Urgent Help Needed - Call between Softphone phones


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?


It can be the settings on the soft phone (that would be my first guess). The other problem can be NAT. Are you behind NAT ?