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

Related: SYS#6782
Change-Id: I4edea96151b31f02bf292b43b757922389375429
---
M ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf
1 file changed, 29 insertions(+), 0 deletions(-)

diff --git a/ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf b/ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf
index 9400780..9ca85e5 100644
--- a/ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf
+++ b/ttcn3-asterisk-ims-ue-test/asterisk/pjsip.conf
@@ -106,6 +106,7 @@
 rewrite_contact=yes
 from_user=238010000090828
 from_domain=172.18.11.103
+direct_media=no
 
 [volte_ims]
 type=auth

