Hi,
I’m trying to do the following in a dial plan. I have a call that is coming in over DAHDI (ground start, not PRI or E&M). I have it set to start in the from-internal context. Once the call has connecting, the remote party needs to post several digits of dtmf. Based on this, we dial the appropriate sip extension on another machine.
My problem is that when the call connects to sip, i lose the audio path. I can see that the remote system is sending back an RTP stream to Asterisk, but I’m not able to hear it. If I remove the Background and just dial direct, it works fine.
[from-internal]
exten => _X.,1,SIPAddHeader(x-channel: ${CHANNEL}})
exten => _X.,n,Dial(SIP/${EXTEN}@172.40.4.25)
exten => s,1,Set(TIMEOUT(digit)=1)
exten => s,n,Set(TIMEOUT(response)=1)
exten => s,n,Background(silence/9)
Any ideas?
Ben