RTPEngine vs Freeswitch PSTN Issues

[M B]

Hi all.
Clearly no expert :)  
I am trying to route SIP calls via Linphone/MicroSIP -> Opensips/RTPEngine -> Freeswitch -> PSTN.
All webrtc calls work fine. SIP calls directly to PSTN also work fine. Phones registered directly on Freeswitch are ok too. However, SIP calls via Freeswitch to PSTN don't have any audio. Following is the SDP exchange at Freeswitch. This (to me atleast) appears to be an RTPEngine configuration issue. Any thoughts please?

https://pastebin.com/XtNYkQ8y

[M B]

Edit: Just fixed (firewall ports issue!)

[Salvinder Parhar]

Pastebin URL you posted is not accessible. Its showing as a private paste. 

Include your E2E SIP call flow log and OpenSIPS/RTPengine config.

[M B]

Salvinder,

Thanks a lot. It was a problem with firewall ports. Appreciate your help.