No RTP request from Asterisk/FreePBX


I’m unable to see any RTP being sent out from FreePBX.

But could see the below capture when rtp debug was enabled.

Service=“PJSIP”,EventVersion=“1”,AccountID=“123456789”,SessionID=“[email protected]”,LocalAddress=“IPV4/UDP/10.2X.YY.ZZ/5060”,RemoteAddress=“IPV4/UD
== Setting global variable ‘SIPDOMAIN’ to ‘20.XX.YY.ZZ’
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio CoS mark 5

Service=“PJSIP”,EventVersion=“1”,AccountID=“123456789”,SessionID=“[email protected]”,LocalAddress=“IPV4/UDP/”,RemoteAddress=“IPV4/U
– Executing [[email protected]:1] Set(“PJSIP/123456789-00000016”, “CALLERID(num)=+123456789”) in new stack
– Executing [[email protected]:2] Dial(“PJSIP/123456789-00000016”, “PJSIP/TRUNK1/sip:[email protected]:5060”) in new stack
– Called PJSIP/TRUNK1/sip:[email protected]:5060
– PJSIP/TRUNK1-00000017 is ringing
– PJSIP/TRUNK1-00000017 is ringing
– PJSIP/TRUNK1-00000017 answered PJSIP/123456789-00000016
> 0x7f33d800d2e0 – Strict RTP learning after remote address set to: 10.25.XX.YY:36208
> 0x7f33d8046450 – Strict RTP learning after remote address set to: 106.XX.YYY.ZZZ.12750
– Channel PJSIP/TRUNK1-00000017 joined ‘simple_bridge’ basic-bridge
– Channel PJSIP/123456789-00000016 joined ‘simple_bridge’ basic-bridge

Can anyone help me on this please.

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