Vodia PBX

Well we actually did a lot of changes to the Polycom phones (webinar is coming up). I would suggest that you check out the latest template and compare it with yours. Or maybe you can just provision one Polycom through PnP and then see how it works now.

You might have to re-provision the phones so that they pick up the new setup?

It is in 61.0 (check out https://vodia.com/doc/releasenotes610)

Well but at least they could check if the callback option is offered?

Are they using the callback function? Maybe it would be easier to just remove the callback number as soon as it has been used once.

Whow you have a very proactive team there! Marking it at the beginning of the read out would only solve half of the problem - the calculation if the message is available is done the voice rendering is started with the introduction prompts. Marking it already there would be IMHO too early because there is really a chance that the message will not be delivered. Is there anything that can be done there on the organizational side?

There are two ways for sharing: Making a copy into others mailbox where they can listen to it like it was their own mailbox; and linking it to other peoples mailboxes where they are essentially accessing the same message. So I guess you want to use the later one?

Multicast is a special way to transport messages in the Internet. In the LAN this is mapped to a broad cast, but if you want to send messages out into different segments you need to program your router accordingly. Hint: This is not trivial. But there is plenty information about multicast in the Internet that can help you getting this done.

That SIP password is not available as a variable. But with the web password the user can log in and change the SIP password.

Got it now. Right now there is no variable for that, but you could hack something e.g. by using a name scheme for the alias. For example, if your domain has the name abcdef.com the alias could be failover-abcdef.com, which you can then reference with failover-{domain}.

{domain}?

For a simple external call redirection you can use the IVR node, especially the ActionURL there should be useful. As long as the PBX does not have to listen to the users voice input this could be a way to handle things in a simple way.

For now the solution is to run this inside iOS in the Chrome browser.

Aha I was not aware that private people cannot use TLS on DTAG... At plain sight I agree it would make sense to fail over. However the whole RFC 3262 area is very tricky; I would at this point say it is easier to explicitly specify the transport protocol.

Ehhhhhhhhhhhhhh so going back to UDP works for you? That would answer the question if your IP address works, but it would still make me wonder why TLS does not work. And if UDP works, maybe TCP also works (which is a lot more stable than UDP).

I am pretty sure that you can register only from a IP address that is on the Deutsche Telekom network. Maybe if you have the ticket open with them ask them if the address you are coming from is okay.

If you want to use the trunk from Deutsche Telekom you have to come from a IP address that belongs to Deutsche Telekom. For example, trying this out from here (Boston) will not work - we would also get a 408 error.

Ehh you are you on a IP address that belongs to Deutsche Telekom? Otherwise you will not be able to use their service as far as I know...

Well that is not good... IMHO it should all end up at tls:217.0.3.228:5061 or tls:217.0.3.244:5061. Maybe you also need to turn the SIP logging on, so that we can see how it steps through the DNS records. You are on a recent version, right?

If this is a certificate related topic, try turning TLS log level to 9 then there should be a message that makes this clear.

Maybe a simple TLS problem? Does your PBX trust "T-TeleSec GlobalRoot Class 2"? This is the Root CA that was used to sign the connection to "h2-eps-608.edns.t-ipnet.de".

No just watching the soccer world sup LoL... The question is if this is always not working now or on/off. If it is permanent it would be interesting to see what is in the DNS cache of the PBX (web interface) and if this makes any sense. If this is on/off there must be something else, maybe even a unstable Ethernet connector or something like that.

Try using host instead of dig, there you can also specify the type (e.g. host -t SRV). There were several outages last week on our VoIP phone in Germany (PBX not involved) where the phone would not register. Maybe this is just because too many people are on vacation in Deutsche Telekom right now... Hopefully nothing has changed in the setup on their side!

I would turn on logging for the web client to see the traffic between the PBX and SRAPS.

MWI was added back and will be in the next builds (60.3+). This build will then also fix the Mailbox retry setting (thanks for reporting).