I am a new asterisk user, currently following the guidelines on “Asterisk: The future of Telephony” to set up an Asterisk PBX system on a server.
I am able to place a call between two softphones (using SIP); one on the server and the other on a client PC. I can observe the softphones ring when their configured extensions are dialed, however, I don’t hear anything from the speaker after the call is established.
I feel there is a problem with my Real-time Transfer Protocol (RTP). I may not have configured the system to transfer the media stream after a call has been established. however, when I press the ACCEPT button again i can observe the Music on Hold being sent as RTP packets (when I type in rtp debug) and can hear it via the speaker
I have defined two contexts (in extensions.conf) with the Dial() application eg
exten => 234,1,Dial(SIP/234,20)
exten => 234,n,Hangup()
i would imagine that i should hear sounds from the other phone after the call is connected and when someone talks into the microphone. But I don’t know why I don’t hear anything after the call is initiated.
Can someone help me please?