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The login is provided by the LDAP(Active Directory) server. Your username needs to be in the correct group within the LDAP server. Is this still the case? Is your admin account not locked in the LDAP server?

You can try wireshark to see if the QoS settings are correctly added to the voicepackets. Most of the time this is a QoS issue. Also on the G450 gateway there is an application where you can measure RTP streams. On page 378 of the following manual you can check how to configure and how to run...

Manual shows: Shows that the second leg of the NCT call is dropped due to a trunk hardware problem, or that a vector step is executed that returned and ISDN DISCONNECT message such as a busy vector step Is the provider supporting NCR: https://downloads.avaya.com/css/P8/documents/100180552

If the CM to the Microlog direct connection is not working you can also try a CM - SBCe - Microlog connection. Best is to have a devconnect certified connection for Avaya support purposes.

It should be in your list history. It could be that the logging has been overwritten. Also you can check change logging-levels to see what kind of logging is written to the history log

These numbers are being send by the provider. Do you have special services from the provider where you can dial with less digits?

Does the station has a coverage path? Probably it will follow the coverage path. You can't filter this based on transfer calls. You should wait and hangup the second line when this happens.

I'm installing Avaya Multimedia Messaging behind a SBC for Enterprise. We need to install the SMGR certificates on the SBCe for this to work. Does anymore know which certificates to install and how we can make this to work?

Or can we put it on the main screen. We can use this to search ldap for numbers, but it are multiple clicks to get there.

It looks like you have a loop on your trunk. The incoming trunk also goes out and the channel stays active. Maybe you can try to figure out what happens with your CDR tool

Hi, it is required to have a gateway for dsp resources. Otherwise you can't make a phone to register. Regards, Evert

When we are making outbound calls through authorization codes the authorization code is stored in the history log off the station. Can we disable this from being displayed? Thanks.

1: Do you have a gateway? You need to have a gateway in order to register a phone. 2: Do a change procr and make sure that the procr allows you to register phones and gateways.

You can do a list meas ds1 sum 1v03 to check if the board has errors. If you have errors you should ask at&t to check again.

Does anyone know if these features are already supported on SIP?

When I'm logged into my One-x communicator as SIP client and I receive a new voicemail my MWI is not going to burn. When I logout and login again it is going to burn. When I listen to all my messages my light doesn't turn off. When I logout and login again my light is off. What could this be?

Thanks

We have a new CM6.3 system. We have configured fe-mute and conf-disp button, but the fe-mute is not working. When we push the button the fe-mute is flashing. Does anyone know how we can solve this?

Does anyone know how to remove the option "My Pictures" from the 9641G phone screen? I already tried disabling the USB port.

Indead, only the Flare client and the client on iPhone can't be set through the settings file.