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Ok, thanks. Customer doesnt really want to buy another solution, but is willing to spend some money on our programmers. Thats why im asking. ACIS SME | ACSS SME

Ascom et al? Sorry can't follow. So you saying register a 3rd party SIP ext. to the IPO, from that SIP Endpoint I can sent messages to DECT Handsets? ACIS SME | ACSS SME

Hi Forum, anyone ever tried to write a program (exe, script etc) for the sent message tool (HTTP site of the AIWS2)? A customer is asking if its possible, without buying the alarm server option. I looked at it with our software engineers and they need to know what the site does. We tried...

We are using the ESTOS TapiServer at a customer to do this. But I think this Tool is not avalible anymore. ACIS SME | ACSS SME http://www.kriesten.de

The Avaya TB 157 is just saying something about upgrading from 6.0. And there you have to do 8.0 first, then 8.1. ACIS SME | ACSS SME http://www.kriesten.de

I had that a few times now. Resetting the security settings was always the solution. I have no idea why this is happening. ACIS SME | ACSS SME http://www.kriesten.de

Did you try changing the extension to standard analogue phone? That worked for me. A trace from the call would be helpfull. ACIS SME | ACSS SME http://www.kriesten.de

That is a feature from R9. As far as I know, it is not possible with R8.1.. ACIS SME | ACSS SME http://www.kriesten.de

I can recommend ESTOS. It's a really good CTI software. ACIS SME | ACSS SME http://www.kriesten.de

sounds like you didn't put the ip/fqdn in the oxp server. you will find it in configuration/im ACIS SME | ACSS SME http://www.kriesten.de

Go to your PRI Line in the Manager. There you have the prefix options, there you will find the 00. ACIS SME | ACSS SME http://www.kriesten.de

What do you see in the Monitor? Do you see the incoming call? ACIS SME | ACSS SME http://www.kriesten.de

Hi dbal640, for the incoming: if your ITSP is sending the full number in his invite, like XXXXX579. You have to setup the SIP settings in every user (User/SIP) and put in his full number or you put a "*" in your SIP-URI in local URI, contact and alias. With that "*" the IPO accepts alle...

Use the command run ipoffice or run ipoffice verbose. I hope that will help you ACIS SME | ACSS SME http://www.kriesten.de

Did you tried reseting the phone? As I see it, it is just "forgetting" the IP fnformations you entered in the phone. It is not connecting to the IPO just yet. As soon as it does it will update. Do you have a chance to set up a DHCP? ACIS SME | ACSS SME http://www.kriesten.de

What is the error message when you are trying to login as a user on the login page? Do you get the login page? ACIS SME | ACSS SME http://www.kriesten.de

Thanks for the star! ACIS SME | ACSS SME http://www.kriesten.de

Another easy way: use the Avaya Routing switches and run the command "run ipoffice" on them. It will do everything for you. Link ACIS SME | ACSS SME http://www.kriesten.de

Well, 1 is true, 0 is false. The user can change that value. The user wants to setup different announcements which are played with different .wav files. ACIS SME | ACSS SME http://www.kriesten.de

Hi Tek-Tips Forum, we have a problem with on a customers site. The caller doesn't hear a busy tone when the called ext is speaking. The call just gets disconnected. We are routing the calls through the Voicemail Pro. procedure: incoming call routing -88886209 -> target: VM:209 Voiemail Pro...