Vodia PBX

The reformatting had the JavaScript look at the wrong element. We'll fix that in the next build (68.3.7).

There is a function called call.transfer(target). We'll have to start working on documenting what functions are available within the call...

Message "0" will always be played at the beginning, this is typically some general introduction that every caller needs to hear. The other messages are played one after another, depending on availability. You can mask out messages with a service flag. We have added the text to speech in 69.3.6, so that you don't have to upload or record the prompts any more if the bot voice is good enough.

In that area there is/was the problem that DND was not updated most of the time when you e.g. search for an extension and the visibility of the element changes. We are addressing this in the next build (69.3.6). However calls seem to have worked in the 69.3.5 build, probably more reliable than DND but maybe not entirely reliable. Anyhow, as soon as 69.3.6 is available, please try that one.

The 69.4 version is the certified Teams version which was built some time ago. But we keep the 69.3 branch up to date, so thats why the build date is newer.

Ok should be updated now.

Well of course we recommend that the phone gets automatically provisioned so that we don't have to go through all sorts of settings and make sure that they are set the right way. If you manually configure the phone and use TCP or UDP, it would still use RTP/AVP but include a crypto line, which would be ignored by the PBX and the call should proceed normally. Some devices propose multipart/MIME with an RTP/AVP and another RTP/SAVP part in "automatic" mode, which is terribly confusing and that needs to be explicitly turned off but if I remember correctly, the GXP does not do that.

But that would be only a problem if you don't provision the phone with the updated template?

Hmm just checked the 68.0.39, it should really contain the T44 models. Maybe try to upgrade one more time?

What trunk provider is this? Maybe it makes sense to add a template. I would always set the country code in the trunk setting in case that it is being used from another country. Then the number presentation is helpful to understand inbound numbers and correctly convert them into the internal +-notation. For outbound there are many variables that can be put into the header settings (check the log for trunk on level 9 when you make an outbound call). Is this problem about inbound calls or outbound calls?

The Yealink T44 should be in 68.0.39.

You might have to set the minimum TLS version to 1.1 or 1.2. If you are running the PBX for a long time, that setting might be still from many years ago.

You cannot run the whole phone system on the smart phone. If you want to use the Android phone as a client, you would first have to install the PBX somewhere on a server, preferably in the public cloud so that you can reach it from anywhere with your smart phone.

The main difference between 68 and 69 is that SMS is now (mostly) in the tenant. When you upgrade, you might have to set the SMS provider up again. Apart from that, I would choose a non-trivial prefix for the SMS provider — this is essentially the password to post content to the PBX. The log should contain some clue what the PBX is doing with the incoming web request, I would turn on web server logging even if it is very noisy just to make sure that you see the request from Telnyx and when its clear that the PBX detects it as SMS, you can then reduce it to those log messages (e.g. SCRIPT). But there should be be a way to figure out what is going on looking at the logs.

I wanted to point out that we have arrived at 69.3.5. There are a lot of fixes and features, it's all in the release notes. Some highlights: Every call now summarizes the jitter and the packet loss of the RTCP packets in the call. This is an invaluable help in troubleshooting quality issues, especially when mobile devices are involved. In the tenant, click on the call record and then there click on CQDR. The SIP packets for the call are also available, which also makes troubleshooting easier. You can still enable PCAP if you want all details of the call. The queue management in the user front end now contains also a mode where you can overlay weeks and days, e.g. to better determine staffing hours. We have also increased the duration for keeping records for this without totally overloading the PBX. If you upgrade from an older version, the history might be a little short. The PBX now also involves an adaptive jitter buffer that is by default enabled only for the mobile devices. This is helping a lot to improve audio quality in difficult environments. You can now enable or disable every feature on the user front end from the admin level. For example, if you don't want to play the hangup tone, you can do this from the administrator now. We have added checking of call forward numbers, so that users don't accidentally forward to a number that does not exist and cause a lot of confusion. We have optimized the registration performance for most popular VoIP phones, so that you can now have several thousand registrations in a single tenant. This further improves the economies of scale for running the PBX in multi-tenant environments.