Vodia PBX

You should definitively mention that the caller ID contains that string. I think that will tell them immediately what the problem is. We introduced the PnP for gateways not too long ago; I am afraid in Germany you are the among the first use this new feature... Other users used the Patton and manual configuration before with good results; so there is light at the end of the tunnel.

When we set up the Patton template, we used a US gateway in analog mode. Patton probably has more parameters for ISDN, including the one that we are looking for in this post. The German Patton support has the reputation to be exceptional; maybe you can contact them and get a clue. Or if another user of the forum from Germany can answer to this post and post the (working) configuration we can try to figure out what is missing in the template and add it.

Oh you are right. The documentation is incomplete. But you can see how to POST or PUT something by watching the network traffic in the browser (e.g. Chromium has a nice debug windows for that). That will get you started right away.

Entries? In general, sure. REST is for reading, writing, deleting and other stuff.

The routing list is a global parameter just like anything else (ip_routing_list). Just changing its value will have this effect.

I see two possibilities for this. The first one uses a DNS address which must be updated by the router or some other program outside, then the replacement list would be 192.168.1.0/255.255.255.0/192.168.1.10 0.0.0.0/0.0.0.0/pbx123.dyndns.org and this would also require that all devices are able to resolve DNS addresses properly (could be a problem in the SDP, not sure). The other possibility would be to have an external program use a REST call in the PBX to dynamically change the setting for the routing table. The PBX is able to deal with such changes on the fly; however it would require some shell script running on the PBX to get that done. If the PBX is running on Linux, this could be a bash script that is polling some what-is-my-ip-address web site or the router to figure out what address to use and then use another curl call to tell the PBX what the routing table should look like.

Of course you can use a comma in the "display name" part of a SIP header, and also in other places. However I agree, there are lots of devices and services out there which have a problem processing it. For example, if you encode it correctly, CSV files can handle it properly but i many cases the names are just concatenated with commas, and then the CSV breaks. If you want to have more fun, try the following names: Martin "M" Meyer <script>alert("hello");</script>

Well we need to properly escape it in the CSV (do we?) For example "account,with,comma",123,456. Where is it being used?

If you want to run the PBX in the LAN, register phones in the LAN and use a SIP trunk to make outside calls, this scenario should be still okay and no major problems (if the service provider uses a SBC). However if you actually want to register phones this environment must be called "hostile". In theory you can try to fiddle with dyndns, but at the end of the day you are facing a long uphill battle with instable registrations, dropping phones calls and so on. Unless this is a hobby installation, I would stay away from it.

Well then it should be really be the right language. There are still some more places where you can set the language for the call, e.g. in the auto attendant, conference and in the domain also depending on the country code of the caller. But I guess you did not do that or checked it. I guess the only way to track this problem down would be to get a log when it goes wrong. If you can reproduce it, then it should be easy to get that log.

Could be that the PBX tries to send back a web page saying "okay your call is being started" which has content that requires a session. Again, this would not be serious; it just would not look as pretty as it could.

It means that the PBX needs a HTTP session in order to complete a request. As far as I can see this is not related to a phone call. This can happen when a session expires, e.g. a user has not done anything for a longer time and then requests something that would require a still valid session. IMHO not very serious, although the log level is pretty low.

Can you check those extensions for what they have set as language? And what is the default in the domain?

Okay we have imported the old Greek text into the V5 database (we switched to a real database in V5 instead of the unfortunate CSV format we used before). If you log in to your PBX and activate "edit translation" you should be able to pull the translations down from the cloud. If you give us your activation code (private email please) we can also add you to the list of contributors, so that future builds will have it right.

You can control the language on system, domain and extension level, from general to specific. In addition to that, the Greek prompts were recorded a long time ago, and some of the newer prompts might not be available. What you can do is to copy the greek prompts into a tmp directory, copy the US-English prompts into the Greek directory and then copy the Greek over again. Then if a Greek prompt is unavailable, it will pick the English one; better than dead air.

For inbound you need to enable/allow this. This is a security feature of the browser.

Call from phone means that the PBX will use a SIP phone to start the call (through third party call control). Call from browser means that it will use the web browser to make the phone call; this currently works only with Chrome.

Instead of exporting single domains you better copy the working directory to the new server. Exporting a single domain wasn't always 100 % bug-free, for example exporting global phone numbers was a challenge. If you want to split a server up, you can copy the working directory to two servers and then delete the domains that are not needed on that server. If you are using snom phones, you might want to obtain one of those XML-RPC passwords from snom, so that you can automatically redirect the phones based on their MAC to the new server, no matter what has been set up on the phones. Otherwise the question is how the phones were provisioned. If they were provisioned with a DNS address for the setting server, it will work to change the DNS address. You might have to open up the MAC addresses on the new server for provisioning (make the duration e.g. one day, not just ten minutes), so that the phones can provision even if they don't have the password for provisioning (or the client certificate which not all snom phones have). If you want to keep the firmware, you can change the snom_3xx_fs.xml and take out the links (leave the XML body, though). Then the phones will load that file, but not detect any firmware URL and stay with the current firmware. I would recommend in the beginning of the migration; later you can undo these changes and have phones upgrade to the L&G firmware. The other "cliff" you might experience with the upgrade are your trunks. If you have complex trunk settings (SIP headers), you might have to tweak them again to get this going properly. And of course set aside some off-line time when you want to do the migration (don't do it while all users are busy). You might also want to let the Vodia support know about the time frame for a stand-by, just in case...

Yes the web interface is not the same like the audio prompts. Actually you can have one language installed while the other is not. Good news is that for customers the audio prompts are usually more important than the web interface

Sure. Please open a ticket on the support site or just send an email to support (the license information should not be posted publicly )

I would say that speed dial (which it boils down to) is a lot easier than the button programming. You should first manually set a phone up and make sure that it works, then you can go to the settings page and see the related settings and copy that over into the provisioning template in the PBX. Then all of your phones will get it with the next provisioning cycle.

That would mean that you can ave only one call. A blind transfer to the park star code would be better. But did you think about using shared lines? I guess that is essentially what we are talking about. Technically, you will hold the call, but it can be picked up from another phone.

In V5, when you log in as a user, you can make phone and receive calls using the web browser.

Did you try the web interface of the phone? There you can define what a key does. I cant take a look at this from here right now, but I remember that practically any key can be mapped to a specific function. E.g. you could map it to a speed dial. From the PBX point of view, apart from provisioning the phone, there is nothing we can do about how the phone treats key presses. This is handled locally on the phone. A hold in SIP is something fundamentally different than a park.

There is no more free edition for version 4. If you want the free edition, you have to use version 5.