pbx support

All, We have a new version of the attendant console ready for beta. This is a Windows only application. Please take a try it out and provide us feedback. The details and download links can be found here - attendant console thanks snomONE team

Did you already check out the page - http://wiki.snomone.com/index.php?title=Server_Behind_NAT#Separation_by_IP_Address ?

This message just tells the phone about the MWI. The message that you posted telling the phone there are no messages waiting. That's all.

If this cell phone is associated with some extension, then if the remote party hangs up (not the cell phone), then PBX plays back some prompts to the cell phone (just to check if the user wants to perform anything else). If there is nothing else, the cell phone user can just hang up and the call should be disconnected without any delay.

Are these blind transfers? If yes, they can just hangup after performing the transfer. There was some change done on .1075 to "improve" the blind transfer experience that is possibly causing this confusion. If you are looking for yes or no answer for your question, then the answer is no (at the moment).

The "Trunk->Customer specific header" was set to empty in our tests (note that this header is used if the call is outbound on that trunk. In this case, we are doing the inbound). "Trunk->Accept Redirect" was to "No". The invite to the destination (either to an extension or to an external number) had "From" header as original "from" header and "Diversion" as the DID of the 600 (in this case 0311234567).

There is "pref_http_scheme" global setting (in the pbx.xml file) which is set to http by default. You can change it to https, if you want it that way. I assume you are talking about the user portal here. We uploaded the picture from the admin portal and logged into the web interface using https. The picture was was visible on the page. (the page source points to http location, but that should be ok)

We did a quick test here before knowing what exactly you are doing. Maybe there is something you can try before waiting for the new version. Set 0311234567 600 on the Account number(s): field for 600 (instead of 600 0311234567). Set "Assume that call comes from user:" as 600 The default "diversion_style" is set to tel under pbx.xml file. Change it to sip. You can change this Global setting using http://PBX IP address/reg_index.htm?save=save&diversion_style=sip Make a test call With this you should see the INVITE going out with Diversion: "display name of 600" <sip:0311234567@pbx.company.com>;reason=unconditional;screen=no;privacy=off Let us know how it goes while we are making sure that we are RFC compliant.

I think what we will do/verify if we are compliant to the RFC on this. Otherwise, it becomes very difficult to maintain. BTW, can you please explain this setup a bit?

Could you please follow http://wiki.snomone.com/index.php?title=Installing_Audio_Prompts and see it takes care of it?

We will add it to the next release if there is no easy way to do this.

PBX has the capability to detect the codec change during the call. Probably it is detecting one side has changed the codec in the middle of the call(or some NSE event). You can wireshark to verify if that is really the case.

Check out the reply "Server Admin > Access menu bar >mail:click "Allow all users and groups" under post https://discussions.apple.com/thread/2651030?threadID=2651030 Not sure if that would solve your problem.

That means there is no translations for SoHo found in the dictionary (which stores the translations for different languages). No biggie. It will still display SoHo has SoHo.

Is is coming always wrong or on some specific call scenario?

Other option is to increase the "Number of tolerated attempts" to 10 or 20 or so. This should take care of most of the problem. But at the same time, "unwanted" devices (scanners) will be blacklisted slowly.

Just FYI.. When 2011-4.2.1.4025 (Linux) was released, snom720 was not released. That's why running the older version has some issues like this one. That's why I asked the version question at the very beginning. Anyways, glad that you have configured the phone successfully.

Navigate to Admin->Web Page Control and see if any of these files is modified. If yes, you can reset to default by clicking one of the icons on that page. As I mentioned, it could happen if you just type "dom_status.htm" directly on the browser.

There seems to be some issue with PBX. There is one place when checking for the mailbox full, it is including even the recordings. We will fix it in the next version. For now, the way to get around the problem is to move the 'messages' folder entries for that particular extension away(bit tricky). modify the web page to have more than 100 and set that as the mailbox limit. This is much easier to do by changing the usr_core_mailbox.htm from the "Web Page Control".

Can you PM the login info (both web and the system)? Maybe there is an issue for specific account or something. We will check it out.

You seem to directly accessed dom_status.htm(and dom_logfile.htm), instead of clicking through navigation links Or you have modified of these files. This is not a major problem as long as you don't set/save anything from the web page. If you do it can be a problem.

Are you using 4.3 or 4.5 version of the PBX? If you are using 4.3, it may not work right off the bat. You will have to modify the pnp.xml and add couple of PnP files. If you are using 4.5 it should work.

We have seen this with the IE browser. If you use other browsers, you should be ok. Nothing to worry.

we will take care of that in the next version. If you want it before that, you can add blf1 and blf2 after blf to the usr_buttons.htm page.

did you get a chance to PM the login information (both web and RDP) ?