Vodia PBX

Guys, this is not in 3.2. It will be in a 3.3 build, which is not "officially" available yet.

If you are using TLS (which is the default) then you automatically get SRTP. Check if you see a "crypto" header in the INVITE/SDP.

There is such a software available I believe from Sangoma. Thanks to the SIP standard it can be used with the PBX.

Eight in thirty minutes is no reason for concern. Maybe the cable is a little bit instable. But the PBX will not drop a call because of that.

Nono. The PBX must find that out automatically, and it is possible. The routing table of the PBX make that possible. I believe in this current case there must be something wrong with the setup. We are trying to get a login and find out what the problem is.

The filenames are in the XML files in the messages directory. You can "grep" for the names if you want to know where they are referenced.

That means the PBX wanted to get a UDP packet from the OS, but that packet was not available any more. It can happen, but it should be rare.

Hard to say why the provider sends forbidden back. Are you sure you paid your bills? Interestingly, they seem to run pbxnsip as well. They even use the same version as you do!

May be a stupid question: Do you actually see the BYE message showing up on the PBX? Do you see it in the log file? Maybe the PBX is not aware the call disconnected yet.

The last time I saw a VAR dialling 911 and talking to the officer saying "Hi, we're just installing a new PBX and testing if it works. Have a nice day." No firetrucks showing up in front of the building.

"cell_dis" means "Display" and it can contain strings like "(978)746-2777". The cell phone number in "cell" is being used for making calls and it would typically contain "+19787462777" (if you use country code 1).

Try establishing an audio call first then switch the video on (while still talking). Essentially the PBX believes it is T.38. Fax and video are actually similar.

I still can't figure it out. It is hard to believe that would be a CentOS-specific problem. Last resort would be to make a strace of the interaction between the PBX and the kernel.

This will be in the GUI.

Did you try to list all possible ANI in the extension? I believe that the PBX will then try to stick to the ANI provided from the user-agent. For example: ANI: 9787462777 9787462778 9787462779

After leaving the message, does a new .msg file show up in the recordings directory? Maybe it is write protected?

The agent group has a couple of settings that redirect the call out of the queue ("when ringing for ..." and so on).

Sometimes it is also called "whisper mode". That is because when the teacher talkes too loud, the phone of the agent might have a little bit echo which would be audible by the customer. Maybe today the problem is not as big as in the good old analog times when there was no echo cancellation on a handset at all.

Now there is (version 3.3).

Top shows you what processes are active, how much memory is used and all kinds of other interesting stuff about the healthyness of the system. The command line is just "top" <enter>.

Well those templates are moving targets. But no secret. So feel free to take today's snapshot! snom_3xx.zip

Again very strange. Can you SSH in and run top?

Did you select HTTP in the server menu of the phone? It should not generate a TFTP link at all...

Oh that is the call volume. This is for adjusting the gain. This does cost a lot of CPU horsepower, you better turn this off if you don't need it. It is in the Log section of the admin mode.

For remote devices, you usually have to use HTTP because TFTP is not a very NAT-friendly protocol. See http://wiki.pbxnsip.com/index.php/Polycom#...ng_Provisioning on how to do that.