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Is there a way on the real time agent report to see the total talk time for a call even when the agent changes states? So for example, Agent is talking for 2 minutes, put the customer on hold, CMS now shows hold time which starts at 0:00 then when they go back to the customer the ACD time starts...

Does anyone know if there is a way to dial by name using multiple PBXs (Avaya + Cisco) in a similar way the Intuity does? The only thing I can think of is in the Intuity to do a call-answer treatment instead of transfer, but this option will not call the extension, only transfer to that...

OK I am making baby steps towards progress here... I finally got my phone past the VPN by turning off QoS (this was the only thing that worked) .... now once it gets to the internal network, it goes to get its internal IP address (which was statically set by the Cisco concentrator on the way in)...

Yes, it has the proper VPN firmware... could it be that it's looking for a VLAN and the broadband doesn't do this? How does it work for the rest of the world? I can put in the call server IP address but that still won't mean it will get an IP address for the phone, right?

They are definately all 0's.... the same phone will pull an IP from an internal DHCP... where in the firmware (which file) does it let you manually enter DHCP settings? I have been over and over the 46xx files to no avail... what am I missing here?

I have configured a 4610 with the VPN firmware, but when I test it from home it will not pull an IP address... gets stuck on DHCP: XX seconds.... I am plugging the phone into a broadband modem, which a PC can pull an IP address from just fine... how do the current users of this get around the...

OK I got it, it was user error!! You must say NO when it asks if you want to reset values, then YES to reset the phone.

Yes, however the problem is that when the phone is reset it loses the group setting and goes back to 0. I have a ticket open with Avaya but..... it might be awhile...

Although I haven't gotten there yet, the answer to the above question is in the guide the previous posters spoke of: SET NVWEBLMURL http:// XX.XX.XX.XX:8080/WebLM/ LicenseServer However, I am still stuck on one of the first and probably simplest steps... when I put a phone in group 876, and...

Here is the scenerio: Asterisk voice mail coming from an Avaya s8700, via qsig hunt group over T1 trunk. What I am stuck on is getting Asterisk to recognize the extension dialed. What I currently get is the caller's extension, the qsig voice mail number dialed by the hunt group (using AAR), but...

Are you sure you're using login SA? There are 2 customer logins for Audix, VM and SA. VM takes you directly to where you speak of <enter command> with no option of the menu. SA will take you to the menu first, then you have to choose to go to voice mail administration and can go back & forth.

As long as you have the GM's extension in the auto attd as guest greeting, all the attd has to do is press start, <auto attd>, start to complete the transfer.

Never heard of an attendant with no transfer capability, are you sure about that? If its a 302 console, they would use the start button to transfer the caller to your auto attd.... what kind of phone does the attd have?

Has anyone succesfully used trunk flash on a tie trunk before? We have a robbed-bit T1 to another company using an Excel pbx, who we send the call to, and depending on need, their IVR then sends a "hookflash" signal back, and depending on the timers on pg 3 of the trunk form, it then either...