Vodia PBX

AFAIK CNAM (which is not CNAME) can not be signaled on per call bases and must be set up somewhere deep in the carriers network. At least we are currently struggling that out number shows "Vodia Networks" (for weeks already). You can easily control the "display name" of the trunk in the trunk setting. Maybe just try out and see if it makes any difference before you spend too much time on something that you cannot control anyway through the trunk.

No at the moment it is more like the post-paid model. If domain admins change it, their bill will change accordingly.

We could strip down the current name recording and just ask for a key press as an additional mode.

This is for hosted customers who charge accounts based on their usage profiles. For example a mailbox cost less than a full-featured extension. This restriction is also reported in the daily usage report.

Well, it also need to create the files for the buttons. Did you do any customization of the templates? Anything in the html folder?

Well we have the screening function. You can have callers leave their name first (and press the # key) before the system would call the extension.

I have the feeling that the provisioning does not work properly. Can you check if the generated folder contains new files (generated when you did the provisioning)? I am not aware about changes in the buttons area; but there were changes in the provisioning. You probably have to open up the extensions for provisioning again. Also, use 5.0.10o to make sure you don't run into something that has already been fixed.

Serious?

We have made 5.0.10o: http://snomone.com/downloads/snomONE/version-5.0.10o.xml which contains a lot of fixes and is a release candidate.

Is there anything in the generated folder (working directory of the PBX)? There you can see what files have been generated specifically for this extension. Also on the phone, you should be able to navigate to the status menu where you can see what firmware is being used. If the provisioning worked fine, you should have 8.4.35 or 8.7.3.19, which is okay. I see that you have decided to use UDP transport layer for the phones. UDP is not suitable for buttons, please change it back to TLS and try again. Buttons requires at least TCP and if I remember correctly it might even require TLS to work properly.

Trunk: I would put some information into the display and account fields, so that the trunk is able to send something out when the header presentation has nothing better. "Adtran" and "adtran" would be okay. The outbound proxy should be in SIP URI format sip:192.168.1.2:5060; however what you have put in there is also okay. For inbound calls, you need to tell the PBX where to send the call. If the field is empty, the PBX will search for the account in the SIP Request-URI. If you can already receive inbound calls to the PBX then leave it as it is. Otherwise, just put a extension or auto attendant number into field in order to get this going. Regarding snom 320 provisioning, try to put only the IP address into option 66 of the DHCP server (as ASCII characters, not binary). Then the phone should default to TFTP and get its configuration from there.

There are several ways to get this done. If you don't have a country code you can do something like pattern 141* and replacement 141*. If you have a country code (which is always recommended) the PBX puts the dialed number in to the country format (xxxxxxxxxx for US domestic for example) which makes it hard to use prefixes. If you want to use prefixes in the US, you should think of calling a foreign country "Anonymia" with the country code 99. So then you would dial 01199xxxxxxxxxx (pattern 01199*) and the replacement 141*. Alternatively, you can use the PBX ability to set the CLIR status of the account. Then in the trunk you can use the conditional presentation of header information like this: {if clip false}{from}{fi clip false}{if clip true}141{fi clip true} (introduced in version 5.0.5 or so). This gives you total control about what to display to the service provider, and the user can control CLIP with the star codes.

Please contact support (https://www.snomone.com/support) and then we'll take care about it.

You probably have an older version where the name of the executable was still snomONEctrl or something like that. You need to log in on SSH level and then edit the /etc/init.d/snomone file; there you find the executable name. Change it to pbxctrl and you should be good to go. Reboot the system to make sure that the service automatically starts e.g. after a power loss.

Usually the phones don't check the server certificate. If the phones can read any response that it received on TLS, then the transport layer is not the problem here. Maybe you are using a old firmware?

A factor in this is also the firmware for the phones. For 7xx series, the latest release 8.7.3.19 is from November 2011; for the other phones we recommend 8.4.35. Hopefully we will get a firmware update for the phones that fixes a couple of issued found in the mean time.

If you have provisioned the phones with 5.0.10h then you don't need to open the accounts again. I would choose to reboot the phones, just to make sure that they really get the config. Also check if the files in the generated folder are updated after you have rebooted the phones.

Usually you put in a URL that point to the XML file. Try http://snomone.com/downloads/snomONE/version-4.5.1.xml as URL.

Dann scheint wohl mehr als nur das Netzteil kaputt zu sein... "RMA"

Maybe we should vote

Okay we tried that over here. Yealink is a little bit difficult tolerating time zone information. We will fix that in the next build (let me know if you want a build, with the OS that you are using).

Vermutlich ein Problem mit dem Netzteil. Leider ist die Stromversorgung bei diesem Server nicht unbedingt optimal. snom Berlin kontaktieren und ein Ersatz-Netzteil ordern...

I agree. It did work for the US East coast time zone. I guess we need to try UK time zone today.

There is a risk. You must trust your provider, and you must also potentially trust the termination providers that he is using (REFER can be sent through them if they don't filter them out). There is a reason why this setting is "off" by default. usually you need it only for internal equipment. Originally we introduced it for Microsoft Speed Server that always wanted to refer the calls to another location.

Thanks. Do you think it makes sense to add that for all Polycom phones?