asterisk: Introduce TC_ims_call_mo
The test validates establishing and hanging up a MO call:
SIP-UA -> Asterisk -> IMS-CORE.
SYS#6782
Change-Id: I3c6d8c109c392fa6e1036dcb69a7abb90b22fec7
diff --git a/asterisk/SIP_ConnectionHandler.ttcn b/asterisk/SIP_ConnectionHandler.ttcn
index 41a29e4..a14dd03 100644
--- a/asterisk/SIP_ConnectionHandler.ttcn
+++ b/asterisk/SIP_ConnectionHandler.ttcn
@@ -241,7 +241,8 @@
"t=0 0\r\n" &
"a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics\r\n" &
"a=record:off\r\n" &
- "m=audio " & int2str(g_pars.cp.local_rtp_port) & " RTP/AVP 96 97 98 0 8 18 99 100 101\r\n" &
+ "m=audio " & int2str(g_pars.cp.local_rtp_port) & " RTP/AVP 8 96 97 98 0 18 99 100 101\r\n" &
+ "a=rtpmap:8 PCMA/8000\r\n" &
"a=rtpmap:96 opus/48000/2\r\n" &
"a=fmtp:96 useinbandfec=1\r\n" &
"a=rtpmap:97 speex/16000\r\n" &