Vodia PBX

Just updated http://wiki.pbxnsip.com/index.php/Counterpath.

No, this is just about the HTTP/HTTPS socket. You can limit the access by choosing password that the end user does not know. Ouch, don't do that on 3.1.2! But it will work on 3.2.

You can also bind the http server only to your private IP address. In the port settings, put something like "192.168.1.2:80" and then it will bind only to that port... A new version will be available today.

Yea looks good to me... If you need numbers below 100, it is also no problem. Not sure about negative numbers, but 0-99 are definitevely okay.

Yea, that was a design decision ("feature"). Star usually means "clear", this is the way people can correct their input. And pound usually means "enter", which e.g. is useful when customers are using extensions with variable length.

You mean because the phone gets idle? That sounds like a feature to me. IMHO the whole hype around presence is completely over. I understand IM is useful, but I never understood what presence should be good for. My productivy dropped like a stone when people saw that I became online. Hiding my presence is a feature!

Ah, you are probably a victim of the Windows file locking mechanism. Seems like we have to Open and Close the file all the time...

Yea, that is the intention. When you perform a attended transfer to the cell phone, you will be able to hear if the other person picks up or the mailbox. There are other situations (e.g. when the PBX call the cell phone because there is a new voicemail message), but in all these cases the user has to interact with the PBX before the interaction happens.

Yes you can. All you need to do is use a symbolic link to another destination. We start to support SQL also from the PBX. Send an email to support@pbxnsip.com to get more infos on that.

Now that you have Xlite working, try presence! The presence works with the address book. As far as I remember the trick was that the presence is not peer to peer, it runs the the PBX as agent.

Nonono. Audio is always sent over UDP. TCP has huge delay if a packet gets lost. Instead of repeating a lost packet (and letting all other packets wait) you better just play back a little click and then the audio will go on.

Are you using failover in the trunk settings? There was another post today about that topic, see http://forum.pbxnsip.com/index.php?showtopic=2039. Maybe that solves the problem.

Well, that is just a speed dial. IMHO there is no reason to offer pressing "1" here, you might really want to hit the owner's mailbox. And you know that you are calling the cell phone number, even if you might not know the number itself.

Yea, we removed it. If you run version 3.2, you can just put "file:cdr.txt" into the SOAP CDR URL field and you'll get nicely formatted files in the file system.

We'll soon come out with a version 3.2, which does fix the problem.

Okay... From the IVR node, you should also be able to send the call to 8xxx, where xxx is the extension number. Anyway, AA is easier.

Apart from using the same preference number twice the dial plan looks okay to me. Which one first only plays a role if a entry in the lower part of the dial plan should not rbe reached because of a entry in the upper part (defining the exception first and then handling other cases). But usually that is a feature - for example if you have a special trunk for calls to New Zealand put that first, and if it fails you go to a more expensive route that also works for New Zealand.

If you put "1#" into the direct destinations of the AA it will wait for a timeout.

The trunks also have a setting that tell the PBX how to present a number. Depending on the version, that even works for inbound calls. YOu might have to upgrade to version 3.2 to get that working.

Is using a prefix in the trunk an option? In R-O-W companies usually have a common company prefix and then append the extension number there. The other thing is that you can tell the PBX not to change the To/From headers. I believe that is in the domain settings.

Ehh - you mean "IVR Node"? Or auto attendant IVR tab?

The purpose on the "all error codes" is that the PBX can differentiate between the gateway itself being busy or the destination itself busy (talking). In SIP the gateway is supposed to send a 5xx code if it has no more channels available; but there are gateways out there which send 486 (destination is busy/talking). If you gateway does it right and sends a 5xx code when all gateway channels are occupied, you should select that it fails over only on 5xx calls. If it is one of those buggy ones, then you have to failover on all codes. The other failover is timeout. You can control with the timeout setting on the trunk how many seconds the PBX should wait before it generates a 408 response. This will also trigger the failover case as if the gateway would send a 408 code. When the failover happens, the PBX resumes the processing of the dial plan with the next higher number in the plan. You should avoid using the same preference number in the dialplan as it makes this processing kind of random. When the PBX resumes the processing, it will again look for matches. The next line that matches will be executed then. This line can also failover and the game continues until either there is a result (e.g. the call connects) or the dial plan has no more lines.

You mean putting a "8" in front of the extension, so that the call goes directly to voicemail?

Yea, that is a good old problem. If you have only one domain, make sure that you have the name "localhost" either as primary name or as one of the alias.

No, on 2.1 you can put the names into two fields. In version 2 we had "primary name" and "alias" names. The primary name could be "010" and the alias names "9787462777 9787462778" (seperated by space).