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Are there any phones whose display can show queue statistics, ie: calls waiting, etc, on the phone itself without too much trouble with Asterisk? Especially while the phone is in use (on a call)?

I testing gluster (the downloadable vmdk) under VMware Vsphere (ESX 4.0). The screen size is stuck at 800x600. I've tried increasing the size in the guest settings for vmware but that doesn't help. Some of the command buttons are off the screen and are inaccessible.

I have been unable to get callerid name passed from Cisco Callmanager over a SIP trunk to Asterisk. Only the number is displayed. Has anyone been successful getting callerid name?

In a CallManager environment (currently 4.0, moving to 6.1 in the next few months), can Asterisk completely replace Unity as a voicemail system? What works and what doesn't? MWI? Call Handlers? Does everything work via a SIP trunk? Who has done this and is willing to contact me? Thanks.

We have occasional problems with failed transfers. The PSTN call comes into Cisco Call Manager, then to asterisk over a SIP trunk and then to an asterisk controlled SIP phone. The SIP phone transfers back to a CallManager controlled SCCP phone which sometimes fails. Is there a wait to let CallManager handle the transfer instead of asterisk? I have a feeling asterisk is handling the traffic even

I currently have the following setup: exten => 2000,1,Playback(/var/lib/asterisk/sounds/Greeting) exten => 2000,2,Queue(Qabcdef|t) exten => 2000,3,Playback(/var/lib/asterisk/sounds/EveryonesBusy) exten => 2000,4,Hangup exten => 2000,103,Hangup What happens is, that the greeting in step one is played regardless if anyone is logged into the queue. So immediately after the greet, we

We're mostly Cisco CallManager with some SIP and Asterisk. I want someone at one of our locations to be able to dial and number and have Asterisk simultaneously dial several Call-Manager extensions which are set to auto-answer and talk into the phone creating a sort of paging system. We have informacast, but it is too cumbersome for the users. I know Asterisk can ring several phones at the

Does anyone know how to customize the order of the soft keys on a 7960 running SIP? All the documentation I could find is CallManager related. Specifically, I want to move the transfer function to the first set of buttons during a call.