Vodia PBX

Any chance to quick try to IE? Just for the same to find out if that is the problem.

It should not be empty! What browser are you using?

Hot desking had a bug in 3.1.1/3.2.0 with multiple domains. If you need this feature you must use one of the head builds (what OS?).

The new codec selection field is just a glorified JavaScript input form for the gool old list. If you look at the pbx.xml file, you will still see the old list of codec numbers. The change was made to make it easier to select the codec preference. If there is no codec selected (e.g. on the trunk) then that means that the system default should be used. Only in the admin/ports web page there must be at least one codec selected (because that is the default codec preference).

At the moment my only idea is to set up a second system with a three-minute key and then send the call there. There is some prepaid stuff coming into the system; that could be used to limit the call duration as well. That would have the benefit that every minute the caller can head a beep and on the last minute three beeps, so that he knows the call will be disconnected soon.

I would go to version 7.3.14. We are using it, and it seems to be working fine.

Whow! Nice catch!!!

If you can easily resolve this, get a Wireshark trace and see if there is anything on the network. Maybe a packet storm every 15 seconds. I assume it does not make a difference if the call is internal, to the mailbox or external? if the problem also exist for a IVR (mailbox) call we can exclude the jitter is coming from an outside source. Also, can you check the process list on that computer. Is there anything that has a priority above normal? Only processes above normal can give you that problem.

We will introduce a flag that will lock down the codec one it has been determined. Even at the price of later transcoding.

Oh oh. You are right. Will be fixed in the 3.3 version.

If the process does not have the "PROCESS_SET_INFORMATION" permission than the request will fail. Check out http://msdn.microsoft.com/en-us/library/ms684880(VS.85).aspx for more information...

The settings on the phone should not make the big difference. Can you get the SIP INVITE from the phones web trace log?

Yea that could be a hint.

It is really a pain if the PBX does not "own" a routable IP address. You can fiddle with the "IP Routing List" setting to try to undo the algorithm that the firewall does. Very support intensive. You probably need Wireshark to troubleshoot what is going on on the system.

Whow. I would not know how to do this right now. Maybe something like the call always goes to an auto attendant, which then sends the call to a calling card. Not very beautiful, as users always have to do two-stage dialling. I would check if hot desking is an option here. The office-hour user hot-desks to the public phone. When the user is not logged in there, it is just a regular no-permission phone.

Terrible noise is usually a problem with SRTP. There is a flag for the trunk where the PBX can send only 180 without SDP to the carrier. Some carriers cannot deal with early media!

Can you log into the box and check the /etc/init.d/pbxnsip file? is there a "--busy-det"? Take it out and reboot. Check if the problem is still there.

How long is the delay between the RTP packets? What is the exact time between the problems, 15 seconds? I assume the process is running with sufficient permissions and there is nothing in the log like "cound not set affinity mask".

There is no timer. But when you are "touching" the line (e.g. by listing it in the web interface or by making an outbound call) the line should be released. We did some changes for internal calls (someone seizes the line, then makes a call to another extension) and maybe within that scope we can periodically check if there is a line that should be released.

Good idea. It brings up the idea that the manual override of the automatic mode must be reviewed again. Because why resetting it at midnight? Maybe you want to reset it at noon, or 3 PM and so on. Then there is a thin line between automatic and manual mode.

We'll start a FreeBSD Forum.

Hmm. Works here... Also on version 7.3.14. Do you see "No permission for intercom to ..." in the log (level 5)?

The line stays seized if you make an internal call. You can unseize it after the call by pressing the button again or it should also get off after some time automatically (seize timeout). We are putting in that the line gets automatically unseized when the extension makes an internal call. As for intercom, there is a new settings for the permission to perform intercom. Now you cannot intercom to anyone and turn on the microphone! It was a kind of security leak.

Looks like too many digits to me. Probably you have to increase the number of digits in the string. For example, change the string "$w$5e$10c$5d" to something like "$w$5e$20c$5d". It is in the setting "cdr_format", check out http://wiki.pbxnsip.com/index.php/Global_Configuration_File for more information.

No idea how to do that (we are PBX experts, but not forum experts). Maybe just send an email to support@pbxnsip.com and we'll upload it...