Vodia PBX

There were a couple of changes between 2.1.3 and 2.1.6 regarding the codecs, especially in the transfer scenario. I think 2.1.6 is much cleaner there, probably a SW upgrade fixes the problem.

Still don't get it. What time zone did you select? What version of the PBX?

That is not true... There is a SOAP-based API, e.g. for settings extension parameters while the system is running. We don't promote it too much, it is very support intensive and the number of users that really need it is quite low.

Well, curl is a good start. There are SOAP requests that make it possible to do anything, but IMHO curl is much more simple and does the job.

Well the default license for the CS410 does not include recording - we simply suspect this will cause a lot of issues with people turning recording on and then one month later boom! the PBX dies and they ship the device back for repair.

The RAM is not the problem here. Recordings are written to the file system. The 410 has 256 MB (minus a few MB for itself), which could keep a couple of hours. You can also use a sip URI to offload the recording to an external SIP-compliant recording system. If you want to do serious recording, IMHO you must use a PC. The embedded system is not good in storing large amounts of data and it is not easy to pull the data off the box.

Did you see http://wiki.pbxnsip.com/index.php/Recording?

Hmm. 1 should be Sunday. Do you mean that for Polycom "2" is Sunday???

We don't run it here, but Vista was running great so far. Especially if you use IPv6.

So far it seems there is a problem with the FXO driver, we are looking into it right now. So I would say better don't upgrade right now.

Whow, either that system is heavily overloaded or there might be a problem with UDP transport. Could packet loss be a problem? Of course, Wireshark will be able to point the problem out.

Hmm. First question of course is what version you are running on the phone (and the PBX). Second it would be good to have a trace so that we can see what is going wrong. If there is a probability involved, it sounds to me like a race condition. Maybe the phone behaves differently if it received a BYE message during the transfer - or not.

Yes, check out http://wiki.pbxnsip.com/index.php/Dialog_Permissions.

You need to put that file into the html directory (make it if it does not exist). After changing the file or loading it there the PBX service must be restarted...

A semicolon might actually be a reasonable choice - it will make the match pratically impossible. I would be careful with numbers, maybe someone really dials that number and then you have "funny" effects.

Check out http://www.wireshark.org.

There is some interesting information on http://wiki.snom.com/Features/LED_Remote_Control that explains some of the background.

Check out http://wiki.pbxnsip.com/index.php/Localization#Time_Zones - if you have a time zone that we should include in the standard distribution we are happy to put it in!

Yes, stealing calls is possible. If you want to avoid this, just use the speed dial mode in the buttons select box on the PBX (see http://wiki.pbxnsip.com/index.php/Assigning_Buttons). Then you will just monitor the extension status - if you press it you will dial it (that's why it is called speed dial). This is the original BLF idea. And there is always the dialog permission. With this setting you can also control the access rights to information about calls on the respective extension.

So far yes. 2.2 will also add a memory sensor (number 12), but is not released yet. Ouch. Network equipment like switches and routers are also not bug free. We have to keep this in mind when tracking down problems.

Did you use a higher priority for the next entry in the dial plan? http://wiki.pbxnsip.com/index.php/Trunk_Se...tbound_Settings was not clear on this, has just been updated.

In the replacement, you can use a pattern like sip:\*43@\r to make the PBX dial *43. In the pattern you need to use something else than the star character, because that is the code that tells the PBX to look for it's own codes. Maybe you can use a pattern like 012* as prefix and sip:\*\1@\r as replacement, then if someone dials 01243 the PBX would dial *43 on the trunk.

Whow how did you get these HUGE screenshots? First, I would always use the outbound proxy. Many SIP devices don't like it when the PBX says send it to "localhist"... Don't use the STUN server (STUN is nonsense anyway). Question: Why do you use a trunk on the PBX? I think you would register the Grandstream as an extension. Who is registering where? PBX trunks send REGISTER out, but do not receive REGISTER messages.

If you can, use an SNMP tool to poll the PBX for data like how many calls are on. If the PBX does not respond, there is trouble going on. It is *not* a good idea to stress the system with lots ot http/https requests to see if it is (still) responding. SNMP is very lightweight and has practically no impact on the operation.

Ehh... FXO is not FXO... Maybe we really need to think about a couple of settings, like "check polarity change", "loop disconnect timeout time". We are using this new Comcast service and I love it. No such problems, just great audio over a very short FXO cable.