I’ve just set up asterisk and zoiper and am trying the hello world sample. Zoiper connects to my asterisk ok and when I call extension 100 the call is accepted but I hear no audio. From another thread, I learned that asterisk was trying to pass the audio to my public IP and that I might try disabling STUN to use my local network address. As soon as I disable stun when I make a call the call is connected and ends immediately.
This is a call log with stun disabled:
== Setting global variable 'SIPDOMAIN' to '192.168.1.46'
-- Executing [100@from-internal:1] Answer("PJSIP/6001-0000000f", "") in new stack
> 0x74213c70 -- Strict RTP learning after remote address set to: 192.168.1.36:8000
> 0x74213c70 -- Strict RTP switching to RTP target address 192.168.1.36:8000 as source
-- Executing [100@from-internal:2] Wait("PJSIP/6001-0000000f", "1) same = n,Playback(hello-world") in new stack
-- Executing [100@from-internal:3] Hangup("PJSIP/6001-0000000f", "") in new stack
== Spawn extension (from-internal, 100, 3) exited non-zero on 'PJSIP/6001-0000000f'
This is the log for calls that went through but no audio, with stun enabled.
== Setting global variable 'SIPDOMAIN' to '192.168.1.46'
-- Executing [100@from-internal:1] Answer("PJSIP/6001-00000000", "") in new stack
> 0x74214968 -- Strict RTP learning after remote address set to: 185.244.215.203:8000
> 0x74214968 -- Strict RTP qualifying stream type: audio
> 0x74214968 -- Strict RTP switching source address to 192.168.1.36:8000
-- Executing [100@from-internal:2] Wait("PJSIP/6001-00000000", "1) same = n,Playback(hello-world") in new stack
-- Executing [100@from-internal:3] Hangup("PJSIP/6001-00000000", "") in new stack
== Spawn extension (from-internal, 100, 3) exited non-zero on 'PJSIP/6001-00000000'
Any help or suggestions is appreciated.