Vodia PBX

No, just put the number without anything behind it into the src parameter.

Well, both calls from trunks and extensions should work if you turn the according flags on.

Looks like it was really a bug, but it is fixed in head. We will make a beta version available shortly.

Can you send me login data in a private message? We would like to try that out.

Okay!

Try using the parameter "src" in the URL (same as "dest", but for the other side of the call). Then you can define both source and destination of the call. The acconut that you provide will be charged for that call.

You can provide the credentials in the URL (see http://wiki.pbxnsip.com/index.php/Click_To_Dial). But for that the user also must be registered. IMHO everything else is just a big security leak?

Are you calling from a number that is a cell phone in one of the extensions?

Well, both "valerio_capodacqua" and "300" must exist on the PBX in the domain, preferrably "localhost".

Well - sorry at the moment you have to punch in that information one by one... But sounds like something that we can do to make life easier.

Whow. Are you using the name "localhost" as the domain name? The "Not Found" might be easy to fix. However, this is untested terrain and by looknig at the headers you can see there is a lot of MS-specific stuff that might cause issues later. For example, the attachment in the SUBSCRIBE will not be looked at.

Okay - then the Wiki page is probably the most important information for you. If you look for general information on Linux, then the web is your friend on general documentation. The PBX is a real-time service, so the biggest point is to get the service started after a reboot and apart from that there is nothing exciting about the Linux setup.

Oh that is very slow - probably the TCP connection just times out. The good news is that the server obviously does keep the connection alive. I think the easiest is to get an Wireshark trace and check if there is a HTTP interop problem.

Is your web client re-using the HTTP connection? Every time the client opens a new HTTP connection the PBX hits the break. There is a setting in pbx.xml called "http_rate" which tells how many new HTTP connection the PBX should accept per second. If you increase this value, the speed should go up - but also the risk of DoS the PBX through HTTP storms.

I would recommend to get started with Windows first. Having a new product (pbx) and a new operating system (Linux) will make this project hard. Once you got the PBX under control in Windows, you can just copy the files and continue in Linux. Our Linux build does not support dongles yet.

Turn off silence suppression on your phone - the PBX probably thinks that this phone already died (silent as it is).

If you like, try the attached WAV file as ringback file. There is also a busy file available. ringback.wav busy.wav

Well, that might be your conclusion. Most phones actually do not put a call on hold when another call comes in. If you have such a phone, open a trouble ticket with the phone vendor and/or set the number of lines for that extension to 1.

Okay. BTW you can also use SNMP to track the number of calls: http://wiki.pbxnsip.com/index.php/SNMP.

Okay. BTW you can also use SNMP to track the number of calls: http://wiki.pbxnsip.com/index.php/SNMP.

No really. I think Wireshark would be more useful, or a SIP-trace.

No (on the head versions we usually have Windows executables).

Still sounds a little bit like a mystery to me... It would be great to know what exactly is causing this, then it is probably a five-minute fix.

Yea we took that out - it is still there in domain mode, but in admin mode we took it out.

For that you need a "PSTN" gateway. It is not really PSTN that you are calling, but from the PBX perspective that gateway should behave like a FXS (so you need a FXO gateway). Examples would include AudioCodes, Dialogic, Grandstream, Mediatrix, Patton, Vegastream, just to name a few great products.