Vodia PBX

I believe that depends on the browser and the version, maybe even the firewall. Unfortunately the HTML5 features are still in the process of stabilizing in the web world, it seems. The PBX does nothing else than just delivering the audio data, and I am pretty sure that we did not program anything that tells the PBX to hang up after 3 minutes. Anyhow, if you can get us a PCAP from the maybe we can see if there is a TCP disconnect and where it comes from.

Interesting thought. How would the sequence look like? Receive a call from A; hold the call (e.g. with flash hook). Then call B, hold that call as well. Then call *xxx? We have something similar for the establishment of a conference call; but it did not occur to use yet that we could use that also for a attended transfer. The workaround would be to have a conference call for a second or two, then disconnect from the conference and let the other two parties continue the call.

This is a common problem with the firewall. Check if the customer's firewall has anything with "SIP" and/or "ALG" (application layer gateway). It probably detects the SIP traffic and applies a security policy to it, in other words, blocks it. One common solution to this problem is to use TLS and encrypt the SIP traffic. This way, the firewall only sees binary spaghetti and does not mess with the SIP traffic. In the 5.2.3 version we have introduced a new parameter for the provisioning of the domain, in the general settings at the bottom. There you can tell the PBX for what IP address (the customers public IP address) with what netmask (32 if this is only one IP address) the PBX should provision the outbound proxy (the PBX public IP address) and what port (5060 typically) and what transport protocol. For the transport protocol you can try udp. Then restart the phones, let them download their PnP configuration and check if this solved the problem.

Well if there is no change with the calls you would not get any traffic on websocket. Websocket is really just to tell the browser that something has changed, but not what has changed. For that the browser will make a (non-websocket) standard REST call. Try to watch the traffic when there is a call coming into the system, or when someone hangs up. I think then you will understand the logic better.

Does any SIP trunk provider support this yet?

It is probably easier to just go back to the good old version from 2011 or so which was working quite stable. The PBX does not really need the new features that have been put in since then.

The 192.168.7.2 is coming from the USB connector (as far as I remember). Whoever did that on the beagle-bones wasn't a networking guru. You obviously hit the jackpot and have an overlap. If you can log in with SSH there is a file somewhere in the networking section where you can find the address and then possibly move it to another address. The 127.0.0.1 DNS is because there is a local DNS server running which is in turn talking to the real DNS server. That is not unusual especially for embedded systems that have only very primitive DNS clients without cache, though I personally would not do it that way. The PBX has a good DNS client that can handle the caching fine.

For that you would have to "nuke" the mini using a SD card and re-install the PBX.

You don't need to scan the network. Just use the link-local IPv6 address, this is always the same.

It is currently on the sysadmin level when you edit the domain.

What we would do is to to try the JTAG debug cable on it. If you want to avoid the shipping, you could try it yourself. GuruPlug JTAG is what you need to search for,

Well honestly we are also wondering what is going on. The new beagle bones are coming with Debian pre-installed, a distribution that we understand much better and which probably behave a lot more stable than the Ångström one. We don't have any magic backdoor; we would try to use the USB connection to get a grip on the device.

Nun ja Active Directory ist letzlich LDAP. Keine Ahnung was sich Microsoft in letzter Zeit hat einfallen lassen, um uns das Leben leichter zu machen. Aber es ist vielleicht mal den Versuch wert, darauf zuzugreifen.

The "Include following extensions when this extension is being called" is a mini-hunt group that comes with every extension. There you can include other extensions, but not other accounts like hunt groups or ACD. It is meant for very simple key-system like deployments (e.g. home) where even a hunt group would be too challenging for the user,

Yea, that's a kind of classic.

Yes. You can use the REST interface for this. If you want to know how it works, just use your web browser and watch the traffic between the PBX and the browser. It is based on websocket, which triggers the actual REST transaction. You can use Basic authentication instead of logging in if you want to use an external program like curl.

For new deployments you should consider using the 7xx series. They simply have much more memory where it is no problem to have all languages "on board". While snom adds more features to 3xx they need to make some room for then, and it seems that the language prompts are the way to achieve this.

Interessante Idee... Normalerweise haben die Leute ja eher das andere Problem. In einer Rufgruppe kann man den From-Header weitgehend manipulieren, das könnte eine Lösung sein. Z.B könnte man als Gruppenname sowas wie "Unbekannte Nummer" wählen, und damit hätte man dann den Effekt erreicht. Ansonsten würde mir noch das Adressbuch einfallen wenn es darum gehe, bestimmte Nummern nicht anzuzeigen.

As you might have seen on the newsletter (http://vodia.com/news20140827), we have released version 5.2.4. This is supposed to be a "non-exciting" release. In other words, not a lot of new features, but essentially bug fixes and minor improvements. We hope that this release can run in productive environments for a long time unless you need a 5.2.4+ feature.

I would record the PCAP for that extension. Maybe it is a just problem with the firewall.

Ja, bei ein paar Sachen müsste Polycom die Firmware ergänzen, z.B. Anzeigen von Photos oder CMC. Da gibt es einfach keine Standards; wir haben das damals für snom programmiert was man nicht als Industriestandard bezeichnen kann. Im SIP-Standard findet sich ein Sammelsurium an Ideen, u.a. für Bilder; die sind aber einfach in der Praxis viel zu vage und bringen keinen so recht weiter. Wir haben vor geraumer Zeit die Unterstützung von Polycom Buttons eingeführt, wobei allerdings nur BLF und Speed Dial unterstützt werden. Jedes Telefon kann sein eigenes Button-Profile haben. Die PBX unterstützt (unabhängig vom Telefon) ActiveSync. Active Directory ist etwas anderes, hat heute relativ wenig mit Adressbuch zu tun. Bilder werden jetzt durch drag & drop hochgeladen. Naja, es sollte einfacher sein...

Also, you can use a Vodia mini PBX on site that registers a trunk to the paging account and send the call to a local paging account that can do the multicast RTP then.

I would be very careful with static IP addresses. On a server that has no console, you can very easily render the device unusable. I would tinker with the IP configuration only if I am logged in on IPv6 or via the USB connection, so that the IPv4 configuration does not matter. A much better solution is to assign an IP address on your DHCP server, so that it always gets the same address.

Apart from the hunt group approach which requires that you would use three registrations, DECT base stations usually have some group calling feature as well. I am sure that the GS also has that. I would try to set it up locally and have only one registration with the PBX.

In the dial plan, you can use pattern xx and in the drop down select "Not Allowed".