Ticket #475: ported GSM codec to Symbian OS
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@1793 74dad513-b988-da41-8d7b-12977e46ad98
diff --git a/pjsip-apps/src/symbian_ua/ua.cpp b/pjsip-apps/src/symbian_ua/ua.cpp
index 3e33ca4..b80055f 100644
--- a/pjsip-apps/src/symbian_ua/ua.cpp
+++ b/pjsip-apps/src/symbian_ua/ua.cpp
@@ -32,15 +32,15 @@
//
// Destination URI (to make call, or to subscribe presence)
//
-#define SIP_DST_URI "sip:user@192.168.0.11"
+#define SIP_DST_URI "sip:100@pjsip.lab"
//
// Account
//
-#define HAS_SIP_ACCOUNT 0 // 0 to disable registration
-#define SIP_DOMAIN "domain"
-#define SIP_USER "username"
-#define SIP_PASSWD "password"
+#define HAS_SIP_ACCOUNT 1 // 0 to disable registration
+#define SIP_DOMAIN "pjsip.lab"
+#define SIP_USER "400"
+#define SIP_PASSWD "400"
//
// Outbound proxy for all accounts
@@ -53,8 +53,8 @@
// Configure nameserver if DNS SRV is to be used with both SIP
// or STUN (for STUN see other settings below)
//
-#define NAMESERVER NULL
-//#define NAMESERVER "192.168.0.1"
+//#define NAMESERVER NULL
+#define NAMESERVER "192.168.0.2"
//
// STUN server
@@ -368,7 +368,7 @@
cfg.id = pj_str("sip:" SIP_USER "@" SIP_DOMAIN);
cfg.reg_uri = pj_str("sip:" SIP_DOMAIN);
cfg.cred_count = 1;
- cfg.cred_info[0].realm = pj_str(SIP_DOMAIN);
+ cfg.cred_info[0].realm = pj_str("*");
cfg.cred_info[0].scheme = pj_str("digest");
cfg.cred_info[0].username = pj_str(SIP_USER);
cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;