No audio received when call recording with MixMonitor

I have asterisk 13 (from source) installed on a centos 6.6.

Everything works well apart from when recording calls/using mixmonitor via a sip trunk.
When the call is established, the receiving party can hear the caller but the caller cannot hear the
receiving party. Further to this, a blank call record file is created.

I have tried various ways of activating MixMonitor (with the macro below and without it) but the results are always the same - no audio received from receiving party.

My dialplan:
exten => _X.,1,NoOp()
exten => _X.,n,Macro(automon)
exten => _X.,n,Dial(SIP/${EXTEN}@provider,30)
exten => _X.,n,Hangup()


exten => s,1,Set(MONITOR_FILENAME=${STRFTIME(${EPOCH},%Y%m%d-%H%M%S)}-${CALLERID(num)})
exten => s,n,MixMonitor(${MONITOR_FILENAME}.wav,b)[/i]

Asterisk console output:
== Using SIP RTP CoS mark 5
– Executing [xxxxxxxxxx@from-internal:1] NoOp(“SIP/hontra-0000001a”, “”) in new stack
– Executing [xxxxxxxxxx@from-internal:2] Macro(“SIP/hontra-0000001a”, “automon”) in new stack
– Executing [s@macro-automon:1] Set(“SIP/hontra-0000001a”, “MONITOR_FILENAME=20150319-143023-hontra”) in new stack
– Executing [s@macro-automon:2] MixMonitor(“SIP/hontra-0000001a”, “20150319-143023-hontra.wav,b”) in new stack
== Begin MixMonitor Recording SIP/hontra-0000001a
– Executing [xxxxxxxxxx@from-internal:3] Dial(“SIP/hontra-0000001a”, “SIP/xxxxxxxxxx@provider,30”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/xxxxxxxxxx@provider
– SIP/provider-0000001b is making progress passing it to SIP/hontra-0000001a
– SIP/provider-0000001b answered SIP/hontra-0000001a
– Channel SIP/hontra-0000001a joined ‘simple_bridge’ basic-bridge <4df88d08-1d57-4d4d-8792-8766472cc927>
– Channel SIP/provider-0000001b joined ‘simple_bridge’ basic-bridge <4df88d08-1d57-4d4d-8792-8766472cc927>
– Channel SIP/hontra-0000001a left ‘simple_bridge’ basic-bridge <4df88d08-1d57-4d4d-8792-8766472cc927>
– Channel SIP/provider-0000001b left ‘simple_bridge’ basic-bridge <4df88d08-1d57-4d4d-8792-8766472cc927>
== Spawn extension (from-internal, xxxxxxxxxx, 3) exited non-zero on ‘SIP/hontra-0000001a’
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/hontra-0000001a

Can anyone help or put me in the right direction?


Managed to resolve this issue. It was nothing to do with the call recording function.
As i did a bit of research, most one way audio issues are caused by incorrect NAT settings and this was no different. I amended setting on the firewall and everything started working correctly.