Vodia PBX

Whow. Do you have the SIP messages in the log for that?

No the password policy has nothing to do with that (it only happens as JavaScript in the web browser). Hard to say what went wrong. Maybe the snom wizard kicked in and entered a new password.

If the alias list contains a tel:alias, the PBX will use it as ANI for outbound calls. For example: Primary: 4566 Alias: tel:2121234567

Yes. 0xB8 is usually a good choice, that is why it is default.

Okay, we put that into the head version, will be in the upcoming releases of 3.x.

Well, they can hit *0, the star is just being ignored though.

The rewind only works on the message itself... When you are listening to the envelope, there is no stop or rewind.

If there is nothing set, the duration is limited to the maximum duration in the system (reg_settings.htm), usually ten minutes.

Oh, in 3.0 you must put the relative path there (img/main_logo_blank.gif), not the absolsute path.

In Windows, you need to change the registry to enable the tagging of RTP packets for the Type of Service. There is a article from Microsoft on this topic: http://support.microsoft.com/kb/248611/en-us (see http://wiki.pbxnsip.com/index.php/Installing_in_Windows).

Whow. Actually it is no big problem to put that in. Is GMT always fine or does the phone expect "local" time (wherever the phone is)?

Well the phone thinks that this is DoS. It wants you to send the request from the PBX IP address and port. There is a option to turn this off on the phone, but then the phone gets vulnerable for simple DoS. Same thing for TCP. You can set the phone to accept TCP connections (which is off by default), but then again you risk that someone plays DoS with the phone in your network.

When a SIP INVITE comes in, the PBX needs to make a decision if that call comes from a trunk or an extension. The outbound proxy of the trunk is used for that - if it is missing then it is wide open and matches any incoming request. The PBX later assigns the call to an extension if the authentication is okay, but until then it is treated like a trunk call. So it is easy, just set the outbound proxy and give it a try.

Counterpath had a softphone, I saw it running on a Samsung cell phone.

Check if you have a trunk without outbound proxy. That can make the PBX believe that the call comes in on a trunk, and not from the phone.

Yea, the timestamp is the timestamp of the email. Even if there are a few seconds delay between the recording and the email time stamp, sobody seems to have a problem with it.

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).