Vodia PBX

I think this is doable, however with a different setup. The buttons should trigger the click to dial feature of the PBX. The guards should simply calls the appropriate auto attendant or just a IVR node. The click to dial feature will first call the guard and then whatever number was provided (in this case, just the announcement). You will have to do multiple click to dial, one for each guard. For more information on how to initiate a call from a HTTP client, see http://vodia.com/documentation/click2dial The other simple thing you can use is the caller-ID feature when someone gets called. As long as the guards have a display they can see which button was pressed.

Ah. Wir haben auf der Server-Seite da etwas geändert. Der DND-Status muss auch bei dialog-State auf dem Telefon sichtbar sein; das ist was praktisch alle PBX machen und was die meisten Kunden auch erwarten. Da es bei dialog State nur wenig Informationen gibt ausser dass man an/aus/klingelt, wird dort "an" signalisiert.

In the domain settings there is under "Voicemail" an item called "Mailbox direct dial prefix" that should contain the "8". Change it to another number, e.g. "77" or just clear it if you don't need the feature.

Wir haben auch die Firmware auf dem Telefon auf den neuesten Stand gebracht. Vielleicht liegt es auch daran...

The "8" is a domain setting; by default it is the prefix to call the mailbox. You can change it, e.g. to "7". Generally speaking, it is better not to use any prefix for outbound calls. We are living in the year 2015 (soon 2016), everybody has a cell phone and you don't need to reserve a physical cable to make an outbound call any more...

The blacklisting happens either on IPv4 or IPv6. 0.0.0.0 is an IPv4 address whole :: is a IPv6 address. You can try to log in to ::1, which is the loopback interface address in IPv6. See for example https://en.wikipedia.org/wiki/Localhost and how to use it in the browser e.g. https://msdn.microsoft.com/en-us/library/windows/desktop/ms740593(v=vs.85).aspx or http://www.chromestory.com/2011/02/how-to-enable-ipv6-using-google-chrome/ If that is all too complicated for you, just delete the entry from the accesslist file and restart the service. It is not as elegant as using IPv6, but also works.

Well you can still try IPv6. Or just edit the accesslist folder and remote the XML file from there and restart the service.

5.3.2a is also available for CentOS32 now.

So far not much. There was a problem with conference rooms not showing the dial plan in the web interface, a problem with the new snom provisioning server, and some changed for the web interface for PnP. But the 5.3.2a will be a moving target as we work on features and fixes.

The missing records are generally no reason for concern. We have already added the new log to the latest version, it would be great if you can upgrade to 5.3.2a but before you do that please let us know what OS you are running. For the SMTP you can just send me a private message with the SMTP log (level 9). In that message you can also include the OS version.

LOL no that can really happen. For example, the number of CDR per account can be limited, and then the PBX will bit those "dents" into the index. The writing to a row that does not exist is a little bit more serious. Next version will include the table name so that we get a better idea which table is affected. Let me know if you want to have a 5.3.2a build (which OS), which is pretty much the same like 5.3.2 at the moment. The SMTP log messages are also a little concerning. Maybe you can get us a separate log for the email client (attachment is okay) so that we can see if there is anything serious.

It is a complicated topic. The extensions are a kind of internal address book as well, like the domain address book, the personal address book, the google contacts and the active sync contacts. And then we have some customers running the PBX essentially as switch, where the PBX should keep the alias name when it passes it down to the "phone" (which is in that case typically another PBX). And then there are settings on the phone how to display the name and number pair on the phone, which are different for each phone (might depend on the display size). From the PBX point of view, it should use the From and To-headers appropriately and then it is the phone's job to render it properly on the display. There is also a header called P-Asserted-Identity, that (if present) overrides the From-header in the display.

Well we still have CNAM on the to-do list. I guess we will re-visit the topic anyway soon and need to pay special attention to the address book then.

If the local browser on the same computer can get there then that is a good sign. You can double check if the DNS server in the PBX in the status screen makes sense. Older versions are checking for snomone.com, but newer versions are going to vodia.com. The snomone.com domain will eventually get out of service. If it all does not work, turn logging high for the web client in the PBX and for TLS, and then re-apply the license key. If you have a new computer, then you must first reset the license in the vodia.com portal, so that it can store the new computer "finger print".

That should be fixed in 5.3.2.

There is a setting on the domain that tells the PBX to keep the alias name or to pick the "canonical" name of the extension. Maybe this will help to send the ID that you want. Of course this also heavily depends on the firmware of the phone.

Usually this problem occurs when the PBX cannot reach the license server. This can be caused by the firewall or misconfigured DNS. When you change the server you need to reset the license at vodia.com, this is because CPE licenses are bound to a specific server.

What are the first 6 digits of the MAC? Cisco keeps adding more MAC, kind of hard to follow. We need that to filter out VoIP phones in the LAN, otherwise your new printer would show up in the list.

We don't automatically provision those phones. Also we don't have them here for testing. If you already have a phone, you can try to register them to the PBX, in theory it should work if the follow the SIP standard.

My first guess would be a problem with the dial plan of that extension. Can that 200 extension call the destination without any problems in a regular call? Also is this a different phone model? Maybe if this phone is not able to have two calls at the same time (one one hold) it might run out of resources. If you can't figure it out, get a PCAP trace for the extension 200 (http://vodia.com/documentation/pcap) so that we can take a look what is wrong.

There was a bug in 5.3.0 where you don't hear ringback when you call an extension through the auto attendant. It is already fixed and will be in the next release.

I am sure you have seen http://vodia.com/documentation/trunk_custom_headers already. The "display" name is the "name" of the person, in contrast to her "number". In SIP you put that in from of the less and greater than brackets, e.g. "{ext-display}" <sip:{ext-ani}@{domain};user=phone> Unfortunately practically all SIP trunk providers throw the display name away. This is because the PSTN was designed around a thing called CNAM, a concept so antique that it takes weeks (if ever) to propagate a change through the network. The system seems currently to collapse, as the software is so old that nobody is able to maintain it any more. Anyway, the PSTN is simply not able to transport your display name to the other side. On a side note there are a few companies trying to address the problem, e.g. opencnam.com or truecnam.com along with some other problems. Android has also started searching for telephone numbers on incoming calls to figure out who is calling, so you might want to check your Google+ profile again.

In that case I would leave the area code empty. It would mean that people always have to dial the full 10 digits; not sure how much of a big deal this is.

We could integrate that with the screen where you see devices plugged into the LAN.

Sounds to me like a problem with the quality of the DTMF tones. You should be able to see the DTMF in log level 6 in the IVR menu category.