Changeset 1793 for pjproject/trunk/pjsip-apps/src/symbian_ua/ua.cpp

Timestamp:

Feb 14, 2008 1:39:24 PM (16 years ago)

Author:

bennylp

Message:

Ticket #475: ported GSM codec to Symbian OS

File:

• pjproject/trunk/pjsip-apps/src/symbian_ua/ua.cpp (modified) (3 diffs)

pjproject/trunk/pjsip-apps/src/symbian_ua/ua.cpp

@@ -33 +33 @@
 // 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"
 
 //
@@ -54 +54 @@
 // 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"
 
 //
@@ -369 +369 @@
         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);