Hi,
I’m using yate sip client with asterisk , but no audio , I used default asterisk sample config, just add
in sip.conf:
[8001]
dtmfmode=rfc2833
qualify=yes
secret=test8001
context=from-internal
host=dynamic
type=friend
[8001]
dtmfmode=rfc2833
qualify=yes
secret=test8001
context=from-internal
host=dynamic
type=friend
in extensions.conf:
[from-internal]
include => app-confbridge
include => app-extension
[app-confbridge]
exten => 64321,1,Answer()
exten => 64321,2,ConfBridge(64321)
exten => 64322,1,Answer()
exten => 64322,2,ConfBridge(64322)
[app-extension]
exten => _X.,1,Dial(SIP/${EXTEN})
Log: