Dear All, I successfully compile and run PjSIP on Symbian OS S60 SDK. Everything working fine. But one thing is I can't route my audio output to ear piece. After going through PjSIP website i came to know that, #define SND_USE_NULL 0 #define SND_USE_APS 0 Instead of using above, i have to use #define SND_USE_NULL 0 #define SND_USE_APS 1 in bld.inf file and symbain_ua.mmp file. Even though i changed as said above i am facing issue still. The issue is, ABC and XYZ are two users registered successfully to OpenSCR server, when ABC invite(calling using 'M' key) XYZ, XYZ accepts the invite using 'A' key then immediately app. closed ( mobile also restarts ) by giving PANIC Error : USER 21 (which means index out of range / bounce) at user ABC first and then XYZ. I faced the issue in following device. Nokia N95, Nokia E61, Nokia E51. I don't know where i am committing the mistake. Please experts help to sort this problem. -- Thanks And Regards, Srivatsan.D
