Let me provide the details first:
* Asterisk 1.8.32 on CentOS behind the NAT firewall
* Two (2) SIP trunks with “canreinvite=no” and “directmedia=no”
If a call comes from either trunk and is bridged to a local extension there is never a problem with audio. The same is true for outbound calls on either trunk.
If an incoming call from Trunk A is forwarded to Trunk B there is a large percentage of the one-side audio calls.
Has anybody run into this kind of a situation?
Thank you, Vladimir