Vodia PBX

Hmm. You could limit the number of messages in the voicemail to zero. That should have the same effect like a full mailbox.

What you can influence is the display. The PBX is able to change the Caller-ID so that it includes the ACD name, as well as the original caller-ID in brackets behint it. Audio is tricky. Of course it will delay the call setup procedure, which is generally a bad thing. But an interesting idea.

Yes, that is not a bug, supposed to be a feature. Background is that people usually when they forward VM as email, they keep them in "saved" mode in the mailbox, creating a "timebomb"... That is why the PBX automatically drops the oldest saved one to make room for a new one.

Ehhhh.... where is snom ONE in the game? Wrong forum?

Yes you need to tell the firewall that it must trust the pbxctrl.exe. I would not work on specific ports, because the PBX can open a lot of different ports (SIP UDP/TCP/TLS, HTTP TCP/TLS, DNS client, LDAP, SMTP client, TFTP, NTP, and a large range orf RTP ports). You can do it like this: Open "Windows Firewall" (not the advanced firewall) which is in the control panel. Then on the left upper side there is "Allow a program or feature through the firewall", click on it. Then you get a dialog "Allow programs to cimmunicate through Windows Firewall", where you can "add another program" (at the bottom). I usually select domain, home/work and public (all), you never know where the PBX needs to connect to (e.g. for sending an email out).

Hmm. Not the expert here.. There must be something on the phone web interface ...

Can you check what is in these files? Does it make sense? Maybe the phone cannot access those files.

I think that was a bug in the firmware of the phone, and has been fixed in newer versions. Can you try to upgrade that phone?

Do you see anything in the "generated" folder (in the working directory of the PBX)? If you don't have file system access, change the setting on the PBX so that it does not write into the generated folder, but instead shows the generated files in the web interface log (you can change this setting in admin/settings/pnp). That should help troubleshooting what is gonig on. It might have to do with the fact that the multicast provisioning picks usually the private LAN address; if you provision via multicast to the public LAN, then this might now work. But if you put in the IP address of the PBX into the phone's settings server, it should get to the PBX anyway and then start pulling down the config fle (make sure you have "trust MAC" turned on on the same page. If you are going through a router, the PBX will not be able to see the MAC address, then you have to put in a username/password on the snom 300, through the web interface.

No, you probably has to log in by two steps. First you log in as user by using the username/password from the domain, and then you have to elevate your permissions on the phone by using the "admin" PIN.

snom ONE includes a "mini" session border controller that takes care about remote NAT. Things are relatively easy when you have a routable ("public") IP address that you can use for the PBX. As soon as you don't have that luxury things are getting messy. There is a article on that topic on the old wiki at http://wiki.snomone.com/index.php?title=Remote_Phone_Setup. Once you have that done, you can automatically provision your remote phones. In a nutshell, you need to put the address of the server, the username and password into the phone's web interface and from then on it will automatically do all hte magic to get the phone working in a local or remote location.

There are global settings in the pbx.xml file that set the type of service/DiffSrv bits for RTP and SIP: tos_rtp (default is ef5) and tos_sip (default is cs5). When the PBX opens a socket, it calls the operating system to set the values.

Good point, the PBX team thought the same way but the phone team does not... IMHO a security glitch on the phones, as you can easily script the 10000 combinations for a 4-digit PIN code from the web interface.

On snom phones, dont configure such things on the phone. Use buttons instead on the PBX and automatically provision the device. The "dialog-state" that many phones (also snom supports that) has a very very limited features set and was not intended for PBX button emulation, and BTW also has a crazy overhead. That's why we introduced buttons a coupld of years ago. Check out http://kiwi.pbxnsip.com/index.php/Buttons if you want to know more about this.

The snom ONE PBX supports multiple registrations, so if you register two or more SIP devices to the same account, they will ring at the same time. Also, you can have your cell phone ring at the same time, all you need to do is enter the number either through the phone's XML menu or through the web interface of the PBX. The PBX has a feature called "hot desking", this might also be something that you want to check out. http://wiki.snomone.com/index.php?title=Extension_Redirection, an old link from pbxnsip days http://kiwi.pbxnsip.com/index.php/Hot_Desking

What version are you using? This sounds like there is a routing error with the ACK, and it should be solved for some time now. Here is a old link from the pbxnsip days on how to connect Exchange: http://kiwi.pbxnsip.com/index.php/Microsoft_Exchange, maybe this also helps you.

If you are using Windows, you should also definitevely check the personal firewall settings and make sure that the OS does not perform any magic. Also, I would recommend that you (if you are developing) install something like Wireshark to get a true picture on what is going on on the network layer.

These are valid examples for the SOAP trusted IP addresses: 127.0.0.1 192.168.10.0/24 192.168.10.0/24 10.0.0.0/8 127.0.0.1 You might also want to white-list those addresses, so the PBX does not automatically black-list them if it gets "suspicious".

Maybe there is a confusion with the address book? The PBX has a domain address book (which the hunt group is using), and it has extension address books. Problem could be in this area. Also there are settings in the hunt group (and also in the agent group) on how the PBX should present the call. Background information: Some people want to see the name of the hunt group in the display and dont care so much about the caller-ID. So the hunt group has a settings for that ("From-Header").

You can put something like 841 in the last stage of the hunt group, which means 8 + extension = go to mailbox of extension 41. Then in the 41 mailbox, you can setup the email, so that the PBX sends out a email with the WAV attachment.

Check out the domain.xml in the PBX file system. There you find the domain pnp password (which you will have to use for the web login on the phones) and also the PIN that you might need to get into admin mode on the phones.

Yes, there are (at least) two ways to achieve this. First you can set the list of account number(s) to something like "40 41 42" (list seperated by space). The first one will be the primary account. Many phones (such as snom phones) support multiple registration on one phone. Then you can register multiple times to seperate accounts. If you want to use PnP, you can put the phone's MAC into the accounts that should be provisioned on the phone. A lot of old PBX systems required something to dial an outbound number, primarily because those systems had to collect DTMF tones to find out what the user wants to dial. There the prefix was essentially a menu item that you could choose in the menu with the IVR "dial tone". On new system like VoIP, the endpoint is typically smart enough (e.g. by having a display and edit keys) to figure out what the user wants to dial, so there is no need for such a prefix. Also, the prefix is a pain in the neck when using address books or when calling back. Usually, incoming numbers dont have a prefix, and when you store them the device also does not add it. Then when you want to call them, there must be (complicated) rules to put something in front of it, no no. That is why in the PBX we decided that dialling should be pretty much like from a cell phone, where you dial the number tha you want to dial and thats it. The PBX does have one prefix: If you want to dial an extension mailbox, you can dial 8 + extension (this is a domain setting). You can put the 8xx into the last stage of the hunt group, and voila the call goes directly into the mailbox of that extension. What most people do is set up a dummy extension just for that shared mailbox. The mailbox also supports group mailbox functions, so that you can copy the message into regular mailbox accounts (I believe this is the setting "Allow Access for Extensions"). When one of those users reads or deletes the message, it will also be read/deleted in the copied instances (they share the same reference). In that case I would choose the second method above where you provision multiple accounts on one device.

You should use EITHER the SIP IP replacement list OR the IP Routing List. In your case, I would use the IP routing list.

snom decided to support REFER only from snom products. You need snom ONE green in order to have this feature available. If you have a old pbxnsip license, you can as well use a pbxnsip build which does not have this restriction.

Maybe you just want to use a IVR node instead? There you can define a pattern that makes the PBX hang up when the prompt has ended. For example, the DTMF match list "!E!-!" should do this job.