asterisk: Introduce test TC_ims_call_mt
This test validates scenario where a call for Asterisk
from IMS Core comes in, Asterisk forwards it to the only
local SIP UA registered, the call becomes established and
then finally IMS Core hangs up after one second.
Change-Id: I2af846f92b6ebca21d1286f5137cb004b6284ddf
diff --git a/asterisk/SIP_ConnectionHandler.ttcn b/asterisk/SIP_ConnectionHandler.ttcn
index cd50be6..05f872f 100644
--- a/asterisk/SIP_ConnectionHandler.ttcn
+++ b/asterisk/SIP_ConnectionHandler.ttcn
@@ -90,6 +90,8 @@
charstring local_rtp_addr,
uint16_t local_rtp_port,
+ /* Whether to expect Asterisk to re-INVITE to make RTP flow directly to peer. */
+ boolean exp_update_to_direct_rtp,
SDP_Message peer_sdp optional,
CallParsMT mt
}
@@ -107,6 +109,7 @@
sip_body := omit,
local_rtp_addr := local_rtp_addr,
local_rtp_port := local_rtp_port,
+ exp_update_to_direct_rtp := true,
peer_sdp := omit,
mt := t_CallParsMT
}