Elastix how to setup ivr

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Make sure you can boot to your disk and allow the system to fully load before attempting to make configuration changes in the web management console. This takes you to a login screen that requires you to enter the web admin credentials configured earlier. These applets can be rearranged and also you can add or drop by selecting the Dashboard Applet Admin from the menu on the left side of the screen.

The first item to configure is the network settings found in the Network sub tab under the System menu. Most of the information seen above in figure 4 should auto-populate because of the steps taken during the initial setup. Select this link, change the system to Static and ensure that the proper IP address and network mask are in their respective fields.

Now would be the time to log into your router or gateway and reserve this address in DHCP or switch everything to static. Arguably, the most important part of the system is the extension configuration module. These are assigned to various devices from IP phones to softphones.

To configure your extensions, select PBX from the top menu and Extensions from the menu on the left. You will notice that there are several settings however, most I will leave untouched. As seen in figure 5, pick a display name for this extension. When you have a DID from a provider, this is how you will map it to an extension. This will also automatically create an inbound route for your DID.

Most of the items you see in figure 7 should remain at default. Change the secret to something a little easier to input but still secure Elastix creates one automatically but it is very lengthy. Enable your voicemail and choose a numerical password to access your messages as in figure 8 above. The Voicemail Password should be longer and more complex than the example above to prevent unwanted parties from accessing your mailbox.

The VmX Locater is a feature that provides call routing options. The user can input one of the numbers above to send the call to a different destination based on the definitions shown in Figure 8. After completing the setup of the extension, click the Submit button at the bottom of the page and then click on the red box that appears at the top of the page that you are directed to, that says Apply Config , to lock changes.

Take this time to configure as many additional extensions as needed. Next, return to the extension and optionally, click the new link near the top of the page to configure Follow Me settings.

The Follow Me feature is very useful if you are in and out of the office a lot, as it will ring additional phones defined in the Follow-Me List, based on the time set in the Initial Ring Time. Towards the bottom of the page, you can also specify a location to route the call if the call goes unanswered. Change these settings to meet your needs and be sure to save the settings. An example is shown in figure 9 below.

Your devices should now all be able to communicate with each other on the LAN. In the next step, I will connect our trunking provider in this case VoIP Innovations for origination and termination though feel free to select any provider you like. Keep in mind, the settings will likely be vastly different for another provider. Sometimes it is necessary to contact support for the trunk provider, in order to gather the credentials you need in order to connect to the outside world.

Under the PBX tab, go to the Trunks page on the left. You can set a number for the Outbound Caller ID or leave it blank depending on how your extensions and outbound routes are configured. Do not worry about adding dial patterns here as this will be configured in the final step. Finally, go to Outbound Routes, found on the left side of the PBX area and proceed to setup a new route.

Give this route a name and define your caller ID, if required. Make sure to configure dial patterns as seen in figure The only things that may really affect performance are sound conversion and tone detection.

As soon as you've done it, all Asterisk operation amounts to reading the file bytes from disk and sending them over network with just basic wrapping. There are no math, nothing except simple read-wrap-send operations.

Ask your provider to detect tones on his side and send them to you out of band , using RFC Tone detection is quite a CPU consuming operation, let them do it theirselves.

Overall traffic amounts to about 9. For a modern VPS there should be no problem at all. Asterisk rocks. For a few lines a simple P3 or better will do. Don't virtualise the PBX; Asterisk relies on pretty accurate timing. Trixbox includes it by default if you want a simple point and shoot solution. If your VoIP account supports multiple incoming lines then Asterisk will use them just fine.

You also need sufficient Internet bandwidth and decent QoS. For more than one line on a business system I would insist on a dedicated connection so you don't experience dropouts when users are accessing the net. IVR design in Asterisk is not difficult, but there is a bit of a "learning cliff" associated with getting your first Asterisk server up and running.

As stated by someone else, call quality is everything. Pay to have professional grade recordings done for your IVR prompts and your announcements. Ensure that you're using 64k codes such as uLaw and aLaw; GSM cellphone might be cheap on bandwith, but it breaks your customer expectations about quality. I strongly suggest that you put the IVR into it's own dial plan context and then direct calls into it.

That makes managing things like menu choices much easier. For each sub-set of options, use a different dial plan context. Try and keep your menu "shallow". If it takes more than three menu options to get the information your customer is looking for, they are very likely to hang up, or just press "0" to talk to a human.

That defeats the point of your IVR. If you are going to do something fairly cool with database look-ups, or account authentication or the like, I'd recommend using an "AGI" - Asterisk Gateway Interface - application.

If you didn't install the extra sounds earlier, now might be a good time to do that. Before we can use the demo menu above, we need to add an extension to the [docs:users] context to redirect the caller to our menu. Add this line to the [docs:users] context in your dialplan:.