Vodia PBX

You can also manually specify the port in the command line with --http-port nnn and --https-port nnn. This helps to get the service up and running when it is not possible to kill the web server that is occupying port 80.

Hmm Anyway glad to hear it works now.

If you are using both Ethernet interfaces make sure that you have only one IP gateway set. This is a problem if you are using DHCP on eth0 and static IP on eth2. In that case you can solve the problem by using statc IP on both ports or by telling the DHCP server not to provide the default IP gateway for the MAC address of eth0. But you are right, PING must work, otherwise it is hopeless.

Maybe take a look at http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk. Depending on your service provider, you probably find the destination in the To-header. In your case, maybe a pattern like "!([0-9]*)!\1!t!301" make sense.

It is possible to blind transfer participants into the conference. At the moment the ugly part is that the participants have to enter the PIN code for the conference after they are transferred.

Well, because the PBX is media-aware session timer are not neccessary. What happens if the call gets esblished? Do they reject the call?

Try clearing the browsers cache, or just F5.

Try changing the RFC3325 mode on the trunk. Depending on the mode (gateway/register) and the presentation you may get different results.

What was the previous version?

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.