Vodia PBX

This would be http://wiki.snomone.com/index.php?title=Call_Detail_Records_(CDRs), however it really does not say anything about the $r and so on. http://kiwi.pbxnsip.com/index.php/Recording still has it.

We could use the type "Redirect" from the button profile from this. If the user selects "Redirect" and the parameter is a ACD, then we can light the LED up when the redirection is in place. Would that be a solution? Edit: That would not be a solution. The destination right now is the redirection target.

What about Spanish? Maybe the problem is with the Spanish, not the German.

Yes. The SoHo form factor is at least in my understanding not what what you would expect from a 24/7 device that is critical for a company. Therefore, we redesigned it and put it into a snom plastic. Should be available in a few months. But the same software like the SoHo.

You can always create a symbolic link in the working directory of the PBX pointing to any location you want (even on the network).

I guess the problem is the space in the path. You can as well just use "$r/$d/$t-$i-$u-$n.wav" or "recording/$r/$d/$t-$i-$u-$n.wav" ($r stands for recording).

Thats 2.7 GB on disk. 32 bit can address 4GB (in real life 3 GB because of VM allocation). Are you sure you need so many CDR? Maybe make the "Keep CDR duration" shorter, e.g. only one week. You can just move the directory if you want to archive them; you dont have to delete them.

You mean, on a LED button? I am not aware about this feature... If we add this, we would either have it on the ACD button in some special blinking state (hard to read) or we would have to use a seperate button for this. Hmm, not sure... There are so many other buttons that come to my mind if we do this. What about email? If we send an email out in this event, that would be looking more clean to me and it would have the advantage that you can easily document this event.

Ouch. Thanks for the update.

Unfortunately we dont have a SPA here.

Consider using a SoHo (or one of the upcoming minis) instead. This saves you a lot of power (=money) and there are no moving parts. That should be possible. The easiest solution would be to assign telephone alias numbers to the account that should be called (e.g. "40 97812345678 2121234567"). For that you will need two base stations unless the two business locations are actually pretty close (DECT). Also, you must make sure that the packets between the locations can be routed. NAT is your enemy here. Either consider running the PBX on a public (=routable) IP address or use a VPN setup. Hmm. Not sure if the m9 supports that. From the PBX perspective, you can assign distintive ringing in groups, e.g. hunt groups. Maybe the regular voicemail timeout will already do for you. If you assign extensions to one business unit, then it might become easier than thinking that an extension is a hardware device. You can have multiple extensions on one physical device.

I am thinking on how we can get this done the easiest way... Conferencing now allows invitations... We have the call back on busy features for internal extensions... But the web callback is something that really makes sense in so many cases. We need to make this very easy to use, like the scheduling of a page playback that we have recently added. Then you can just put some code in PHP or whatever you are using to get this done. Maybe we can use a agent group to automate finding an available person to do the call.

If you hear that then that means the carrier plays the message. You can put a replacement into the dial plan that looks like this: 1*

Keep in mind we added in 4.5 the possibility to schedule a playback by opening a HTTP URL on the PBX (no SIP involved). This is ideal for a button on the door that is able to trigger a HTTP GET.

If you have the gateway registered with a fixed and known IP address, what you can do is set up a gateway trunk to the address and just route calls over the gateway trunk. The registration to the extension then would be there just to keep the gateway happy. Then you can use the standard approach of the gateway trunk to address those problems. Hopefully the gateway will accept the SIP traffic that will come from the PBX.

Excellent.

Yes, also blue and yellow. You need to re-apply the activation code, though. Plug and play will not work with GS phones, through.

You can register it to an extension. Then in the dial plan, you can select the option to send the call to an extension. We have seen that with other products before. It makes sense if the device operates behind a NAT, e.g. a local E911 breakout (although it does not help to improve stability for 911 to put a device into a nonroutable location that depends on the availability of free NAT table entries of a router). The SIP problem with the registration is that the PBX cannot use the Request-URI for telling the device where to send the call. It must use the To-Header. This is very problematic in the cases of redirection. Fortunately, in version 4.5 we are now very flexible to put about anything into the To-header, so that this should not be an operational issue any more. And PSTN does not present the display name & other stuff anyway.

Nee, es gibt inzwischen die Version 4.5. Wo ist denn noch die 4.2 Version?! Das sollte eigentlich inzwischen alles auf 4.5 umgestellt sein.

Ich würde das Problem lösen indem ich mal den Log Level auf 9 setze bei allem was dafür in Frage kommt und mit die eingehenden Muster anschaue. Z.B. könnte es sein, dass die mit einem + anfangen oder dass es nicht immer 11 Stellen sind.

I only tried from home over the Internet. That was no problem. Maybe inside the office will be a problem if the firewall has problems with hairpinning. A split-brain DNS setup could solve that problem.

The numbers above are just for statistics, and obviously ran over the 32-bit limit. Seems this is something we need to fix, but it is not critical. I would also suspect the CDR table is very full (14 days could mean in total 500 K CDR in memory). But there could be also other reasons outside of the PBX process, e.g. make sure that there are no Windows services waking up e.g. perform Windows updates at offline hours or make sure that there is no virus scanner performing a hard disk scan when the office is in peak time.

So finally I was able to try it out. I installed sipdroid (which was not installed yet), and essentially followed the setup for sipdroid. It was working within a minute. The biggest problem was telling the device to keep the WiFi connection alive, so that it would be possible to call the device even after the display has been turned off. So now I have a WiFi SIP phone for essentially 130 USD, and it even has a camera for scanning bar codes (apps make it happen).

It essentially means that the third party registration limitation has widely been removed. Soft phones are now allowed to transfer calls!

Ich würde auf jeden Fall 4.5 nehmen, denn vorher wurden nur ein paar Registrierungen von Drittanbietern bei der Free Edition erlaubt. Wenn im Log von der PBX nichts auftaucht, gibt es aber noch andere Probleme, z.B. könnte die IP Adresse auf der schwarzen Liste gelandet sein oder das Outbound Proxy bei den Telefonen stimmt nicht.