Vodia PBX
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If you can reproduce this, it would be useful to get a PCAP trace for this and see what is going on.
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Is this on version 68? Can you attach a screenshot?
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Ok looks like this was something that did not happen in Safari, but in other browsers. We'll fix the in the next build — if you want to try it its in 69.2.4.beta.
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Well the question was 15 years ago LOL. The PBX has the "virtual private assistant" for that the recognizes the caller-ID with the cell phone number of an extension, which is convenient and reasonable safe, especially when e.g. STIR/SHAKEN is at play to make sure that the caller-ID is not spoofed.
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1 hour ago, Matevz said:
- when clicking on History tab, it is always a day behind, unless I open it after 12PM; we are NZ Time zone and this started happening after v69
That one probably is really because of the time zone. We'll fix that in the next build.
As for the popup, we'll probably have to use it ourselves and see what is happening...
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Would this be a question for the snom forum?
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We have moved the push server back to the old servers for now — the problem is that we need the new push servers for the new Android app which is also supposed to work with 68. We are checking if we need to enable more ciphers on the new push server.
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Are you still getting the emergency call notification?! IMHO this should have been resolved in the Windows app and also with the 69.2.3 build.
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We'll have a 69.2.3 soon, we'll include the MacOS this time...
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There is a setting "Follow RTP" (in reg_rtpsettings.htm) for that. I guess this one is turned on? Then the PBX should follow the RTP. You should see a message like "Tuning to new SSRC from xxx" (log level 5, media).
The message "Source address for sip:xxxxx has changed from tls:192.168.x.y:39992 to tls:80.226.x.x:43321" is not related to RTP, this is about the registration, it is actually pretty useless if you have mobile device that change the IP address all the time.
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There is actually a new Vodia Android app in the process of getting released, we'll find out how to get it.
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Oh maybe this is because you are using the
8prefix for the mailbox? Does the pattern!E!30!work? It should not be!E!830!. If you want to use the timeout, it should be!T!30!. -
Well you could probably also use an auto attendant for that.
Anyhow if you use an IVR node, the cryptical DTMF list can have a
Tentry (for timeout), like!T!815!and the duration for the timeout is in the "Timeout (secs)" setting. The E entry is used for end-of-audio. Also, make sure that you use a single space as separator for the cryptical list. -
There are release notes for Version 68. As for the beta, we don't publish release notes.
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Well the PCAP show that the number that you have dialed does not exist (removed the PCAP to hide the domain name and IP addresses)... This is not related to port forwarding. You might have to set up a SIP trunk and dial plan to get that working.
As for the media connectivity, just dial *97 to see if you can hear the mailbox announcements.
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In such cases it makes sense to turn PCAP on and take a look at the packets. Even they are not really from the network interface they give you an impression what the PBX feels about RTP and where its coming from.
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Sounds like something simple... Maybe can you include a screenshot of where you set it?
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If you want your users to login, you should make sure that the PBX can present a valid certificate for the domain. More and more browsers will make life hard entering passwords and sending them in clear text to a server does not even have a valid certificate.
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Interesting — it will be interesting how Microsoft feels about adding such apps. On the one hand, it makes the Teams platform stronger, on the other hand it might take revenue away.
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As for the emergency we are about to release 69.2.2 where this seems to have been resolved. Also, we'll have a review of the Windows app coming up this week where wie will look into the tel-scheme issue and why this seems to be so hard.
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We want to use OPUS for the mobile users because it just sounds better, especially when there is a lot of packet loss. But G711 is still the predominant reality in the world of SIP trunks. I. would say if it works, keep it on G711 but true happiness will require that we have OPUS on mobile devices. In any way, I would try to keep a PCAP for analysis reasons to see why there was one way audio.
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Do you see the file
pbxctrlandpbxctrl.datin the/pbxdirectory? What if youmkdir /pbx cd /pbx ./install-mini3.sh
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The screenshot shows the right location. If you don't put the polling address there, it will use the address that it learns when it checks the license server which includes the IP address from where the request was coming from. So if the IP address does not change, you should be good.
If the calls connect properly thats a sign that the signaling works. You can take a look at the SIP call messages and the SDP attachments. if they contain the public IP address, the PBX is doing it right and then the question is if the ports are properly exposed. You can try turning on PCAP for an extension and e.g. call the mailbox with *97 and then check the RTP in the PCAP.
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Even Linux kernels gave up on that one? Anyhow the 68.0.37 is good for production. For later builds, we just include the name beta in the build number to make it better understandable.
Windows app updating account settings
in Windows App
Posted
Well 68 is not a masterpiece when it comes to frontend-design (in case you wonder why we would start a new front end project). The first save button is for the image.
Yes, the second save button would be for the user data.
Maybe we can still add a horizontal line to 68 to make this a little more clear.