Vodia PBX

Okay, we added more forums in a couple of other languages. Lets see how it goes!

I have seen it with my own eyes working... The trouble is I have no idea about configuring Asterisk (I guess that is okay looking at my name).

Did you try using an alias name? Alternatively, using a ANI?

Okay, here is a update. OID 14 and 15 are still a mystery to me, because they expect the trunk number behind the OID number. No idea on how to express this in this language. PBXnSIP_SNMP_mib.txt

Hmm. Could be that the PBX ignores the star code prefix and then monitors the park orbit status; but not sure yet. Maybe you can give it a quick shot.

What you can do is create and address book entry and set the telephone number to the park code. Kind of workabound...

When it comes to one-way audio check out this one: https://www.pbxnsipsupport.com/index.php?_m...ratingconfirm=1 It contains a small list of things that you can verify quickly.

I thing this is the page you are looking for: https://www.pbxnsipsupport.com/index.php?_m...ratingconfirm=1

I think in Asterisk you have to start editing the "dial plan" (which has in Asterisk a totally different meaning that in the pbxnsip world--it is more like a programming language with a lot of \\ and $$ in it).

I know a few ... Do you think we should start a German-speaking forum?

I would use netstat to see what is going on. AFAIK netstat is also available on Mac. There you should see if the PBX still has the UDP socket 5060 open and if there are any unprocessed messages in that socket. Does the Mac have a "personal firewall"? This is usually the source for major problems like there in Windows...

If you are using domains on different servers but one of them contains the name "localhost" it might be very confusing the the PBX. This is because it thinks that the extension of the one PBX belongs actually to the other. The workaround for this is to use (long with more than 5 digits) telephone numbers that clearly indicate that the number is not an extension. Just pretent that one PBX is calling over the PSTN network into the other PBX. Then that PBX knows the call comes from a trunk, not an extension.

It is already a couple of years ago, but I remember there was also a online service available. It was not for free; it cost a couple of cents for every so and so much lookups.

We need something that put the input from the IVR node in the CMC field of the current call. That would also be very useful before someone enters a queue.

I don't understand the question....

If the conference should be recorded, this has to be setup right from the beginning (scheduled conference). Otherwise if this is a spontaeous conference, the question is where the recording should go... Lecture Mode, okay that is something that would be worth adding. We have a feature that sends out an email to the moderator which lists who is in the conference (that's *1). That implies the count. Dialling out on other participants implies the question who pays for that call. If the moderator should pay, then he can as well call out and do a blind transfer into the conference. Preferrably, he can use an attended transfer, because then he can also verify it was not the mailbox picking up. But there are a few more feature neccessary for the conference setup. No question about that.

There are many buggy RFC2833 (now RFC4733) implementations. I assume that when you look into the wireshark either the two "2" are glued together or there is no DTMF at all. Sometimes providers signal they they would support out of band in the SDP, then when the media arrives it is inband.

AFAIK tis is not possible yet with any released version. It is a pitty. You have to use dialog state with in and the PBX usually rejects that as DoS (12 subscriptons per device).

We have done some massive changes with the CDR in version 4, especially when it comes to redirection. In version 4, there are CDR available from the carrier's perspective. It does not matter if the call went to an auto attendant, then got put through to a assistant and finally redirected to an external number. For the trunk call, it would all be the same (like your carrier would bill you). If you can, wait for version 4 and don't invest too much into version 3.

Linux reports the view from the OS, while pbxnsip reports the "inside" feeling. It is okay if those numbers are different. The PBX just tells you the ratio of waiting to processing media.

Does the PBX send a NOTIFY to the phone?

It is done on the fly. And yes, it takes performance. Something like 2 MIPS per call. For a modern CPU it is "okay" if you don't go to the limited. OTOH looking a TB hard drives today, compressing makes only limited sense...

Should be easy. Just register the Asterisk to the PBX like you would register Asterisk to your service provider.

Well, the core problem is that the service provider does not want to present any caller-ID. Otherwise you could easily indicate that the call comes from the White House... We introduced a "hack" for this, where we show the caller-ID of the original call with a new header: Related-Call-ID: j3j45234j52@34.2.3.43 Then the service provider correlates that ID and "clears" the indication of the caller-ID. That is a pretty clean way of solving the problem. So far AFAIK only callcentric supports it.

Yea agreed. However programming this stuff is "non trivial". Also, it is not so much the focus on the PBX to design complicated call flows. For that, there are special applications available that for example process VoiceXML.