system can hear the on-hold music and vice versa when the softphone is put on hold, but when another extension calls the extension using the softphone, the softphone does not receive any voice audio, only the on-hold music. I have checked various ports and tried various solutions, but I can’t understand why the softphone can receive audio when put on hold or receive a call and send audio, but it just cannot receive audio for voice-only. Can anybody shed some light on this? I’m racking my brain trying to figure it out. I have made sure that Twingate ports are not the cause of this.
Is the PBX being used to anchor the RTP media? If not (end point to endpoint), that could explain what you are experiencing. The hold music is coming from the PBX, which you are connected to just fine. Also could be mismatched ports.
ITs all scattershot. Truthfully, you’re going to need to wire shark each leg of the journey to find where the RTP traffic drops. Once you determine that, it should be clearer on what you need to do.
I am using RTP ports 10000 to 60000 and none 5060 UDP ports for obvious security reasons, all softphones and desk phones are configured to use the same setup and this is including service providers. I have also made sure to tell Asterix that it is behind a nat with a fixed IP including the service providers, I’ve got to admit I’m not a fan of Surfshark, I’ve never had to dig this deep into a simple call drops,
I’m confident you are dropping the RTP packets, but no one is going to be able to just guess where. You could start with the basics and make sure all RTP ports are able to traverse, but this seems longer and indirect.
I found the issue, it was the VPN ports were not configured to the way their service would recognise them, believe it or not, it was something as simple as telling it to use the ports in this configuration 10000-60000 I was hyphenating between ports and not using this symbol -I can’t believe it was a simple as this it’s only when you said about dropping something and made me check the documentation for the VPN service provider. Thank you so much for your help.
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