Vodia PBX

On 4.5, DND means that calls are being rejected; however you can define exceptions of people who can call the person on DND. Typical scenario is a boss who is on DND when the secretary is available. The secretary can still call him. On 4.5 there is also that boss/secretary feature where the boss can turn redirection on/off with a button on a snom phone. It is not DND, but could also be a solution to the problem.

Sure. In the trunk setting, there is the setting "Destination for incoming calls" where you can for example choose to match digits and if that fails, define a default account where the call is being sent.

Well Firefox uses DTLS, which we first need to implement. The core problem is that for public WebRTC usage, most of the browsers should support WebRTC. Internet Explorer, Safari and many mobile browsers don't support it and it is questionable if they ever will (Microsoft spent more than 8 billion dollars on Skype and Apple has Facetime). So at the end if the day,we need to ask the extension users to use a specific browser anyway and currently the answer is: Chrome.

No. But you can use your V4 activation code as coupon for the V5 license.

Ops, die Frage ist "hängengeblieben". Es gibt prinzipiell zwei Möglichkeiten: 1) ATA verwenden. Das ATA wird wie eine normale Nebenstelle registriert. Fax-Anrufe sollten dann auf dieser Nebenstelle landen. An dem ATA wird dann ein analoges Fax-Gerät angeschlossen.Vorteil: Man kann damit leicht wie gehabt Faxe senden und empfangen. Nachteil: Wenn Pakete verloren gehen brechen Faxe sehr leicht ab. Als ATA kommt z.B. SPA112 von Linksys in Frage. 2) PBX als Fax-to-PDF Konverter verwenden. Die PBX kann eingehende Anrufe mit Fax in Emails verwandeln. Vorteil: Kein Papier mehr und die Faxe können leichter in digitaler Form verteilt werden. Nachteil: Fax senden geht damit nicht.

Hmm, indeed strange. For now, I guess there is no choice but staying with 5.1.3. We'll have to investigate.

Should be a setting in the domain voicemail settings.

Ah you are not using Internet Explorer? This browser is not able to play WAV files, believe it or not.

I would assume end of next week. Still struggling with a few nitty gritty problems.

Well regarding the ports, it might take a minute before it can finally (exclusively) bind to the ports 80 etc. I would not be worried about that. Just double check that you don't have tow PBX processes running at the same time. But it seems that your PBX configuration got wiped out somehow before the new image started up. File system full?

Ops... Well then you are in the position to make this a rock-solid installation . What exact DiffSrv you are using IMHO does not really matter, as long as it is not the default QoS tag for the other IP traffic. Maybe you can set up a test where you have a phone call and send a packet storm for example downloading a huge file from an internal server, without the QoS rule on the router and then with it. It should make a difference.

If you are using version 5.1.3 or higher, the PCAP are already decoded. If you are looking at raw PCAP directly from the interface (either on the phone through the web interface or on the PBX), well it will be hard to crack the decryption hopefully. I think Wireshark has a way to decrypt SSL traffic, but I have never done that before and it sounds like a lot of work. Using the 5.1.3 PCAP on the PBX is a lot easier.

Well if you have no control, setting QoS bits is a waste of time. Then you can think about using a traffic shaper to increase your chance to have good quality when other services are on the line as well.

What version was this? We had some issues with recording directories in 5.1.2.

I think the solution with the setting on the phone sounds like a good solution. It does not block call waiting per se; it just makes sure that you still deal with the current call and not with the incoming call.

Okay, please open a ticket on that so that we can get this resolved. One of the key questions is if you are using a global trunk or you are using the "try loopback" function in the dial plan.

Well you can still clear the country code in the domain, then the PBX cannot interpret numbers. But IMHO you will loose the possibility to match address book entries, call back numbers and things that need to understand what number was dialed.

It will be tricky to make a distinction between internal end external calls (you could set G.722 as the first codec for extensions, but G.711 for trunks). Most customers don't worry about the quality of internal calls as they don't directly generate revenue (smoke signs are good enough). The focus is on having a great outside perception with customers, and there the simple answer is that G.711 is the best codec.

Difficult topic. We recently had a carrier that needed the numbers in +-Format in the From- and To-headers and in E164-Format in the P-Asserted-Identity . We have added some more representations in order to deal with such requirements. But the dialed number is always in some format. It is amazing how complicated the world makes the use of telephone numbers! I think we need to look into the details how the provider wants the numbers in which headers and then set up a trunk template accordingly.

There is layer 2 and layer 3 QoS. Layer 2 is practically VLAN, where you give the packets in that VLAN a higher priority than packets in the default VLAN. The Ethernet switch is here the device that you need to control. If you do it right, you can have a ISDN-like quality level in your LAN, even if there is very heavy traffic in the LAN. You can set that up manually, but better is to use LLDP which is a kind of DHCP for the VLAN. Layer 3 is using some bits in the IP header to tell the router with what priority the packet needs to be routed. The point here is that you need to have the router under your control. Those QoS bits are practically never used when you send a packet over the public Internet. But you can use then for example if you have a large corporate network where you control the routers. From the PBX perspective, the PBX stays out of the discussion. You need to make sure that the PBX has a interface that has the VLAN set up right. The PBX tags like the phone the packets with reasonable bits on layer 3, so that the subsequent routers can process the packet accordingly.

G.729A is compressing the voice to 8 kbit/s, which is useful if you have limited bandwidth at the cost of sounding a little bit like tin-can. G.722 is using the same bandwidth as G.711 (64 kbit/s), but does a more clever encoding than G.711 so that the higher frequencies can be heard by the other party. Whatever codec you choose, you need to be aware that whenever the call gets into the PSTN, it will be 99.99 % of the time on G.711 (a-law or u-law). If you are coming from G.729A, you have additional transcoding costs. Surprisingly, this is also the case for G.722: even though it is a better codec than G.711, you will have a inferior performance than in a system that uses only G.711. This is because translating from one codec to another codec can not lift the quality up (invent a better sounding signal) unless you have artificial intelligence engaged. Ten or twenty years ago bandwidth was still a problem; but IMHO most offices today have no problem sending a megabit to the Internet, so G.729 is kind of outdated. G.722 sounds nice at first glance, but it is difficult to keep it along the path from caller to callee. IMHO today you get the best end-to-end quality still using the good old G.711 codec.

We have introduced a setting in 5 that determines what to do when the extension is busy (busy on busy). Depending on that the redirection settings are used or a busy code is sent back to the caller. I am not sure if that solves your problem; if you can install a small test system somewhere with a free license where you can test if this works for you on 5.2.0.

I would make a backup and move the system to 5.2.0 or at least 5.1.3. As for the snom 710, there is a new firmware available that fixes many problems (you might experience some of them). This is by default part of the 5.2.0 release. If you want us to upgrade the appliance from SSH, no problem open a ticket with the SSH access and we'll do it for you.

If you want to loop back calls they must be telephone numbers, starting with +.

For time being, just use the attached file (rename it to webtemp.js and drop it into the html folder in the working directory of the PBX). Make sure you delete it when the next release is available. webtempl.txt