Vodia PBX

Danke, das Wiki sollte das jetzt so zeigen.

If you are using version 4.5 and you changed the admin password in the web interface of the PBX, then that password is also worth a try.

Contact the local support, and they will get you a manual activation key then. Also, if you can, upgrade to 4.5 Delta which has an improved HTTP client.

If you disable "camp on" in the domain, then the callback will not be available and it will skip the annoucement.

Not only is the softphone market difficult from a technical point of view, it is also cut throat. There is so much free competition out there in the SIP space and also in the VoIP space in general that is it difficult to make money. Counterpath is a public company, you can see how much they are turning over. In the SIP space, I think it is fair to say that Counterpath has one of the the best soft phones if not the best. Customization would help from a marketing perspective as it would look to the native customer that snom has its own soft phone; however a deep integration that takes advantage of the buttons or even things like visual voicemail, extension status indication and so on are not possible. Thats why we are working on the attendant console. If customers are interested in large device interoperability (iPhone, Android, PC, Linux, Mac, you name it), then Counterpath Bria is the better choice.

It should not make a difference. However, if you have a snom free license (and not the latest software version), you should enter an empty "manual" key, so that it would go to the right license server. After that, you should be able to acticate the code. Because the PBX keeps the entered code in a setting, changing the upper/lower case makes the PBX detect a change and re-load the key from the server. Again, this behavior was also changed/fixed in the latest version; so in the future this process should be easier.

The SIP From header has essentially two fields: The "display name" and the URI, similar to an email address. The routing is all about the URI, and what the phone displays is typically the display-name. The PBX typically does not mess with the display name, as it is really hard to say if the display name contains a name (lets say, in Japanese UTF-8 encoded) or something else valid. The only way is the address book; however in this case that does not sound like an option to me. The provider does say this call is from "40373722222", it would be better if that would be in a better readable format.

The username is "root" and the default password "nosoup4u". If port 8722 does not work, also try port 22 as newer versions are using this standard port.

So other calls are working stable? Are the phones registered all the time? You should have the system send out emails when the extensions looses registrations (in the registration tab for the extenson, there is a select box for this).

Actually in defense of 3CX and as I read the email, it does not mean it will not work; it just says that 3CX does not indent to spend time on supporting it. It is quite understandable, as supporting a cross-vendor interoperability can cost significant amount of time, and even if you would purchase a soft phone license for a few bucks, it would probably cost more money to support it than the revenues from the license. We would also not support the 3CX softphone like any other soft phone; however if the phone works we would not have a problem with it either. The soft phone world is difficult; not only the SIP interoperability is the problem: you also have to deal with a variety of platforms, hardware, network setup which all make it difficult to guarantee that you can make good phone calls.

The old SoHo units were running a seperate web server, and that one has its own version number. Just log into the PBX and do a software update only on the PBX. If oyu dont have it, you can make a software upgrade on the Linux level, log in using SSH and then cd into /usr/local/snomONE, then download the image, rename it to pbxctrl and restart the system.

The time is set on the operating system level. Which OS do you have?

Okay, seems like they present the number always in global format without leading plus. Then the pattern !([0-9]*)!+\1! might do the trick already. Use !([0-9]*)!+\1!50 if you want to default to 50 if the destination is not found.

Sometimes interoperability is not only about the service provider in general; sometimes it depends on the route. This is because of the SIP proxy model: The service provider might forward the request to another service provider, which has their own specific setup.

Well the log shows that there is some automatic recording on. That can fill up the hard disk quickly.The service should run "forever", practically until the next SW update.

Es dürfte kein Problem sein, CentOS 6 zu verwenden. Wir verwenden ja keine anderen Applikationen und das was wir an Libs verwenden ist seit vielen vielen Monden stabil. Es gibt keine direkte Anleitung für CentOS (bei Linux setzen wir wesentlich fundiertere Kentnisse voraus als z.B. bei Windows). Es gibt noch ein paar links aus den pbxnsip-Zeiten unter http://kiwi.pbxnsip.com/index.php/Installing_in_Linux für Linux, ich glaube die haben es noch nicht ins snom ONE wiki geschafft.

Hard to say, the log shows nothing that would explain it. I assume the file system is not full?

Yea, thats a tricky problem. The next wave coming out has that problem solved. I remember that you need to do something on the /etc/network/interfaces file; and you should use the latest/greatest firmware (4.5 Delta), which also makes sure that you cannot screw the IP configuration up so easily from the web interface. I would suggest to upgrade to that firmware version, then use the web interface to configure the IP address again, which will write the /etc/network/interface file, and after a reboot maybe the problem is solved.

Ehhh, maybe try to seperate the codes. I guess the PBX gets confused.

If you want to run the PBX in the cloud yourself, you can just go to a standard hosting provider. If you chose a dedicated appliance, you will get a good performance (jitter). With virutal appliance you have to be careful. I heared that e.g. EC2 from Amazon did a good job, but we did not measure the performance 24/7. We ran our PBX in a hosting center since 2006, our good old pbxnsip was always "in the cloud" (they did not call it that way by then). This makes a lot of sense if your users are spread out, and it also dramatically simplified the NAT setup.

Ja nicht abschiessen, sondern neu starten... Ja mit CentOS sollte man das Probem nachstellen können.

I think we are working on a code change that specifically sets the caller-ID of the extension when using DISA.

Unfortunately, the routing list is initialized only during the startup of the PBX... That script would right now have to restart the service. If you use the IP address to register devices from outside, all that registrations will get lost when the service provider changes the IP. That is not very beautiful either. Ongoing calls have the chance to stay up if the handsets are able to change the IP destination when the RTP stream suddenly comes from a different IP, but not all phones do that. Seems like your service provider is suggesting you to run the service in their cloud... Or maybe they just want to promote IPv6 . Maybe we need to add something that changes the settings on the fly, so when someone enters the IP routing list from the web interface it gets updated immediately. Regardless of this problem, this makes sense anyway.

ausprobieren

But it would be very easy to provision the parameters that you want into the phone--avoiding the need to log into the phone at all! Just set the parameters that you want to set on the PBX, and reboot the phone. That might be easier than trying to solve the riddle with the password. BTW this has one positive and wanted side effect: End users cannot screw the phone config up and then call you for help.