Kishore Kumar Lenka wrote:
> Thanks a ton....
>
> I have following queries . Hope you will help me..
>
> Suppose I have my sip server (OnDo sip server from Brekeke) running in my
> pc at 192.168.5.54.
>
> I have to call from "sip:192.168.5.78" to "sip:192.168.5.79" in this case
> do i need to add
>
> 1.
>
> #define SIP_DST_URI "sip:192.168.5.54:5060" //where the sip server is
> running
> #define HAS_SIP_ACCOUNT 1 // 0 to disable registration
> #define SIP_DOMAIN "192.168.5.54"
> #define SIP_REALM "kishore" // Real Machine name (my system name)
> #define SIP_USER "400"
> #define SIP_PASSWD "1234"
>
> complile->put this application into my device->and start calling from one to
> other.
First you'll need to use version 0.7 from the SVN trunk (you
mentioned earlier that you're using 0.5.10.x) so that you can put
asterisk ("*") as the credential's realm.
Then use the setting below:
#define HAS_SIP_ACCOUNT 1
#define SIP_DOMAIN "192.168.5.54"
#define SIP_REALM "*" // only works with 0.7 from SVN trunk
#define SIP_USER "400"
#define SIP_PASSWD "1234"
Then as the destination URI, put:
#define SIP_DST_URI "sip:500 at 192.168.5.54"
that is assuming that the other endpoint is registered to the proxy
with username "500".
Btw, have a look at the link in http://pjsip.org/trac Wiki page for
troubleshooting audio problems.
-benny
