Vodia PBX

Yes. See http://wiki.pbxnsip.com/index.php/Auto_Att...t#Night_Service. A dirty, but flexible trick!

I tried it here. For me it works - of course... However, I was using snom version 7.3.14 and there all SUBSCRIBE messages now show Expires: 0. Needed to change that to 3600, then it was much better.

What phone type? What firmware?

I would just use one auto attendant and 24 service flags. One falg for each hour. Then use the night mode feature to distribute the call to each cell phone: Service Flag Account: 7100 7101 7102 7103 ... Night Service Number: 9787462777 9787462778 9787462779 9787462780 ... You can even make exceptions for the weekend and for the holidays!

No, the blue line is the number of call legs. Usually one call equals two call legs. The red line is the number of calls, and in that graph the maximum was three calls.

You can try to set the ANI for the trunk and choose the mode "No Indication" for Remote Party/Privacy Indication.

Hmm. What if we just introduce a setting that says "maximum number of connected calls" in a agent group? In the above example, set it to "1" and then that poor guy who is on duty can work without phones ringing around him.

TFTP is UDP, telnet is TCP. Check out "netstat" from the command line to see what ports the PBX uses.

Do you get a admin email about the disconnect? That would be a clear sign that the PBX thinks the call dropped. Also, check if the AudioCodes has VAD turned on (turn it off if that should be on).

Full CO-lines are not triggering the failover... Only a full PSTN gateway can trigger that trough a SIP response (ideally, a 5xx code).

It sends an email? I would not know about that nor how to turn this off.

Which config file are you using to get the phone up and running? What firmware version?

Well, you need to know the IP address of the PBX server. Check out http://wiki.pbxnsip.com/index.php/Login for more information.

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.