Vodia PBX

Well, 2.0 had hand-crafted images for the menus on the top. That was obviously not very smart and has been replaced in 2.1 with a simple html interface that just uses a few background images (some people were asking if we can change the color and the answer was "ouch"). F5 is your friend now. If you use your own design, most of the logic should still work, but just make sure that the cor_xx.gif images and the main_*.gif stuff is moved away. Sorry, but in this case it is really hard to stay 100 % backward compatible...

On a scale between 0 and 10, how useful is http://wiki.pbxnsip.com/index.php/Audiocodes?

The clear log is fixed. *87 must always result in audio, if there is no pickup available then there should a IVR annoucement.

Today we made another version. There was a problem with the RTP pass through and the keeping of the SSRC and the packet numbers. Fingers crossed this topic can be closed now. Maybe it was also the reason why people were reporting DTMF problems. http://www.pbxnsip.com/download/pbxctrl-2.1.0.2107.exe

Version 2.1 has global settings called "timeout_hold" and "timeout_connected" which can be manually edited in the pbx.xml config file. You can set it to any value bigger than 10 seconds. Depending on the phone and the software version, the phone should send some kind of keep-alive traffic during mute. Silence does not mean there is no RTP!

Yes, http://www.pbxnsip.com/download/pbxctrl-cs410-2.1.0.2105.

Yea, we changed something to make logging faster (and avoid locking out other threads), and the clear log was overlooked. Should be fixed in the next. Thanks for reporting!

Try "sip:192.168.1.3:5061;transport=tls" as outbound proxy. If you see "Sent to udp:192.168.1.3:5061" then the phone uses the wrong transport layer. I guess you should also reboot the phone after that. And try to use version 7.1.19 or higher on the snom 370, previous versions might be buggy. See http://wiki.snom.com/Snom370/Firmware.

Sorry, should be there now. I think so. Outstanding issues (top of my head) are: (1) Some strange stuff with the mailbox when you enter the wrongpin the MB starts recording itself (?!). (2) CO-line monitoring seems to be fixed, but need to verify. (3) There is still a case where T.38 does not work (properly?)

Can you post a screenshot or something like that? Would be great if others have the same problem...

We are still testing and finding issues. So for production the latest and the greatest is still 2.0.3.1715. The latest Windows build of the day is 2.1.0.2105 (http://www.pbxnsip.com/download/pbxctrl-2.1.0.2105.exe), if you like you can try this one out and help exposing it more to the real world. We are running it already, but still have one or two tickets open that we want to close before releasing it. Once we are through with 2.1 QA, one thing is sure: This will be a great release addressing many features requests (TAPI, email spooling, G.722 and Re-INVITE, multiple mailbox messages, ...) and fixing several issues that we had in 2.0.3.

Look at http://wiki.pbxnsip.com/index.php/Troubles..._Trunk_Problems for trunk problems. Also, you can look at the BYE packets send from the PBX. There you can see how many packets the PBX received. In one-way audio cases, that is very useful. If it all does not help, you can use Ethereal to see what is "truely" going on on the cable level.

There are images available for the Teles box that run 2.x. The upgrade should be relatively easy, once you are ablt to figure out how to put a new image there. My suggestion is to wait until 2.1 is out and then upgrade to that version (we will include the NetBSD build for 2.1 as well). Should not be too long until then.

What did you use for outbound proxy? Something like "sip:192.168.1.2:5061;transport=tls"? The transport=tls is important because otherwide the phone will use UDP. Do you have a SIP trace from the phone's web interface? PCAP are useless, due to the nature of TLS we will not be able to see too much...

Nono. The port number is not the point here. Proxy Server/Registrar Server are also not the point here. There is a setting called "outbound proxy" - I think it is in a different config page. In the config file it looks like this: sip outbound proxy: {ip-adr} sip outbound proxy port: {sip-udp-port} No need for DNS. Lets keep things simple! Then you should be able to put domain names into the proxy and registrar - they should be the same (e.g. company.com).

1. First, you need to subscribe to dialog state. Through http that is difficult, but if you can do a SIP SUBSCRIBE that should be easy. Just put a contact like "Contact: <http://192.168.1.2/dialog_response.htm>" there. Through HTTP is a little bit difficult at this time, you would have to do this trough SOAP. However, we will quickly add a type field to the web request, so that a regular http post can do the job (in the registrations tab, there is something at the bottom that adds a registration). If you just want to test it, just create a static registration without a type, and edit the XML record in the file system. After that, you need to restart the service though to read the change. 2. The notifications then are walking out as http post requests to the provided location. They contain the same body as the SIP NOTIFY in a SIP SUBSCRIBE dialog would do.

I think you can copy and rename the aa_busy_callback.wav and use that one.

ACK. Needs to be fixed before we can release this.

Well, so far the answer is dialog state. There are a couple of phones that support this standard and that kind of works, for example subscribing to the dialog state of CO-lines. But it is not such a good answer if you want to have something on your PC. We are thinking about AJAX, or maybe Flash or just running a native Windows application.

The call list is just another table in the PBX. So in theory, you could read that table using SOAP. But I am not sure if that is a pragmatic way to go. Alternatives: 1. Register a dummy user-agent and subscribe for the dialog information. 2. 2.1 will introduce "buttons", which are instant messages that carry call information. You still need to use SIP for that, but it is much easier than dialog information. 3. You can also subscribe for dialog information using HTTP. Then the PBX will push that out via a HTTP request. Everything not very appealing...

Well, that has been addressed in 2.1. In the meantime you will have to live with escape characters and complete extended regular expressions...

Set the outbound proxy of that phone to the PBX (e.g. the IP address) and set the domain name of the registration to the name of the 2nd domain. If you want to push it, don't use the name localhost in any domain.

Most people just trunk into Asterisk. Just use the gateway mode and you should be all set.

Why 2.0.3.1707? There is 2.0.3.1715 which is the last release and this image likely fixed that DoS issue (I remember that intermediate build had a problem). There is a global setting called "max_udp_invite" which defaults to 10. If you have more than 10 new calls per second, you might need to increase that value.

Well, the firmware files that you get for the Polycom phones. It is a little bit difficult to get, you need to talk to your reseller if you want to go the official way. But I think it is also possible to find it on the Internet.