Vodia PBX

Your dialplan can be just like this: 1;Twilio;;*; Just a single line that matches anything (which is the default dial plan) Is the SIP trunk registered? Without successful registration there will be no inbound call.

I do trust in /proc/xxx/fd. There must be some kind of operation that is causing this - e.g. some problem with a FAX or some problem with streaming MoH that is causing this. Other services do not seem to have this problem. Is there any clue in this direction? Do you have "expired" files in the working directors? The ports seem to be on RTP ports, so this must be something media related.

WebRTC works only if you have a secure connection (HTTPS). Did you login through https?

Just go to the software update page - there you will see the download links right on the page.

It will be in 63.0.3 (PBX) - should be available end of the week

It will require a SW update ...

You can see it here -

It actually did update the display if you use a private line (which is the default). However the information was at the very end. We'll change that in the next version, then it should be a lot more obvious.

Oh - got it now. We so far had the idea to use the provisioning process to provide a better experience - however there are a few new problems with this approach that we need to address, for example devices that are not automatically provisioned and how to map button profiles onto those accounts. But what we can do is to re-provision the phone and set the display name differently. This should work with many phones and solve most of the problem.

It seems to be a popular demand - we'll add that to the next version.

Did you try the drop down when setting up the SIP trunk on the PBX? That should set up the right headers and everything for the Twilio trunk.

There is a flexible way for determining what ANI to use in the SIP headers for a trunk INVITE at https://doc.vodia.com/trunk_ani - however it mostly depends on what the provider is able to do in terms of showing off-net numbers. Keep in mind that spoofing caller-ID is a problem and the provider somehow needs to verify that the number is valid.

Seems that Firefox needs an additional step (see https://stackoverflow.com/questions/32225904/programmatical-click-on-a-tag-not-working-in-firefox) we'll add that to the next build.

I don't think there is a bug in lsof. In the previous post it looked like there were multiple listings for the same port - which is of course raising questions. What would make sense is to check /proc/xxx/fd if that matches the output of lsof in terms of open sockets.

The PBX feeds numbers into the dial plan in the way people in that country would read the numbers. In Australia that would mean that domestic numbers would start with 0 and international numbers would start with 00. If there is an area code and the number would be in the area, numbers would start without the "0" in the beginning. So if you want to match international numbers you would use a pattern "00*" and then in the replacement also "00*" so that the matched number would occur behind the "00" prefix.

Well there was a log of euphoria around ENUM many years ago - however comparatively little action followed so far. The way ENUM works with the Vodia PBX is to include a special flag in the SIP URL that comes out of the dial plan "enum=true", e.g. use a replacement pattern sip:+61\1@\r;enum=true.

The imager.js should be included in the 63.1 dat file - sorry it was missing because we have changed the build process.

There is a bigger topic "click to dial". There is some information at https://doc.vodia.com/click2dial as a stating point. I would suggest that you are sending the request to the PBX instead of the phone, because the phone could be offline, behind NAT, in a separate VLAN or just change its model or firmware version. Especially if you have control over the backend you can do this very elegantly without any passwords exposed to the web front end or to the PC application.

We are also scratching our heads how we can get the extension thumb nail onto the phone via the provisioning process. So far we could not get any answer from the snom interop partner management. It works beautifully when done manually from the web interface. But there seems to be no provisioning way to do that.

The idea with the v10 template was to get rid of all those phone specific templates that are getting really hard to maintain. The D735 should pull the template snom_v10.xml - do you see that in the PnP history for the extension? There was a lot of confusion about the device names, but for the D735 there should always the "D" in front of the number. Last resort would be turning on the logging for the web server to level 9 and see what URL the phone it trying to pull from the PBX.

Hot desking works completely independent from the phone type you are using. This should work on a analog terminal adapter as well as a top notch VoIP phone. Did you try internal calls? Or does this happen on external calls? For external things can get complicated depending on what ANI rule is active, therefore I would check internal calls first.

By default the "camp on" feature is enabled - if you disable it and the mailbox is disabled too then the call should get rejected.

The availability of the files might depend on the language that you have installed. You can see in the audio_xx directory what files are available, including those for the auto attendant.