Vodia PBX

The latest version at least has the option to include the FXO gateway log in the PBX log. Maybe it makes sense to write those messages also to the file system (embedded system, so be careful!), they should include FXO messages.

There are two ways to detect hangup. Impedance change (which seems to be used in the USA) and polarity change. Detecting the first is easy, the second is detected when the first time there is a polarity change. There is no setting (neccessary) for that. Unfortunately, in the analog world, things are sometimes instable. In our office, we have had Verizon lines which must have been very very long. After switching to Comcast "Digital" (which is also FXO, but only a few feet long) things looked much better and there was no more problem with DTMF, hangup and so on.

If you use plug and play, then the PBX will provision a action URL that retrieves the address book in XML when you press the address book button.

On the CS410, we have the nice problem that the input is analog - that means in addition to the DTMF tone, we have a reduced signal and added noise. That makes DTMF detection not easier. Maybe try 1024 and 2048... The error rate will probably also increase. I guess you can imagine that I am not a big fan of analog lines... Especially if they are long!

I get one email per call now, that does the job for me. Outlook does the job or sorting the emails into the right folder. The beauty of this solution is that is always up-to-date (no need to wait until midnight) and it is very easy for the PBX (no huge emails at midnight).

Hmmm..... Could be..... You need tel: alias for inbound matching? That could be a problem... Do you have several domains? If that is not so, then you could just use regular alias names without tel.

Oh then you just need to put that DID into the "Trunk DID" of the trunk. Did you try that?

Try setting RFC3325 on the trunk. That should solve the problem, Broadvox is usually quite well on being standard compliant.

As discussed over the last few weeks, there is a new 2.1.6 available. The release notes are as usual available on the Wiki at http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.6. The primary goal of this version was to increase stability and usuability, for example we removed some mouse traps like including tftp in the backup. The following versions are available: http://www.pbxnsip.com/download/pbx2.1.6.2446.exe (complete InstallShield) http://www.pbxnsip.com/download/pbxctrl-2.1.6.2446.exe (bare executable) http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.6.2446 (CS410) http://www.pbxnsip.com/download/pbxctrl-debian4.0-2.1.6.2446 (Debian 4.0) http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.6.2446 (RedHat ES4) http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.6.2446 (SuSE10) Unfortunately, we had to move from Debian 3.1 to Debian 4.0, but people said that 3.1 was quite old anyway and it seems that the new image also runs on the old platform. However, of course we recommend to upgrade to Debian 4.0 as well. Please let us know if there is any problem with the new build.

If you have a FAX, check out the latest image http://www.pbxnsip.com/protect/pbxctrl-2.1.6.2443.exe. Set a direct destination of "F" to your FAX extension in the auto attendant and see if the PBX redirects the call to the FAX device. I don't have a FAX here... Obviously I was able to survive without it!

Well the PBX runs the Goertzel algorithm (see http://en.wikipedia.org/wiki/Goertzel_algorithm), as most other applications that look at only a few lines in the spectrum. By default, the algorithm is in floating point. Then the threshold is easy to dertmine. Obviously we are fighting problems with fixed point arithmetic, and in order to solve this we just "amplify" the signal by the value dtmf_gain / 256. If you have really loud signals, there will be distortion - but it seems that DTMF is always relatively silent so that danger is not too high. BTW we are also looking at the FAX CNG tone (that is the tone in the middle of the 9 values), that should make it possible to detect FAX in the auto attendant.

Whow maybe we should make 512 the default then!

Did anyone check out http://www.linux-ha.org/GettingStartedWithHeartbeat? I think that does solve the problem for Linux-based installations.

Strange. Maybe just change the system name and hit the save button in the admin settings page to make sure that the file gets written. You can use notepad to see the file, not beautiful but you should see the XML tags.

Yes, that link should do the trick. You can use for example notepad to edit the file. First check if the file now contains the setting, then if that is the case you don't have to edit it.

Well the problem is that there is no common standard. Essentially one PBX - one attendant console. So we have to do our own :-(

Are there any? I know the OC is overdue - we changed the project from Visual Basic to C#...

It is in the Global Configuration File, http://wiki.pbxnsip.com/index.php/Global_Configuration_File. Yes, it is invisible - in this case for a good reason!

Whow that is really very silent. The threshold is 50, and those values are far away from that. The problem is probably that we do the algorithm in fixed point, 32 bits. That is a must in the embedded world, on the PC we could use floating point. Maybe we can solve the problem by having a DTMF gain (at least available as static setting), I have prepared a build at http://www.pbxnsip.com/protect/pbxctrl-2.1.6.2442.exe that contains a global setting with the name dtmf_gain. The default is 256, which means factor 1. 512 would be factor 2 and so on. Maybe you can try this version and see how far we have to amplify in order to get the cell phone DTMF working... and of course it would be interesting if landline still works!

Setting the time was a pain in the neck in the embedded system. There were a lot of installations that did obviously not have a DNS server to resolve ntp.org or had no Internet access at all. Then the system gets a bad start and things go south ways.

It is in the email section for the extension ("Send email record for all calls").

Check out http://wiki.pbxnsip.com/index.php/Installi...P-PBX_Appliance. P.S. Maybe it is worth waiting a few days. 2.1.6 is coming around the corner and it also includes some benefits for the CS410.

There are two things that need to be done. First, you need to need to periodically copy the file system from the active to the standby host, ideally using a differential copy. The second thing is to switch from standby to active. When this happens the standby service needs to be started. I guess we need to come up with a shell script that does this.

Sometimes providers send from different addresses than they receive from. This is not very NAT friendly and it makes it very difficult to authenticate incoming requests. In this case it makes sense to have one trunk for outbound and another one for inbould.