Vodia PBX

This will be in 2.1, including the firmware provisioning (16 MB and more). It should be working in the latest RC (make sure that you don't have your own pnp.xml file).

If you have just one domain, just call it "localhost" and the PBX will happily accept any domain name in SIP packets.

Ouch. That was an "old friend". The problem is that the internal table for mixing audio is quite short. Thanks for the tip! We made a quick hot-fix image at http://www.pbxnsip.com/download/pbxctrl-2.1.0.2098.exe. Please do a quick verification (we could not do it here yet, so make sure you can move back easily).

Okay, what about this: The PBX keeps a list of the agents. Everytime that an agent picks up the phone that agents moves back to the list: Initial: 1 2 3 4 5 Call 1: ring 1, ring 2, ring 3, 3 picks up -> 1 2 4 5 3 Call 2: ring 1, ring 2, ring 4, 2 picks up -> 1 4 5 3 2 Call 3: ring 1, ring 4, 1 picks up -> 4 5 3 2 1 And so on? That should be easy to implement.

Try *90123 if 123 is the extension that you would like to wake up!

Ops, yea probably a upgrade issue as the PBX now differentiates between P-Asserted-Identity and P-Preferred-Identity.

I would prefer Monday dinner. Does anybody have an idea where we could go? I only remember a sports bar in the staples center, could be a fallback-solution. Maybe someone knows LA a little bit better than me...

Least idle?? Most idle would be my guess. Or stuff like LIFO.

:-/ Are you talking about the Diversion header usage? AFAIK that should have not changed - unless you use the "charge" field in the trunk that goes to the Exchange.

In theory it should be possible with the latest 2.1 version after a call has been initiated to switch to video. This is just like T.38 - first audio session and then switch to "something else" - which can be video, T.38 or whatever. Worth a try, but I never tried it.

Here we go with the latest: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2097.exe (raw Windows executable) http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2097 (CS410 appliance) http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.0.2097 (Debian 3.1) http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.0.2097 (SuSE10/32-bit) http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.0.2097 (RedHat ES4/32-bit) There were still problems with one-way audio and unwanted Re-INVITEs.

That will be covered in the 2.1 documentation - once that it is finished...

It is a good idea to keep numbers in a normalized form. If you are in the USA reagion (10 digits fixed length), you should tell the PBX so and select the dial plan in the domain accordingly. Outside of USA people start to believe that the number +xxxxx (+ sign indicating that it is a globally routable number), I am amongst them. Also that means that you should program the PSTN gateway so that it sends diallable numbers. Same in address book. BTW because of this it is a bad idea to put prefix into the dial plan. At least for "normal" calls they should be diallable without a prefix.

Try the patterns 75xx or 75([0-9]{2})@.* In the replacement, try: sip:*75\1@\r;user=phone or sip:\*75\1@\r;user=phone (not sure if the escape symbol is needed)

The PBX probably sends out a MWI whenever there was a message recorded for that account. BTW it also sends out messages from time time time (around one hour) so that lost messages get recovered. It should not have any negative side effect as the message count does not change and the UA should realize this.

Yepp, this should already be fixed, if you can try the latest RC1 image.

Yea, there was a lot of trouble with SRTP bugs. In the 2.0.3 version we decided that we don't support any "backward-compatible" bugs any more and to plain (and possibly painful) SRTP. The workaround is obviously to disable SRTP on the snom phones.

Yes, 2.1 RC1 has it.

Whow! That finding is very cool!!! That could explain why we sometime see (even on very powerful machines) jitter coming up as a problem. We will add a setting for 2.1 so that this can be set up as a setting. Seems that Linux also supports this, and it is probably not a bad idea to include it there as well.

Hmm. Does the phone send a redirect message to the PBX? You can try changing the mailbox pickup timeout to 15 seconds, so that the PBX will redirect the call to the mailbox, not the phone.

You need a phone that uses SIP as interconnect protocol. Or you can use a ATA, but then you will have limited functionality.

Aastra also has "BLF" settings that we got working (at least the lights were changing state). I would try manual configuration here. You can also program speed dial buttons, which come handy for call pickup (e.g. program it to *87123 if you are watching extension 123). Cisco: IMHO they don't support that yet in SIP. Skinny seems to be more important to them than interoprability. Grandstream: Not sure. But AFAIK their extension board also supports dialog state, which is what we are doing. Manual configuration. Polycom is a little bit more complicated as it is practically impossible to manually configure it (well, maybe worth a try), see http://wiki.pbxnsip.com/index.php/Polycom. There you need to use the address book and mark the entries that you want to watch. If you enter that in the extensions "Watch the calls of the following extensions" the PBX will automatically provision that for you. We did not figure out how to make them pickup a ringing call, but at least you can see that another extension is busy. Snom: If you can live without Plug and Play, take a look at http://wiki.pbxnsip.com/index.php/Snom, there you can see how to use it with snom phones.

If you are using 2.0.3.1715, better use a from header in the form dallred@vensureinc.com. There was some trouble with the parsing... If you telnet to that server on port 25, you should see a welcome message coming. If that is not the case there is a problem with the SMTP server.

There is echo cancellation, done by a DSP subsystem.

2/10 means that two agents are added to the list of ringing agents every ten seconds. The selection is random - amongst the agents that are not having a phone conversation on the PBX or are in the "recovery time".