@Pentium5 wrote:
Hi all,
I have two locations running FreePBX 13.0.192.14 with Asterisk 13.16.
These locations are connected via PJSIP trunk over OpenVPN tunnel built between Asterisk servers.
Therefore, each Asterisk machine has two PJSIP transports: one on a physical interface for local endpoints, the other on a tunnel interface for the trunk.In the above scenario, SIP signalling works fine, but RTP is not flowing between remote endpoints.
During troubleshooting I noticed that once a connection is established (INVITE -> 200 OK -> ACK), the outgoing Asterisk reinvites the remote Asterisk with SDP of the local endpoint, thus trying to establish the direct media connection.
Of course, this scenario will not work as the remote Asterisk does not know anything about local network.
Adding static routes through tunnel interface will not resolve the issue as tunnelling is performed on the Asterisk servers, not on gateway routers.I have canreinvite=no in global PJSIP settings on both ends, but this parameter does not help.
I found out that Asterisk supports setting media flow mode per endpoint (parameter direct_media). When I added manually direct_media=no to pjsip.endpoint.conf, I got RTP streams flowing.However, I have not found a corresponding parameter in FreePBX, extensions. Moreover, FreePBX does not communicate direct_media setting to Asterisk (no such lines in generated pjsip.*.conf files), and the latter treats it as yes according to the defaults.
Having this issue open, I am asking for a piece of adivce on the following:
1. How can I disable direct media in FreePBX UI on an endpoint / extension level?
2. If (1) is not possible, how can I disable direct media mode entirely for the whole node - taking into account that canreinvite=no does not work as designed?I can provide configuration settings, packet traces and other supporting materials upon your request.
Thank you very much in advance.
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