asterisk: Implement and test SIP Digest Authorization
Related: SYS#6782
Change-Id: Ib469f1906927a3f246876040086ff115fbf4c032
diff --git a/sip/SIP_Tests.ttcn b/sip/SIP_Tests.ttcn
index aee1a82..4c6be1c 100644
--- a/sip/SIP_Tests.ttcn
+++ b/sip/SIP_Tests.ttcn
@@ -274,7 +274,9 @@
}
/* OSC -> SIP */
- as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm, *,
+ as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm,
+ tr_Via_from(tr_HostPort(sip_addr_ext.addr.nameAddr.addrSpec.hostPort)),
+ *,
"INVITE", 100, ?, "Trying", *));
alt {
@@ -328,7 +330,9 @@
SIP.clear;
/* 180 Ringing should not contain any SDP. */
- as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm, *,
+ as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm,
+ tr_Via_from(tr_HostPort(sip_addr_ext.addr.nameAddr.addrSpec.hostPort)),
+ *,
"INVITE", 180, ?, "Ringing", omit));
/* MSC -> OSC: After MT user has picked up and sent CC CONNECT */
@@ -342,7 +346,9 @@
}
/* OSC -> SIP: OSC confirms call establishment to SIP side */
- as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm, contact := ?,
+ as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm,
+ tr_Via_from(tr_HostPort(sip_addr_ext.addr.nameAddr.addrSpec.hostPort)),
+ contact := ?,
method := "INVITE", status_code := 200,
seq_nr := ?, reason := "OK",
body := expect_sdp_to_sip));
@@ -506,7 +512,9 @@
/* MSC -> OSC: Indicate GSM side release */
MNCC.send(ts_MNCC_REL_ind(cp.mncc_call_id, ts_MNCC_cause(0)));
/* OSC -> SIP: Confirmation to SIP side */
- as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm, *,
+ as_SIP_expect_resp(tr_SIP_Response(cp.comp.sip_call_id, sip_addr_ext, sip_addr_gsm,
+ tr_Via_from(tr_HostPort(sip_addr_ext.addr.nameAddr.addrSpec.hostPort)),
+ *,
"BYE", 200, cp.comp.sip_seq_nr, "OK", omit));
}