Asterisk 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,

One-way audio issues are almost always a problem with RTP traffic getting lost through a NAT firewall. The ones that aren’t related to a NAT firewall are just RTP misconfiguration.

Since the one phone is the problem, the settings in the phone would be my first suspect. I’m guessing there are NAT settings for that phone (over and above the ones for the rest of the system) that are causing you problems.

Tell us more about your phone and tell us about your extension configuration. Also, the logs around a failed call might yield some interesting information that we can use to help you troubleshoot this more.