Vodia PBX

Well, the communication between OCS and the PBX currently runs through the mediation server, which is like a PSTN gateway (but translates OCS SIP to plain SIP). If you want to ring communicators that are registered on OCS - that sounds difficult to me. It would probably be much easier to register the communicator clients on the PBX. But I am not the expert here, check out http://wiki.pbxnsip.com/index.php/Office_C...ications_Server. Gateway: If they are talking SIP I am pretty sure it will work. PSTN gateways are easy to get working.

That sounds like a problem with the assignment of a dial plan to the extension. The domain has a default dial plan which applies to all extensions, check if it is set. Alternatively, you can also check the dial plan that has been assigned specifically for that extension. If you can call other extensions, that means that there is no problem with NAT.

One port is probably for the PBX and the other one for the PHP. You can find out what ports are being used by which application with the netstat command.

If you can afford to restart the PBX you can also change the files on the file system. Then you can avoid the whole SOAP trouble.

Well obviously there is inband DTMF detection going on, but it seems the volume is too low to have a reliable DTMF detection ("Whisper DTMF"). Maybe whoever is sending the DTMF should increase the volume.

Maybe you can just use a name like this: First name: Joe W. Last name: Blow (Company)

I also don't 100 % understand the problem. You mean that you need a third field in the extension that represents the company name?

At the moment that is really a problem. Most of the call centers that I have seen just use a projector on the wall that shows the web page - from the admin login. The PAC will solve the problem in a proper way.

It is a significant jump from one domain to multiple domains. You need to be more precise with the naming. The important part is on the phone: Outbound proxy: It is not really neccessary to set up DNS for that, you can use the IP address. You can also put the same outbound proxy in different domains, e.g. "pbxserver1.hosting-company.com". But of course, it is much more flexible to use real DNS names for the outbound proxy, for example "client1.hosting-company.com" or "pbx.client1.com". Maybe it is time to look into DNS SRV records. Domain: There you have to be strict. The domain must string-match the domain name on the PBX. It is the only chance for the PBX to find out where the request should be sent to. My suggestion is to use the outbound proxy all the time. It makes features like call a missed call from the phone much easier, the phone will not try to bypass the PBX and go direct. Also, you should not use the name "localhost" any more (unless you want to "catch" requests that are not going to any of the local domains).

You don't have to configure anything, just make sure that the routing on the host is okay. The PBX can deal with any number of interfaces.

There is something in http://wiki.pbxnsip.com/index.php/Outbound_Calls_on_Trunk. I guess you have to try RFC3325 and the other modes, and for the outgoing caller-ID just leave it empty. But I agree, the whole caller-ID presentation is a pain in the neck. I wish there would have been a clear standard from the beginning.

Oh you mean you want to use analog as "trunk"? Back to the roots? Then that should just be a matter of the dial plan!

What did you put in there? What version are you running?

I think the easiest is to just try it out. You can just edit the ringtones.xml file and set the Alert-Info header as you like. All you need it to edit the file and load it through the web interface (admin/settings/configuration at the bottom). <?xml version="1.0"?> <ringtones> <tone name="custom1"> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom2"> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom3"> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="custom4"> <vendor type="alert-info"><http://127.0.0.1/Bellcore-dr4></vendor> </tone> <tone name="internal" type="internal"> <vendor type="alert-info"><{to-uri}></vendor> </tone> <tone name="external" type="external"> <vendor type="alert-info"><sip:1234567@test.com></vendor> </tone> <tone name="intercom" type="intercom"> <vendor type="call-info"><{from-uri}>;answer-after=0</vendor> </tone> </ringtones> Try an internal or external call and you should see the Alert-Info header set in the INVITE. You can modufy the ringtones.xml file on your own and see how the INVITE changes. Maybe we don't haveto change anything in the PBX to support this feature.

Okay, but then we are talking about the called number indication. The time between the first ring and the second ring on FXO is used to send additional information, e.g. the time/date, the caller-ID and it can also indicate the called-party (see DDN, for example http://www.nmscommunications.com/manuals/6709-13/appc.htm). That information is in SIP carried usually in the To-header, not in the Alert-Info header. If you are using a ATA, check the options to send the called number on the FXS using the MDMF format.

Not yet. How would you tell the other side to disconnect the call? The only chance is to wait until the other side hangs up, then we could go back to the first menu.

There are two important settings. One is the domain name and the other one the outbound proxy. For domain identification, the PBX uses the domain name. The phone uses the outbound proxy for routing purposes. If you keep these two things seperate, it should work fine. And if you are using plug and play, it should also work fine.

The 3-party is today a mainstream feature of the SIP phone. There are some phones that are able to have more than 3 parties in a conference (for example, the snom 200 was able to do that). Then this is no problem. If you want to transfer the participants into a PBX conference, you must blind transfer the participants into the conference room. If there is no PIN everything is easy. However, if you want to use a PIN things get complicated. AFAIK Polycom supports a button that blind transfers all calls into a conference room, that seems to be the mainstream way of moving a conference to the PBX. The big question is authentication and overlap avoidance. You don't want to accidentially bump into another conference held by another extension.

The PBX already supports distinctive ringing, for example in a hunt group you can select that DR you want to use. There is a XML file ringtones.xml (see http://forum.pbxnsip.com/index.php?showtop...ed&pid=2246). I don't understand how you want to use DR to make the FAX selection. I also don't understand why you want to use the same extension number both for FAX and voice.

What you can do is to define the "mailbox escape account" on either domain or extension level. If you point it to an auto attendant, callers can go there after leaving a VM message (after pressing #). Then they can dial whatever they like.

The logo has changed during the renovations for 3.0. The new logo should have 250 x 50 pixel, with transparency.

That means the PBX is supposed to send out RTP, but does not know (yet) what codec to use. It can happen when the codec negotiation takes longer. If it still appears when the call is already connected then it is a sign that something went wrong with the codec negotiation. We took this message out in later versions, it is not really neccessary and it cost a lot of performance.

Only if you register the trunk to the other PBX. Then from the PBX on public IP's perspective, that registration is just a regular extension.

Did you see http://wiki.pbxnsip.com/index.php/Outbound_Calls_on_Trunk? 2.1 changes the ANI presentation, supposed to be an improvement.

That number fits into the switched telecom world. In IP world it is impossible to present a reliable number, as there are so many factors that influence this: Is the UA using TLS? What kind of challenge mechanism is used? How many codecs are bring offered? Is the mini-session border controller involved? Does the Pc have hardware support for IP? Is VPN being used for Internet traffic? Is there UDP fragmentation? What OS is being used? How big are the routing tables? How much cache memory does the CPU have? And so on. There is a page on the Wiki that touches the topic (http://wiki.pbxnsip.com/index.php/Hardware_Requirements), but you will not get a satisfactory answer. An I think looking at the above question, it is really not very serious guaranteeing a realistic number. BTW there is a setting that limits the BHCA on UDP transport layer. This is just to protect the server against DoS. It is a little bit like asking "How many pages can you print with Microsoft Word per second" - it also depends...