Asterisk 11.6 cloud


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

Kind Regards,

Sounds like the gateway is unable to route media to the the PBX. Note that the conference will invalidate directmedia (“re-invite”) setting and all media will make two hops over the cloud.