Vodia support

I'm sorry but that's not support in CSV however you can use "presence" the values would be 4001,4002,4004 however it's up to the phone to subscribe to the extension this will just allow the PBX to send status. Watch the presence of the following extensions (* for all): This setting is similar to the previous setting, but it doesn't provision calls. Instead, it asks the device to subscribe to the presence state of the listed extensions. The system will forward the presence state of the phones, but the phones themselves will be responsible for generating the presence information.If you wanted to subscribe to to extension it's best to provisioning the phone and setup BLF on the buttons page.

It's possible for the user to change their ANI if they have admin or domain rights only. In user mode they don't see the ANI field so your safe.

Can you assign the AA dial plan to a particular extension and see if you can dial the extension from the other machine? If the PBX can call using that dial plan and talk to the other machine then we are half way there. If you get an error then there you'll need to post a SIP trace of the call. Best regards

Hi, you can use the attached .csv file to add all the extension you would like to modify. I have left the upper portion empty so you can add your own details on how you want it. http://plugvoip.com/Extensions.csv

We can run this scenario but in theory it should work. Have you assign the particular dial plan on the AA?

Hi, if you can log into the phone then that's a good start. Navigate to SIP 1. Enter your SIP Configuration obtained from your online PBX account. 2. Under Servers, look for Outbound Proxy. "Enter your Your Vodia PBX IP or DNS" Outbound Proxy can be the same. Example: Address: Vodia PBX IP or DNS Port: 5060 Transport: UDP only Local SIP Port: 5060 3. Click on submit 4. Click on Line, Under Line 1, Look for Identification Type in the Following Information in their respective fields: Address: Your Vodia PBX IP or DNS Auth User ID: Your Extension number Authentication Password: SIP password of your extension. 5. Under Line 1, Look for Server 1. Type in the Following Information in their respective fields: Address: Your Vodia PBX IP or DNS Port: 5060 Transport: UDP only Register: Your Vodia PBX IP or DNS Press submit.

Hi, I am sorry I don't understand your question can you re-phrase it? are you referring to the n the address book where your can click to dial a particular number?

You can console into the unit VIA, USB but still ask you for root password which is your log in for your PBX. You're best bet would be to flash the unit and install the PBX software. This link describes how to factory reset the unit. http://derekmolloy.ie/write-a-new-image-to-the-beaglebone-black/ Or send us the unit and we'll flash it for your but you'll be responsible for shipping cost. PM me if you want the address?

You can match the following. This will match the last 6 digits !^([0-9]{6})$!\1! or This will match the 1st 6 digits. !^([0-9]{6})!\1!. Here is some other possibilities. https://twitter.com/vodiaPBX/status/503911838166573056 http://http://www.vodia.com/documentation/ivrnodes

I'll try my best to answer your question. If you want to limit the number of call to a particular extension you can set a value of 1 under the registration setting in the extension edit page.

If I understand correctly, you want to dedicate a particular trunk to use with join.me? and you want to transfer to outside user or do you want to keep it internally?

So, Trunk---->AA----> Option 1 or Option 2------->If they choose Option 2----->HG--->Final stage 8500 for example, it should send the call to VM can you try it on your deployment. Below is my setup and works just fine.

Hi, Is the HG using the correct dial plan?

Hi, You can use an IVR for this scenario upload your audio then add an E on the DTMF setting. After the user reaches the IVR, it will then play a message and hang up after the message is played.

You mentioned a few scenarios. What is the call flow here I am confused. You want the incoming call to flow into the AA they choose option 2 ------> Goes to a HG-----> Phones ring-----> No one picks up------->Then Goes to VM? Or You want the user to leave a VM when they reach the AA?

Hi, There isn't any setting on the extension for outgoing faxes, this will be between the PBX and the trunk provider. The ATA would be the one providing the T.38 protocol, we will just transmit it out to the trunk provider until the transaction is complete. I usually tell user to register the ATA to the trunk provider which would eliminate an extra step if it's a registration setup.

This is pretty old not sure if it will help but it's worth a shot. http://kiwi.pbxnsip.com/index.php/Microsoft_Exchange

Can an Agent be logged into as an agent and have calls routed by using either preference from user agents, Random, ring longest idle first and then also be an agent for After hearing ringback? Yes It should not but if the agent is on a call he will simply hear a beep, the caller will hear ringback as well but the call will stay in the ACD until the next step is giving under the "Caller in queue" also you will have to keep in mind of the users mailbox timeout. A loop in the system is when a call is sent back to the source, example AG---->redirect the to the same AG ----->redirect the to the same AG ect The PBX has a setting for this in the pbx.xml that will allow 11 loops after that it will disconnect the call, this setting is set to protect the PBX integrity at the final 11th loop the PBX will disconnect the call.

For admin setting controls Admin----System settings For user control settings. User settings. located in admin--->Customize---->User Page Control.

Hi, How many numbers do you have with callcentric? It's possible to create extension in callcentric and map those numbers to those accounts and create trunks with them.

Hi, You'll need to send a fax transmission to the extension in order for the PBX to play the FAX tone otherwise nothing will happen. If you would like to test the fax feature you can go to faxzero.com and send 1 free fax to your PBX. Make sure you're provider supports FAX t38. Lastly make sure your email is setup correctly and that you can actually receive email from the PBX.

Check under the domain -general setting. The setting is called "Transfer reminder after" and there is a drop down that you can choose a value as well.

Typically the calling card behaves the same as you would call your own extension but you won't get the prompt from the PBX to dial 1 to make an outbound call. On the calling card account there is an authentication mechanism that will prompt the user for the PIN, once entered they can make an outbound call.

what happens if you remove the area code from the domain do your local call parse through? Does it show your number without the area code?

Have you tried the various settings under the "Rewrite global numbers drop-down?