Recent content by SupportDude

Do you have anything configured for the extensions making the call in 21-17 or 21-19? -SD- www.mysupportguy.net

Did you try Night Modes? Should be do-able. -SD- www.mysupportguy.net

I have one phone on the system that gives an error when I try to assign values in CME6. The error is "Some data could not be applied". The destination extensions I entered remain in the fields, but the text turns red. I've compared it to other phones that do have CME6 configured and everything...

Maybe you can just give them a config file instead. I usually just provide license migration XML file and a current config file. You'll get a LAC code from Avaya which you will use in the PLDS. It will migrate all of your current ADI licenses to PLDS licenses. This is why you have to provide...

This call originates from the Polycom 6000 (3rd Party SIP) conference phone. Telco prompts for LD Code input. If calling from a 9650 set, you get about 5 seconds of silence, then a tone, then a recording instructing you to input your authorization code. If calling from the Polycom, you hear...

Has anybody ever connected a Lifesize icon 600 SIP conference phone to an SV8500? Any integration tips appreaciated. Thanks! -SD- www.mysupportguy.net

You guys are doing great just to get the POS upgraded. I've been through 3 different USB drives and two UCMs so far, and still no joy. And two of the USBs are on the list provided by Avaya that reportedly work. I've done several of these without a problem, but going from 9.0 to 9.1 has really...

Waiting for firmware. Do you have manager open and configured as the TFTP server? -SD- www.mysupportguy.net

Can you tell us which call(s) in the trace experienced the problem? -SD- www.mysupportguy.net

Are the phones connected to a module in the control unit or an expansion? Is there anything in the display of the sets? -SD- www.mysupportguy.net

I believe you can if they are on the same subnet ... but why would you want to? -SD- www.mysupportguy.net

Can you post another trace with the INVITE message? On the monitor SIP tab enable; SIP (verbose) SIP Tx SIP Rx Post up the entire trace from beginning of call to the end. -SD- www.mysupportguy.net

Looks to me like you're getting the 200OK prior to receiving the SDP from the Cisco side. The 200OK ends the call setup. Basically, the Avaya doesn't know where to terminate his transmit path, so the Cisco user can't hear the Avaya user. The Avaya side sends the SDP in the "Invite" transaction...

@jlk53 .... I never got it to work. The provider was able to insert the main number, the outgoing CLID we wanted to send, on their side. Have you tried any of Intrigrants suggestions? -SD- www.mysupportguy.net

If you're on 9.0 you could configure a Park and Page action within voicemail, which will allow you to specify a destination if the call is not retrieved from park. -SD- www.mysupportguy.net