asterisk: pjsip.conf: Disable remote bridging between local SIP and IMS
While implementing a first ttcn3 test validating the MO call scenario (SIP-UA -> Asterisk -> IMS-CORE) [1] I was running into the scenario where, after the first SIP INVITE + 200 OK + ACK, Asterisk was sending a RE-INVITE to both parties to attempt to remotely bridge them (RTP traffic flowing directly between them without passing through Asterisk).
This happened in part because I'm so far configure A-LAW on both sides so asterisk figures out it can do so. I still need to change IMS-core to EVS only.
In any case, regardless of the codecs used, my understanding is that asterisk should never attempt remote bridging when using the "volte_ims" endpoint, since that network segment is separate from the local network where the local SIP UAs are located.
The "direct_media=no" option just addresses the issue; Asterisk no longer tries to re-invite after the call is established
