I have the pjsua_conf_connect() /* Callback called by the library when call's media state has changed */ void on_call_media_state(pjsua_call_id call_id) { ??? pjsua_call_info ci; ??? pjsua_call_get_info(call_id, &ci); ??? if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) { ??? // When media is active, connect call to sound device. ??? pjsua_conf_connect(ci.conf_slot, 0); ??? pjsua_conf_connect(0, ci.conf_slot); ??? } } I tried running in debug mode and try to catch the value by doing this void on_call_media_state(pjsua_call_id call_id) { ??? pjsua_call_info ci; ??? pjsua_call_get_info(call_id, &ci); ??? if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) { ??? // When media is active, connect call to sound device. ??? int succes=pjsua_conf_connect(ci.conf_slot, 0); ??? int ok=pjsua_conf_connect(0, ci.conf_slot); ??? } } and watch the value of success and ok, the value is 0 which means the function call returns success. BR, Joseph --- On Tue, 7/29/08, Benny Prijono <bennylp at pjsip.org> wrote: From: Benny Prijono <bennylp@xxxxxxxxx> Subject: Re: HELP!! Voice Connection During A SIP Call To: "pjsip list" <pjsip at lists.pjsip.org> Date: Tuesday, July 29, 2008, 3:52 AM On Tue, Jul 29, 2008 at 1:47 AM, Joseph Maiquez <josephmaiquez at yahoo.com> wrote: My problem was no voice was transmitted during a phone call that's why I can't hear the voice of the person on the other network. I think someone had faced this problem before here is the link.. http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2008-June/003452.html I think that discussed different problem. ? Im sorry but i dont have any idea on how to transmit voice. ? This is what I did. My guess is probably you're missing the call to pjsua_conf_connect(). ? Call the pjsua_call_make_call function to begin the invite Sip Call Then on the other end, call the function pjsua_call_answer then the call is successfully established but no audio is enable. How can I enable the audio during the call? this is what I wanted to know.. Hope you can help me guys, a sample code on how to make it will be very useful.. But we already have the samples (simple_pjsua, etc.). What's wrong with them? ?-benny _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip at lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080729/7bca5dfc/attachment-0001.html