Vodia PBX

Yea, it has a (simple) LDAP server built-in essentially because that's the only way you can have an external address book on the snom phones.

Well what is your dial plan?

The settings are reasonable. The failover should work, but it will take 15 seconds before it will kick in. We talked about the failover when not registered and it seems it makes sense to put that in as well so that the PBX does not even try the DSL trunk.

The problem is that the PBX reads the XML files only during bootup and then after that does not care if someone changes these files. Notifications cannot be done through the file system in the PBX (it would be crazy expensive if the PBX has to always check if the file has changed). Notifications can be sent in general through SOAP, and the PBX has a interface that makes it possible to perform changes in the internal database. Those changes BTW would then be comitted to the file system right away. There is a documentation available on this subject at http://kiwi.pbxnsip.com/index.php/Access_to_the_Database if you consider going this avenue.

Nice! Thanks.

If you want to make a speech first there is little choice you must use paging. If you use multicast paging it will be limited to the LAN (and to snom phones), but it will be very CPU friendly. Then whoever wants to respond needs to call you back the regular way. I think I hear this in the supermarket a lot ("extension 12 please call 43"). If you dont need voice in the beginning you can use a hunt group--if we are talking about a handful of extensions. Don't do this with 20 extensions, this will not only create a major CPU hickup, it will also create a packet storm on the network. The CPU power goes somewhere!

Das war generell. Um auf den Webbrowser des Telefons zu kommen muss man das Domänen-Admin-Passwort eintippen (ein Passwort für alle Telefone weil sich kein Admin für jedes Telefon das Passwort merken will).

Registration and failover are two different things, we can say they are practically not related. The failover does not care if the registration is active or not, it is only about the question if the INVITE gets any response within the failover time. This is because registrations can be active for hours and the PBX then has no chance to see if the internet went down or not.

Wir haben diese trivialen Passwörter aufgegeben. Mag zwar sein dass die bequem sind, aber das sihc einfach tickende Zeitbomben. Daher werden jetzt beim erstmaligen installieren zufällige Passörter gewählt. Die können beim automatischen Provisionieren durchaus beibehalten werden; beim manuellen Provisionieren sollten sie aber auf jeden Fall neu gesetzt werden.

There might be a problem with the way the numbers are represented. 00331234 is not 01234... But there was another post with a similar problem, so we are checking this anyway.

I remember it was a decision that the snom 300 display is so small that it does not make sense to have the address book there like on the 370. Not 100 % sure, but you can check the snom_300.xml and the snom_370.xml file in the web interface (customization area) for differences and then change it.

For that you would have to use SOAP to talk to the internal database. This will be some "investment" (work) to get this working. We used to have some PHP scripts on the old wiki (now kiwi), not sure if you want to go this avenue. The alternative is to have the PBX send the voicemail out by email and then process them automatically. That might be easier to set up.

You can load the 2nd greeting (with the number 0). Then the PBX will read that out tentatively while the phones are ringing.

Is this a transfer with early media (the call not connected yet)? This is tricky in SIP. If you can get us a PCAP or a LOG with the SIP messages (attach it, please) we can find out what the problem is.

Which file is missing? The 821 pretty much provisions like the 820, it should be only the first file that is missing the rest is like 820. Check out the generated directory which files are there.

Check out this link: http://forum.pbxnsip.com/index.php?/topic/3228-skype-for-sip-and-pbxnsip/, it is not baby step but seems to be straightforward...

There was a post how the PBX identifies the trunk (http://forum.pbxnsip.com/index.php?/topic/4034-inbound-calls/), if you have several trunks check if maybe another trunk matches with a higher precedence. In version 4, we introduces the explicit inbound IP address list in the trunk (at the bottom), this might help to tell the PBX directly where it can expect traffic from. That might solve the problem without extensive research.

We will probably provide a new software build by the end of this week, which will include stuff like the missin 821 support.

There are options on the phones (at least snom phones) where you can say what they should display. That might be a workaround. What you could also try is to use the address book to match the numbers to names that the PBX will then put into the display name.

Well, this is a endless topic, especially when redirecting the calls. Every provider does it differently. In the next major release, we will offer text-programmable settings for this problem, so that you can put anything in the headers in question. For version 3, the only thing you can do is to "play" with the header presentation setting for the trunk (RFC3325, No-Display, and so on) and see which one works.

If you are talking about the web interface's way to indicate that a file has been uploaded thats a different topic. It will always display the empty field. The question is if the file has been uploaded and you can play it back.

Okay, schon mal einen Schritt weiter. Wenn der Provider 404 Not Found liefert ist die Nummer offerbar im falschen Format. Einfach mal im Rufschema im Replacement-Teil sowas wie 004920123456 eintragen (eine existierende Nummer) und sehen ob die angerufen wird. Vielleicht will der Provider die Nummer immer im 0049-Format. Andere Möglichkeiten sind +49 oder 49.

Soltle so aussehen (in der Text-Darstellung): 100;SIP;;*;; Es sollte auch im Log ersichtlich sein dass der Ruf wirklich auf der Leitung "SIP" rausgeht. Notfalls nochmal überprüfen ob auch das richtige Rufschema in der Domäne und bei der Nebenstelle eingerichtet wurde. Wenn das alles nicht hilft, gibt es vermutlich doch Ärger mit dem Service Provider. In diesem Fall im Trunk bei "CLIP Standard/Anzeige von Nummern" auf "Keine Anzeige" stellen.

Ist im Dialplan (Rufschema) der Trunk richtig zugewiesen?

You can provision also multiple handsets. For this you can assign the MAC into multiple extensions. If the multicast PnP should not work, you can use the IP address of the PBX and put it into the Provisioning Server setting (network section in the web interface of the m9). If that does not work you probably have a problem with the certificates (what does the PBX say in the certificates who it trusts?).