Vodia PBX

See http://wiki.pbxnsip.com/index.php/Troubles...TMF_Not_Working. Maybe your ITSP is still not providing RFC 2833 DTMF.

I think you need to open a trouble ticket. From the forum it is difficult to track this issue down. There is a new page in the Wiki that describes the process: http://wiki.pbxnsip.com/index.php/Trouble_Ticket_Processing.

Hmm. We really don't do anything special. And we are using this every day (can't live without it any more). Maybe you have to put a "1" in from of the cell phone number (Callcentric expects numbers to be in the 1xxxxxxxxxx style). Also, there is a feature in Callcentric that limits the number of calls, maybe that stops you from making another call.

Sure, we have a big interest in talking to OCS. We need to act like a endpoint, because IMHO the most intesting case are branch offices or departments that run the PBX locally, but want to be part of the whole organization. I think an endpoint always has to authenticate itself. Maybe one day Microsoft will also support Digest authentication, as all other vendors do and then it will probably no problem. Kerberos might in fact be easier, it is documented well.

Unfortunately, OCS uses the "NTLM" authentication scheme. The problem is that there is no documentation on how to use it and Microsoft did not tell us yet... It is clearly on the radar, but technically at the moment a big problem.

The the second pattern is the problem, see http://wiki.pbxnsip.com/index.php/Dial_Plan. Well, it might be tricky to get star codes through the PBX. Try the pattern "9\*([0-9]*).*". Another method would be using the replacement pattern to generate a star. For example, make the prefix 99 replace a star: Pattern: 99* Replacement: sip:\*\1@\r

I guess you should open a trouble ticket on OTRS. Please ask km@pbxnsip.com for an account.

If it says "please hold the line" that means the redirection after timeout kicks in (not the cell phone feature). Then it is probably a problem of the dial plan that this call does not get through or the call really does not connect. Check your redirection settings...

We also took a snapshot of the user manual from the Wiki. It is available at http://www.pbxnsip.com/download/user_manual_203.pdf.

Check out http://wiki.pbxnsip.com/index.php/Microsoft_Exchange.

When I reply, I get a "Attachments" section with a UPLOAD button. You should also see that option.

The SIP part looks beautiful. You may try http://www.pbxnsip.com/download/pbxctrl-2.0.4.1759.exe to if it makes a difference. I remember there was a bug fix in 2.0.4 in the media relay part, maybe it makes a difference. P.S. You can also attach PCAP files for message posts (use the UPLOAD button).

Hmmm. Agreed, we need moderator a PIN. Next version.

You can also use external gateways. Just add another trunk. For example, you can add another trunk for an ITSP, and also for another FXO gateway.

We had that proposal before (can't find it, though). The problem is that it is generally easier to search by topic. But maybe we can have a bug section and once that it is not a bug any more we can move the topic to the right forum. Or vice versa.

Ouch. Maybe the bug is fixed by now, but could be we don't have a stable build with that yet.

Well, there is some text in http://wiki.pbxnsip.com/index.php/General_User_Settings. It is more or less a front-end to the PnP mechanism, so that the phone gets the settings automatically. Not very sophisticated yet. We are dreaming about phone-dependend provisioning profiles...

Well, the pbx by default asks the phone to use pool.ntp.org for getting the time. If you don't like this address, you can change it in the pbx.xml file (restart required). Maybe the 2nd phone cannot resolve the DNS address.

Do you have the SIP packets from the log? Would be interesting to see what is wrong.

Well, NAT is a very difficult topic. I have seen routers that allocate ports by the fifo-principle with maximum of 32 ports (and keep them open forever). Until I figured it out, it was driving me nuts. Therefore, if you have another router, it is worth a try and see if it changes the behavior.

Good question. In an ad-hoc conference there is no moderator. I am not sure if it is a good idea to allow everyone to kick everyone else out.

Yes, the system presses the "E" key when the playback finished.

Well, the IVR node should do that job: http://wiki.pbxnsip.com/index.php/IVR_Node. You can even do more with it.

In the latest version (2.0.3) the moderator can terminate the conference with *9. See http://wiki.pbxnsip.com/index.php/Scheduling_Conferences and http://wiki.pbxnsip.com/index.php/Conferencing.

Currently we are using the following files for version 6 of the snom phones (see attachment). You can put them into the html directory and edit them. Be careful when there are updates - your changes then might become obsolete (that's why we prefer not to give them out). The files for 320 and 360 are the same, therefore there is only a snom 320 file. You can always put files into the tftp directory. They override the files generated by the PBX. snom300_6.txt snom320_6.txt snom3xx_6.txt