DTMF to SIP client

Asterisk Asterisk Support
1

I created a selection menu performing well when I call directly to the asterisks. I set the server as a client of another asterisksa SIP. Now I can listen to the welcome Playback() but the Background () apparently no longer receives input digits.

sip.conf

register => aaa:password@pc/6000
extension.conf

exten => 6000,1,Answer(500) exten => 6000,n,SayNumber(999) exten => 6000,n,Goto(usr,s,1) #include "usr"
usr

exten => s,n,Background(hello-world) - I hear this exten => s,n,WaitExten(500) -after that hangup() exten => 866076,1,Playback(you-entered)

2

hi:
Please check the dtmfmode in sip.conf and make sure the terminal meets that.

3

Dtmfmode on the client(asterisk) and the server(asterisk) is set to auto RFC2833. Terminal, SIP phone (Ekiga) as I already wrote direct dialing works so it must be ok.
Both installations are asterisks versions from Debian Lenny(I think 1.4.21.2) in principle with the original configuration.

4

Normal dialing from a SIP phone is done in the SIP dialogue, not in the RTP stream. The complete number is sent at once, not as DTMF digits,or their proxies.

5

If I understand correctly "the connection does not mean that the DTMF is working, " but with direct connection menu works so DTMF works.
The question remains how to make DTMF work with redirection?