Vodia PBX

I think you need to use the "divide and conquer" strategy here. Meaning trying to narrow down the problem. Essentially you have to make sure that the phone can talk to the PBX and that the PBX can talk to the provider. Your enemies: NAT and firewalls, and maybe even bandwidth. To make sure that the phone properly connects to the PBX, try calling the mailbox on the PBX a couple of times, and stay connected at least for a minute. Try calling another mailbox and leave a message, then (e.g. using the PIN) and listen to the message. Make sure that it is the same of what you actually said. Then when that is working, try calling your cell phone a few times. If you have random problems, you probably have a problem with the bandwidth. If it never works inbound or outbound, you probably have a problem with the SIP connection. Some more tips are on http://kiwi.pbxnsip.com/index.php/One-way_Audio and http://kiwi.pbxnsip.com/index.php/Troubleshooting_SIP_Trunk_Problems. You also might consider just using a hosted service. There are several providers offering hosted services based on snom ONE, please send me a private message to get contact data.

What you can always do is mixing your own WAV files and using them instead of the build-in ones. You can even create your "own language" (bilingual), for example use one language name for that that you don't use (lets say, Greek) and copy the default wav files into the audio_gr directory where you can edit them without any side effects to the other languages (for example, copy all audio_en files into the audio_gr directory to use English as the template). What many people do, is use one ACD and then mix the announcements in the different languages that they have. If you want to different language selections, you can also use two separated ACD. While they can be configured separately, this has the disadvantage that they have to compete for the same agents. This may result in unfair scheduling.

Hmm. Did you set anything up on the VMware regarding CPU Resources? For example, you could reserve some MHz for the appliance. You can also set the Scheduling Affinity. We know from non-virtualized environments that the processor affinity is important to keep the OS from swapping the process from one core to another, which causes tremendous jitter.

Actually, html/img (in the working directory of the PBX) would be better. You can then use HTML like this: <img src="img/yourimage.png" />.

There was a related discussion on http://forum.snomone.com/index.php?/topic/6359-the-mos-detailed/. The MOS values in the CDR are represented as fixed point values; to get the floating point values just divide by 10. Analyze audio level has nothing to do with it. This settings is useful when you want to calibrate analog gateways, but is has nothing to do with MOS.

Well, it is a long story and it started with Cisco and the usage of option 66 of DHCP. The default protocol was and is TFTP. Unfortunately, TFTP is not very NAT friendly and not secure either, but it works in the LAN and the MAC on layer 2 of the packet is a (weak) token of authenticity The phone fetches via TFTP essentially the real location, which is then https-encoded. So when you are in the LAN, the IP address of the PBX is okay. If you are outside of the LAN, things get more complicated; then you have to enter the whole URL and a username/password.

The great flexibility with the web interface comes with the price that this area is not 100 % backward compatible. But you can easily change it; just log in as system admin, to to web appearance and edit the reg_header, dom_header and usr_header files to what you want to see there.

Not 100 % sure about the updating of buttons. I remember that it *should* automatically apply the button settings, meaning that it sends a request to the phone to reload its configuration. It also might take some activity to set the buttons into action. Of course a complete phone reboot should make sure that the buttons are properly configured.

Getting closer... Just set it to "192.168.4.100" (don't copy the quotes). The PBX will then do the rest.

Hi Vladin, sorry to be a little bit superficial here. NAT is a huge problem in two-way communications, just mentioning that is does not work will not help finding the problem. Do you have a routable IP address (AKA "public IP")? How are the remote phones supposed to find the RTP media address of the PBX. I assume that the signalling is working; this can e.g. be done by a port forwarding. But you must make sure that also the RTP ports are properly routable.

To check if the phones were automatically provisioned, you can check on the file system the "generated" folder. There you find all files that have been generated during the PnP process (unless this feature was disabled). If you can actually log in to the phones without any password, chances are that those phones were not automatically provisioned. If you are in the LAN, it is actually not that difficult to provision them. Just copy & paste the MAC address into the PBX extension settings (registration tab), and enter the IP address of the PBX into the settings server setting of the phone (hidden under advanced/update/settings server in the phone web interface). The long version is on http://wiki.snomone.com/index.php?title=Plug_and_Play with all kinds of scenarios, even videos about it are available.

I agree... But how do you want to do this? Have a special code that tell the PBX to ignore the edge next time? Or would it not be more practical to have the LED on the phone come up at 5 PM and when the secretary sees that, she just presses the button and presses it again when she leaves? That is already possible today.

MOS = Mean Opinion Score. Usually it includes a lot of factors like microphone and speaker quality; however the PBX only focus on the factors that can be measured from the network point of view: jitter, packet loss, delay and codec. The score can be between 1 and 5, where 5 would be something like "CD quality" and 1 would be like "smoke signals". If you have at least 4.0, it is already pretty good. The maximum you can achieve with ISDN is 4.2 (using G.711 codec). The MOS can only be measured well if both sides support RTCP properly, and RTCP-XR makes it possible to gather reliable measurement data.

Mit B- oder D-Kanal im ISDN hat die PBX sicher nichts zu tun, das ist ganz klar das Revier des Gateways. Ich habe von Patton in dieser Richtung aber bisher nur Gutes gehört, inkl. Support. Ich würde die einfach mal fragen, und sicher können die auch genau sagen was auf SIP-Ebene signalisiert werden muss. snom ONE kann das dann umsetzen.

Well, it is a long time ago... What was changed is that the automatic settings define the change, not the absolute state. For example, when the flag is supposed to turn on the service at 6 PM and someone manually turns it on at 5 PM, then the automatic action does not change the state (because it already changed to the new state). This solves the problem that the secretary may leave earlier, but it does not solve the problem that she wants to leave later.

Could be that that function got lost during the introduction of the web template editing...

snom is working on one, prototypes are already available.

I just manually set up a SPA 112. It was pretty much straightforward, I don't remember changing anything. You might want to check the T38 settings and make sure that it changes to T.38 when it detects a CNG tone. The T.38 interoperability does not only depend on snom ONE and the SPA, it also depends also on the gateway or service provider. Anyway, for me it worked with no problems and I would just give it a shot.

Die in Frage kommenden Files sind audio_moh/bi_gong1.wav, audio_moh/bi_gong2.wav, audio_xx/bi_night_on.wav, audio_xx/bi_night_off.wav, audio_xx/bi_active.wav und audio_xx/bi_inactive.wav.

Ja das Wort "rumfummeln" würde ich nicht verwenden. Es gibt hier zwei Möglichkeiten: 1) Backup, dann bei der derzeitigen Version die verschiedenen Optionen (RFC3325, No Indication, ...) kurz durprobieren. Das dauert 10 Minuten in birgt keine Gefahren in sich. Wenn das nicht zum gewünschten Ergebnis führt, kurze Recherche welche Settings in Deutschland mit Patton und snom ONE funktionieren (z.B. http://forum.snomone.com/index.php?/topic/4836-conf-patton-4552-with-snom-one/) back up machen, Vorgesetzten informieren, Upgrade einspielen (Executable austauschen), und die Settings probieren. 4.5 bringt eine Menge Features die die Produktivität der Mitarbeiter erhöhen .

If you use PnP with the PBX, the PBX provisions the firmware link. You can set it in the PBX admin mode under settings/pnp. Just copy the link there and reboot the phone. If you want to remotely reboot (registered) phones, you can do that from the domain account list, at the bottom there is a item to reboot the selected phones.

snom ONE gibt es auch als virtual appliance http://downloads.snom.net/snomONE/virtual_appliance/snomONE-CentOS-appliance.zip.

Right. It is important that on the uplink you send the packets in the right order; then when they are in the hands of the service provider, they are all the same again. I have heard that when the kids come home around 3 PM in the afternoon, even the backbone starts dropping packets because of all the games going on and there is very little that you can do about it except building up your own network.

Yes. The thing that does not work (compared to pbxnsip versions) is the automatic generation of provisioning files for non-snom devices.

Probably you are experiencing the "NAT" problem. SIP does not work like HTTP and you might have to go through a complicated setup if your server is not running on a public (routable) IP address. See http://wiki.snomone.com/index.php?title=Server_Behind_NAT for more information.