Vodia PBX

It is generally not a good idea to ring all agents "immediately" because this creates a major hickup in the system, sometimes even in the network. IMHO gradually calling all agents is not so bad.

In the hosted environment, CO-licenses should not be the problem. But the good news is that we could just use the trunk allocation mechanism to limit the number of calls in that domain.

What version are you running? Could be a bug that was fixed in the latest & greatest.

That sounds like the PBX provides a contact in the SIP invite that cannot be routed in the phone. Does the Contact header contain something that the phone can reach? Maybe you can post the INVITE (from the PBX log) here and check if the IP address in the contact header is what you expect.

It will be included in 3.1.

It gets straight to the point and says "978-746-2777 is trying to reach you. Press 1". That has the benefit that you hear the original caller-ID (solving a common problem when redirecting the call) and you hear it right at the beginning.

:-S Yea we had the same issue with the calendar document. Seems it is something with the certificate and the https paranoia. Maybe you need to get a thousand dollar certificate before your browser eats this........

Are you getting anything? What does the web browser say? Are you using https?

Try "save as..." and then open it with a text editor.

At the moment the meaning is "call", not call legs or "trunk calls". We had this discussion recently, and the result is not clear yet...

We have introduced a new setting for the trunk that tells the PBX hoe to represent a international telephone number (+ sign in the beginning, 011, 00, and so on). It is still not 100 % stable. Not sure if you want to try that one out.

Do you have control over the Exchange server? Maybe you can try turning TLS on, or check what authentication types you can turn on there. Last resort would be to just trust the IP address of the PBX and don't use a username/password at all.

Well, it will definively take time. There is no concrete plan available yet. In the meantime the workaround is to partition the userbase and run several processes on a multi-core. A server farm on a single chip.

Are you recording all calls or just the calls to that extension? Because after a redirection, the call technically does not go to the extension any more.

Well, you have to use SMTP. I know it works with the currently published version, there are several installations talking to Exchange with SMTP. But I am not the big Exchange expert, on the PBX side you just need to point the SMTP server to the Exchange server (that part is easy ).

Hmmm. There is a setting "max_loop" (default should be 10). Maybe that is the problem here?

Well, what we could do is having a link in the files dom_ext*.htm for prev and next.

Oh, do you have a hangup timeout on the auto attendant? Maybe it is still active from the first AA and then kicks in when the PBX is back in AA state?

yea. But we need to split up the table, in order to make it possible to index it. This will be a little bit messy, that'S why we don't have it yet. But it is noted.

The text "Service Flag" is misleading (does not say anything). We'll rename it to "Specify time when system calls the cell phone", hopefully that makes it easier to understand what the PBX does.

You can also load it through the webinterface (admin/configuration at the bottom). Then you don't have to restart.

Okay, that sounds reproducable. We'll track this topic and post when there is a fix available.

Ops, will be fixed in the next build.

AFAIK Aastra does not support "buttons", only "dialog" state (sometimes calls "BLF"). With that CO-line emulation is not possible. The only thing that you can do is use star codes, e.g. to park and pickup calls.

Works for me... I assume you are using a direct destintion for the "1"?