We have deployed the following scenario:
An asterisk PBX over an MPLS link in a remote site with 100Mbps bandwidth.
Ip phones and voice gateway, deployed locally with the voice gateway (patton) interconnecting a PRI channel on the PSTN.
Ip phones and the voice gateway, are configured in reinvite mode, so they transmit the RTP audio traffic directly over LAN.
Everything is working fine, but when an outbound call is made (let’s say to a mobile phone over PSTN) and this one is transferred to a conference room (which resides on the PBX), the conversation in one-way, sistematically.
More specifically, the external user can hear the internal ones, but the internal user can’t hear the external one.
Asterisk version is 11.6