Vodia PBX

Hmm it should just work. The user needs to have an email set for the account, but that's about it.

It should be in 68.0.37.

Generally speaking, when you set the log level to 9 for email it shows you what variables are available. Most emails are limited in what they have in terms of predefined variables. The queue statistics are a little bit more elaborated, maybe you can use them.

Should be there now.

Fanvil X210-V2 should be in 68.0.37. We can also make a 68.0.37 Win64 build if it is needed.

Yes. The page title is "RTP", so the "port range start" is actually about the RTP "port range start" on that page. Admittedly it's easy to miss that.

Did you try 69.2.4? Also we have seen that certain SIP trunks don't handle the changing of the SSRC in RTP very well. If the problem occurs only when talking to a SIP trunk and not when talking to an extension, this might be the problem. In that case the allow_ssrc_change needs to be set to false for the SIP trunk.

There are two different topics. One is the problem of sending bulk SMS to e.g. a group of parents and the other one is group messaging where everyone sees everyone else message. They both are obviously valid requirements. So far we have worked on the bulk part of it; as we see users starting to use it we will have to address the problem of transferring conversations around in the team, similar how you would transfer a call. As for the group chat, we actually had that in the early plannings. However in the real world, as of today, it seems most users don't use the PBX for such group chat. This is the world of the mobile phone chat apps, including iMessage and Android Messages that do that job very well.

Thanks we had it pretty much like that now; however the LDAP was still generally turned on; its probably better to turn it off for the LDAP devices.

Sometime the SIP trunk providers also change their stuff. In this case, this might have been a good thing.

From a non-UK perspective kind of hard to understand why they fight VoIP so vigorously. RTP and SRTP are the same. You can set the port range in /reg_rtpsettings.htm, however don't make the port range too narrow so that there is enough room for finding the free ports for a new call.

Not sure if STUN or TURN would help bypass carrier blocking VoIP. I would assume you would have to use VPN for that. SIP ALG are also not a problem because the apps always use TLS (invisible to ALG). The SBC also take care about all sorts of NAT. As long as the mobile phone can reach the PBX over HTTPS and SRTP, there should be audio.

This should be in the 69.2.4 .dat file.

Was that 10 minutes or 10 seconds? There was a 10 second delay internally for certain calls what should now be near-immediate with 69.2.4.

Would be great to know it this actually works for you...