Nanang Izzuddin <nanang <at> pjsip.org> writes:
>
> Hi,
>
> I have no idea about CMdaAudioPlayerUtility API. Just FYI, pjsua
> sample app implements ringtone by creating wave player port and
> register it to the conference bridge (or audio switch when
> APS/VAS-Direct is used). Connecting this port to port 0 will play the
> ringtone, disconnecting it will stop the ringtone.
>
> BR,
> nanang
>
> On Thu, Feb 25, 2010 at 5:15 AM, Ricardo Antonio Ponce Gomez
> <ricardo2500 <at> hotmail.com> wrote:
> > Hi all.
> > I'm trying to play a ring tone when an incomming call is there. I'm using
> > the CMdaAudioPlayerUtility API and it just work fine the first time,?i can
> > hear a .wav audio file in the loudspeaker, but after I finish with this
> > call, and there's a second incomming call my ring tone it's gone. I'm dong
> > something like this:
> > _LIT(tono,"ring.wav");
> > CPlayer *player = CPlayer::NewL(tono);
> > //on_incoming_call callback
> > ....
> > pjsua_call_answer(call_id, 180, NULL, NULL);
> > player->Play();
> > ....
> > //on_call_state callback
> > ......
> > if (ci.state == PJSIP_INV_STATE_DISCONNECTED) {
> > if (call_id == g_call_id){
> > ?? ? ? ?g_call_id = -1;
> > ?? ? ? ?player->Stop();
> > ?? ? }
> > .....
> > Do you think i'm missing something? Is there any other better option to do
> > this?
> > I'm using VAS.
> > Thanks & Regards
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> >
> >
>
>
Hi,
Thanks Nanang for your help. Now I'm facing a similar problem creating the wav
player in this way:
pjsua_player_id player_id;
in on_incoming_call callback.
....
....
pj_str_t awav = pj_str ("C:\\Data\\ring.wav");
status =pjsua_player_create(&awav,0,&player_id);
if (status != PJ_SUCCESS)
console->Printf(_L("\n->error creando player\n"));
status=pjsua_conf_connect(pjsua_player_get_conf_port(player_id),0);
And then I disconnect the conference in on_call_state and in on_call_media_state
callbacks.
With this code I can hear the ringtone when I get an incomming call, the problem
is that when the call is finished the next time I get an incomming call my
ringtone is played in the earpiece instead of the Loudspeaker. Then, if I hang
up the call before answer it, the next time I have an incomming call the
Ringtone is played in the Loudspeaker again.
So I tried adding this before I connect the wav player to the conference:
pjmedia_aud_dev_route route;
route = PJMEDIA_AUD_DEV_ROUTE_LOUDSPEAKER;
status =
pjsua_snd_set_setting(PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE,&route,PJ_FALSE);
if (status != PJ_SUCCESS)
console->Printf(_L("\nError cambiando a altavoz\n"));
But I allways get the same result. Do you have any idea about what could be my
problem?
Thanks in advance for any help.
