Ignore:
Timestamp:
Feb 14, 2008 1:39:24 PM (15 years ago)
Author:
bennylp
Message:

Ticket #475: ported GSM codec to Symbian OS

File:
1 edited

Legend:

Unmodified
Added
Removed
  • pjproject/trunk/pjsip-apps/src/symbian_ua/ua.cpp

    r1688 r1793  
    3333// Destination URI (to make call, or to subscribe presence) 
    3434// 
    35 #define SIP_DST_URI     "sip:user@192.168.0.11" 
     35#define SIP_DST_URI     "sip:100@pjsip.lab" 
    3636 
    3737// 
    3838// Account 
    3939// 
    40 #define HAS_SIP_ACCOUNT 0       // 0 to disable registration 
    41 #define SIP_DOMAIN      "domain" 
    42 #define SIP_USER        "username" 
    43 #define SIP_PASSWD      "password" 
     40#define HAS_SIP_ACCOUNT 1       // 0 to disable registration 
     41#define SIP_DOMAIN      "pjsip.lab" 
     42#define SIP_USER        "400" 
     43#define SIP_PASSWD      "400" 
    4444 
    4545// 
     
    5454// or STUN (for STUN see other settings below) 
    5555// 
    56 #define NAMESERVER      NULL 
    57 //#define NAMESERVER    "192.168.0.1" 
     56//#define NAMESERVER    NULL 
     57#define NAMESERVER      "192.168.0.2" 
    5858 
    5959// 
     
    369369        cfg.reg_uri = pj_str("sip:" SIP_DOMAIN); 
    370370        cfg.cred_count = 1; 
    371         cfg.cred_info[0].realm = pj_str(SIP_DOMAIN); 
     371        cfg.cred_info[0].realm = pj_str("*"); 
    372372        cfg.cred_info[0].scheme = pj_str("digest"); 
    373373        cfg.cred_info[0].username = pj_str(SIP_USER); 
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