Vodia PBX

What kind of conference are we talking here? Three-way conference is the business of the phone; if they want more participants they must use the conference server - that setup works completely different (they need to dial into the conference).

Yes, that means that if offers only G.729 and OCS does not accept that. You must also offer other codecs like G.711.

Oh you should upgrade at least to 3.2. The 3.0.1 probably does not detect Turkish prompts yet. 3.2 even contrains a Turkush web interface!

Oh, are you using MAC? Grrrrr. MAC does not have /proc/net/arp... The PBX now needs to find out what MAC is what IP address. That little part is missing in the BSD implementation yet!

Here is a build: http://pbxnsip.com/protect/pbxctrl-suse10-3.3.0.3163.

You mean that you see a directory audio_tr with the WAV files in it, but you cannot see the option in the web interface? WHat version are you running? Last resort is to rename it to audio_sp and select "Spanish"...

Sooner or later they have to write some shell scripts that automate their processes. curl can be of great help, as most of the day to day steps can be done through the web interface. But other jobs like managing DNS obviously needs to be done outside of the PBX itself. The big question is if they want to give the customers access to the self-adminstration web interface or they want to do this through a set of scripts that use SOAP to change certain settings on the PBX.

You mean on blind transfer? It is a kind of feature, because it tell the caller that the other side is now ringing and someone is (hopefully) running to the phone. But it is an interesting point - the caller does not really care if it is a blind transfer or attended transfer. Maybe we can make this a setting. I believe the current behavior is fine as default behavior, but maybe someone wants to play music instead.

The SDP offer is in the INVITE packet. It should be just "above" the trace you showed.

Usually the problem is that the 3.2 version tries to be smart and interpret numbers if you set the country code. One question is how many digits you are using for national calls (10/11 digits). Then in the dial plan, in USA you must use 10 digits always. If you use the old pattern 1* you probably run into problems.

If you are using the 3160 build, then just move the timezones.xml files out of the way. We added the Russian translations for the time zones, should be looking better in the next build!

Currently we are at http://www.pbxnsip.com/protect/pbxctrl-3.3.0.3160.exe

Yes, that option is obsolete now. Now you just need to trust the MAC on the cable, this is now in the PnP tab.

T.38 is "not trivial"... If the PBX would have to advertize it after detecting the CNG signal, it would be responsible for converting G.711 into T.38 . I totally agree, the whole situation is not very satisfactory. I believe almost the whole VoIP industry agrees... Modems were designed for analog lines, not for digital RTP packets.

Static and calls dropping are usually a sign of network problems. I guess the PBX is not in the same LAN? Maybe the connection is a little bit instable. Can you see who is sending the BYE message? Do you see a Reason header that explains why the call was disconnected?

http://pbxnsip.com/protect/pbxctrl-rhes4-3.3.0.3160 is the latest & greatest.

Fingers crossed... I believe so!

If you like, please try http://pbxnsip.com/protect/pbxctrl-3.3.0.3160.exe. That might also contain additional fixes for RU.

Are you using a country code? That might be a problem; and you can also check on how the PBX represents global numbers on the trunk. OCS likes the "+" notation, at least before R2. A look at the SIP trace should help finding the problem.

Yea, that's a bug we already fixed in 3.3. http://wiki.pbxnsip.com/index.php/Release_...#Call_Recording

On the M3,under "Management settings", set the "Configuration Address" to the IPv4-Address of the PBX. If you use a later version of PBX 3.3, then you will even get the right time zone .

There is a global settings called cellphone_timeout. The default is 20 seconds. You can change it in the Global Configuration File, see http://wiki.pbxnsip.com/index.php/Global_Configuration_File.

The translation is missing? Any change to try an upgrade? If so, what operating system?

This topic pops up from time to time. The PBX does not change the volume. We could change the volume during the recording, that would not be such a biggie. But IMHO having problems with the volume is a clear sign that there is something wrong with the gain of connected devices and it should be corrected there.

In version 3.3, we are doing some neccessary changes. The good old MAC-address based config only works if the phones are in the LAN and the server can see the MAC address on the cable (layer 2). Otherwise, you have to provide the username/password in the phone (see http://wiki.pbxnsip.com/index.php/Snom#Set...name.2FPassword). Probably that hanging is because of this. Also, consider using the latest 3.3.0.3160 version, http://pbxnsip.com/protect/pbxctrl-3.3.0.3160.exe.