Vodia PBX
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Sure. There are currently three things: 1. Enable the TAPI service provider to show incoming calls 2. Use AJAX or something else to refresh the calls in the web interface whenever something changes 3. Have a native Windows application displaying buttons like on a hard phone.
IMHO 2. looks like easiest way at the moment and it would also enable a lot of potential other things. Once the PBX has a way to tell the web browser "hey there is something" we can do a lot of interesting things.
Opinions, hints?
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Are you using option 66 to point the devices to the configuration server?
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Unfortunately, the recording subsystem just opens the file without creating directories.
If you want to sort files into directories, you need to run a cron job that moves the file from time to time into the proper location...
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IF someone could catch the SIP traffic it would be extremly useful...
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Could it be one ring? Maybe the PBX is waiting for the caller-ID, which is sent between the first and the second ring.
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Sometimes things are quite easy
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Are you mixing UDP with TCP? Could be we found something there...
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Well, well...
QoS could be reasonable with layer 3 tags. But he is right, practically nobody is doing that today. That's what the PPT speaks about having a 2nd DSL line and use that one only for VoIP - QoS by having physically seperated lines. Seems the router industry still has a long way to go. Until then, pragmatic solutions will make it happen!
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Maybe because there is an overlap with *00 as in "dial the cell phone" :-)
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Yea, some phones already have that - but it probably makes sense to have this also with a feature code on the PBX.
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For 13 extensions that should be sufficient. Be careful with the IP addresses - one is enough, make sure the other four are not giving you unneccessary problems.
For good user experience on audio, see http://www.pbxnsip.com/download/qos.ppt.
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Are you using 6.5.12 (http://wiki.snom.com/Snom320/Firmware/Release_Notes#6.5.12_release)? Seems there was a fix for lost subscriptions.
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Works for me - I just set the three digit speed dial in the domain address book (manually add en entry). Then dial that number, works. Maybe the problem is that the dial plan on the phone triggers the enter button too early?
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Well, the "dialog-state" style still presents the caller-ID and that feature is still available.
The "buttons" style also provides the information, however the snom's don't display that (yet).
No matter what you do - the customers always complain :-))
But a privacy policy concept that goes beyond on/off is still not available.
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Well, you can still move that hunt group to a ACD. Then the agents can process the requests one by one.
Alternatively, consider some "tricks" for the agents to write down caller information on paper while answering calls. I mean it, I heared from a very professional call center they do that by color...
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Please give http://www.pbxnsip.com/download/pbxctrl-2.1.1.2205.exe a shot. There is a new setting in the Admin Settings/Ports section.
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Try it out...
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I heared some time ago that receptionist with a lot of calls associate calls with colors. They mark a key with a color. I thought that was an interesting idea.
A less stressing method is to use a ACD for the receptionist. Then he or she can process the calls one by one, and the PBX can even say the initial greeting "welcome to company xy, you are talking to abc".
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There is a procedure defined in http://wiki.pbxnsip.com/index.php/Trouble_Ticket_Processing.
The reason why we prefer public communication whenever possible is simply because it scales much better than 1:1 support. When 1:1 support is required, a login (with a description on how to reproduce the problem) makes it much more productive than looking at megabytes of log level 9 traces.
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Okay, lets keep a "Speed Dial" mode in the back of our mind - or use the one which is arleady available... But that one does not light up the LED. Might be not so diffilult to do that. Lets see if others also feel the need for that.
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Warning: 399 192.168.1.10 Password does not match
Seems like the phone has the wrong password.
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The transfer is a attended transfer: A call comes into the main number, put on hold, then transfered to voicemail by pressing 8+the extension.
Okay, for attended transfer we can reprocude the problem. Will be fixed in 2.1.1. Interestingly, an attended transfer into conference also did not work (which really makes sense as the user can enter the PIN code for the transferred party).
Workaround: blind transfer works. After the call is connected, press the Xfer button, then 8+ext, then Xfer again.
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A couple of questions: Did you perform an upgrade? If yes, what was the previous and what is the new version?
Are you performing an attended transfer or a blind transfer (or call deflection)?
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Internal calls do not run through the dial plan.
If there is a huge delay that usualy points to problem in the routing. Sometimes the SIP requests take a long way out to the Internet and then back through the firewall.
If you turn the SIP packet logging on you will see more about the routing of the SIP packets and where they come from. That might help to locate the problem.
TOS Settings
in General Setup
Posted
Well, if your QoS is relying on layer 3 then you should do it. If you are just running your traffic in the LAN, it practically does not matter.
See also http://wiki.pbxnsip.com/index.php/Installing_in_Windows and the link in there http://support.microsoft.com/kb/248611/en-us.