Asterisk DTMF collection howto

Hi,

Been trying to to do a simple a DTMF collection System through Asterisk.

Stuck at getting the DTMF input Prompt.

Added the below in extensions.conf

[test]
exten => 123,1,Read(NUMBER,2)
exten => 123,2,SayNumber(${NUMBER})
exten => 123,3,Goto(1)

the error received is :
NOTICE[12078]: chan_sip.c:18160 handle_request_invite: Call from ‘’ to extension ‘123’ rejected because extension not found.

wat did i miss…?

Regards,
Rahul

So, you’re picking up a phone and dialing “123” and then it dumps that error to the CLI?

yep… The Phone in this case is X-Lite Soft Phone.

I can make user to user calls… i.e. 6000 to 6001.
but 6000 to 123 does not work.

Rahul

it looks to me that you have a problem in Sip.conf. I’m not sure but may be you are using different context for you SIP user.
would you please plot the sip configuration for the user you are using?

are you sure the context=test in sip.conf?

this the Line i found in sip.conf

context=default ; Default context for incoming calls

Rahul

change it to:
context=test, and reload Asterisk

changed that…

Also added this to the extensions.conf

[voicemenus]
exten = 7001,1,Goto(voicemenu-custom-1,s,1)

[voicemenu-custom-1]
include=default
exten=s,1,NoOp(PIN)
exten=s,2,Answer()
exten=s,3,Read(DIGITS,3)
exten=s,4,SayDigits(${DIGITS})
exten=#,1,Hangup

Am using the GUI for Configuration… and PIN is the voice menu i created through that.

Get the error as

chan_sip.c:18160 handle_request_invite: Call from ‘’ to extension ‘7001’ rejected because extension not found.

Oh man
since you are calling extension 7001 and it’s in context voicemenus, so you should make the context for you sip user equal to voicemenus.
context attribute for Sip,IAX users is actually the block in extensions.conf where any call from that user is going to be redirected to and processed there.