Vodia PBX

No, there is a new code for that. The PBX does not automatically assign a code if you just upgrade the system. Check the feature codes of your domain, there is something new called "Forward calls to domain accounts". Pick something that is not assigned yet (e.g. *51). Then you can either just call *51 and the PBX will change all auto attendants, hunt groups and agent groups of the domain. Or use *51xxx is you want to change just a specific attendants, hunt groups or agent groups. Listen to the annoucements. You can either set the redirection or clear it.

There are several dial plans in a system. The phone has one and the PBX has another one. Obviously, the phone dial plan depends on the model. The PBX supports the automatic provisioning for dial plans; this is a setting in the domain. However, only a few phones are provisioned with that. IMHO it is easy and okay to tell users that they have to press the dial button on the phone. Everyone knows it from the cell phone today and seem to be able to get over it. It also has the advantage that users don't have to use those stupid prefixes like "9" to make an outbound call. A usable dial plan for the phone is only possible in the USA (NANPA) and only if you don't intend to make an international call. Otherwise you have to work with timeouts (another difficult and unpredictable topic) or ask users to hit the enter button after entering the number. The same discussion existed in the SIP world outside of the NANPA area (so called "rest of the world", where the other 95 % of people are staying). ISDN offers "overlap dialling", where the network tells the phone when the number is complete. To cut the long story short: That feature did not make it into SIP; this feature ends on the PSTN gateway. The dial plan on the PBX is only used for trunks. If you want to call an extension, you don't need to put that into the dial plan. If you have only one trunk, then the dial plan can be very very simple. Just put a "*" into the pattern and select the trunk. That's it! The default configuration already has it.

Well, how long did you wait? There should be a timeout eventually. May be five minutes.

There is some documentation on the Wiki in the dial plan section (http://wiki.pbxnsip.com/index.php/Dial_Plan), look for "Try Loopback".

At the moment you can do that with the IVR node. You can use the From-pattern there to send some calls into one direction and all other calls into another direction. The PBX will use the IVR node account only for a "millisecond", there will be no audio played back. Not very convenient, but a possible option to address the problem.

It would be great if we can get a PCAP (Wireshark) trace of this when it happens. Maybe there is a problem with the jitter, or DTMF. We are shooting in the dark right now when it comes to this...

We have added that code to version 3.4. You can just dial a new star code and then enter the destination number. You can do that (as with all star codes) also from your mailbox, so that you can use this feature also from the outside world. A release should be available soon. If you want to try it out, send an email to support@pbxnsip.com and tell us what operating system you need.

We have put it into the latest code. Send an email to support@pbxnsip.com and mention your operating system. Then we'll give you a snapshot of the latest version ("leading edge"). We like to get some feedback on this before we make it generally available.

The PBX is a PBX... We don't plan to change it's nature. Anyway, as long as your clients wants to have a fixed DID/ANI per extension (which would be a trunk for him), it should be easy. The session border controller makes it a pretty useful setup, as you don't have to buy another external one. However, on the billing side you will have to invest something. You can usually use the CDR from the PBX, as long as the customer does not tra to transfer calls. Maybe it would be useful to turn off transfers so that the trunk is a "dumb" trunk just like you get it from the big switches today.

This is because a sales guy insisted that a "good" product must have a hard wired IP address on the "LAN" port... The interesting part begins when someone sells two CS410 into the same LAN.

As far as DTMF is involved the fix is in the latest branch 3.3 build. Could there be any other source for this problem? Does this problem also occur for calls between an extension and the mailbox (internal user leaving a voicemail)? Maybe it makes sense to Wireshark the traffic between the PSTN gateway and the PBX as well.

Whow. Interesting findings! Additionally, it seems there was a mess up between IPv4 and IPv6 in the DNS resolution of the NTP (seems to be limited to the CS410 PBX). The next build will also fix that.

Well, from the PBX point of view if should use the PBX extension caller-ID when you cann from your cell phone into the system and place an outbound call. And when the PBX forks a call to the cell phone, the cell phone should see the original caller-ID. Especially the latter case is difficult with most service providers. Usually the serviec providers will not allow the PBX to present any other caller-ID than the caller-ID of the subscriber. The PBX uses the prorietary header "Related-Call-ID" to indicate that the use of a foreign Caller-ID is valid and the carrier can present the Caller-ID of the original call.

When you enter the number and blur the input field, does the web client show the number in the format (xxx)xxx-xxxx?

Do you get midnight emails? Depending on your time zone the midnight events may come at a time when you would not expect them.

You may need to set the country code in the domain so that the PBX can convert the numbers into the right format.

Those who like to try a beta build out please send a private message to support@pbxnsip.com and please mention what OS you are using.

Weird. Did you get that riddle solved?

There must be something wrong with the setup. If there is no registration and there is no redirection (or cell phone forking) then the mailbox should defintively pick the call up. It would be beautiful to send the voicemail by email, so that they don't have to wait the whole week until they get their voicemails!

Hehe. From a technical point of view, that is actually a feature. A tricky one. It is not so easy to resume a call in another context in the middle of a playback. You can avoid this problem by using the blind transfer into voicemail. Because the VM will always pickup, this blind transfer is save and there is no need for an attended transfer.

Or just upload an empty file. Workaround! Or a picture that looks just like a shadow...

Well, it seems that the user rejected the call. That is defintively a feature!

What you will get is a CDR "fragment" for each segment of the call. Because it is impossible to summarize the whole call when going from one segment to another, the PBX generates a CDR record for each segment. That's the idea behind those internal CDR. In order to make the Auto Attendant look better, it pretends that the original destination was the extension. That part is debatable. However, we still have this option weather the PBX should change the From/To-fields. If you turn it off, then chances are that the original From/To fields make it through this CDR chain.

In the later versions we actually have something. Officially it will be a 4.0 feature, but AFAIK it is also in 3.3. You can use it this way: When you are logged in as system administrator, enter a URL into the web browser that looks like this: http://pbx/reg_status.htm?save=save&reboot=next this will reboot the system on the next occation when there is no call active. If you want to reboot the system at midnight, you can use this one: http://pbx/reg_status.htm?save=save&reboot=wait What the PBX does is simply terminate the process. Then the external monitoring script has to take care about the restart of the PBX. In Windows, the service manager has to do that job. In Linux, MaxOS and FreeBSD you will have to write your own little dirty supervisor program that automatically restarts the PBX after it terminated.

you mean the setting cannot be changed any more? Or do you not see the groups when the user logs in?