Key presses aren't working (DTMF)

Overview

DTMF - Dual-Tone Multi Frequency - is the technology through which a customer can press a button on their telephone to navigate your PBXtra 's MAIN menu.  Different trunks implement the DTMF technology in different ways - most often out-of-band (i.e. not carried inline with the audio information), but there are trunk types that pass DTMF inband (such as a VoIP trunk using the 'inband' DTMF option!). 

When a customer tells you that they cannot navigate your MAIN menu, your PBXtra is likely not configured to detect the DTMF tones in the same way that your carrier is passing those tones.  Typically, it is very easy to fix DTMF issues.  Fonality's Admin Panel provides all of the settings necessary to select from among the most popular VoIP trunk DTMF types: inband, rfc2833, and info.

If you have DTMF trouble, please reference the section below that matches your trunk-type.

VoIP trunks

Test your account internally

1. Log into your web interface and visit the Options --> voip page

2. Choose the voip account in question and click on the account name

3. Open the Advanced section

4. There are three different dtmf mode options

5. Change the dtmf mode to anything else

6. Click "Apply All Changes"

7. Place a test call

8. Repeat for all three types until you have found a setting that works

Contact your carrier

If none of the dtmf modes work, then contact your VoIP service provider.

RFC2833 Compliance

Many SIP service providers, such as http://Bandwidth.com , have moved their service to a fully RFC2833-compliant system, which works on asterisk-based systems running version 1.4.  Fonality's PBXtra product is based on Asterisk version 1.2.17.  You will need to contact your provider and let them know that you are using an earlier version of Asterisk in order to properly configure the DTMF tones sent by the provider to your phone system.

T1 and Analog trunks

Don't use VoIP ATA's

ATA (analog-telephone adapter) devices are boxes that hook up to the network as SIP clients and register to providers, such as Vonage. Converting a SIP trunk to analog and back to SIP again can break keypresses, as well as a host of other potential issues. If you have SIP service coming into an ATA, consider registering your PBXtra directly to your SIP service.

Relax DTMF

The Relax DTMF setting reduces how strictly the system checks a tone generated from a keypress.  By relaxing this check, the system will accept a wider range of tones for the same keypress as each phone and each carrier may present the tone with a slight variation.  You can enable Relax DTMF via the following instructions:

1. Login to the Admin Panel

2. Select the Options --> settings page

3. Set the Relax DTMF option to "yes"

4. Click "Apply All Changes"

5. Make sure to place a test call

Change the Volume

Sometimes PBXtra is unable to detect a caller's keypress because the volume of that keypress is too low.  Customers can adjust the receive volume for a PRI/Analog trunk via the following instructions:

1. Login to cp.fonality.com

2. Select the Options --> cards page

3. Choose your T1 card from the drop down (you will repeat this step on

4. for each card):

   1. Click "Edit" for each port and increase the "RX Gain" to 3

   2. Click "Apply All Changes"

   3. Make sure to place a test call

Your test call will either show improvement or make the situation worse.  In either case, you can adjust the volume again (up or down as the situation requires) and test again.  In this fashion, you will 'tune' the system's volume settings to perfection.

Contact your carrier

Ultimately, it is the carrier's responsibility to relay keypress signals. A common setup that some carriers will provide for a 'Flex T1' combined voice & data network is to bring in single or bonded T1 line(s) into an AdTran or Cisco switch that can convert frame relay (T1) to ethernet and act as a SIP client, converting SIP on the outside to analog or T1 signalling up to your PBXtra , which has been known to break keypresses. You will maximize the functionality of your system by running a dedicated T1 line into your system or registering directly to a SIP service provider.