No dial tone - Grandstream BT100


Ive just got my asterisk system working with a Grandstream BT100, can make and recieve calls fine but the big problem is that there is no dial tone.

When I pick up the phone or press the answerphone button I do not hear a dial tone, ive trawled the web and gone through all the documentation for the phone and there is no option to turn it on or off.

Does asterisk make the dial tone?

Does anyone have any ideas? the only thing I can assume is that phones are broken and will need to be sent back as I can’t sell a telephone that doesnt have a dial tone.

Thanks for any help you may have.

the grandstream dialtone is connected to its registration. if it cannot register with * then you will not get a dialtone, and the ‘phone jack icon’ in the upper left of the screen will not light up.

Thanks for your help I figured it out in the end and documented what I did might be of help to someone else.

While setting up my home Asterisk system I found a problem with my BudgeTone’s (BTs) phones. When asterisk was configured correctly the phones had no dial tone when you picked up the handset. You could make and recieve calls fine, but there was not dial tone.


This appears to be caused by the phones not being registered to asterisk correctly. You may see an error about the phone not being registered in the asterisk console when you try to make or recieve a call.


I was using a configuration where each phone has a static ip address and an extension 100 - 104 programed onto the phone. This is the username used to register with asterisk.

Here is the example for the extension 100 phone:

  1. Logon to the config screen.

  2. Enter the “sip server” address.

  3. Enter the “SIP User ID” as 100.

  4. Enter the “Authenticate ID” as 100.

  5. Leave password blank.

  6. Enter “Name” as 100.

  7. “Sip Registration” should be set to “Yes”

  8. “Unregister on reboot” should be set to “No”

  9. Update the settings and reboot.

Now for the asterisk configuration. Open the sip.conf file and enter:

type=peer ; Friends place calls and receive calls
host=dynamic ; This peer register with us
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info

Note the host=dynamic line, even though the IP is static set this to dynamic or for some reason the phone won’t try to register with it. Ive used Soundpoint polycoms that when a static ip is used on the phone the sip.conf needs the host directive configured as static.

Restart asterisk and then restart the phone and you should now get the dial tone as the phone is now registered.