hi,
i have added ring start,ringback, and ring stop functions in my
program,
i have given all default value for that
and now my main function is:
int main(int argc, char *argv[])
{
int i;
pjsua_acc_id acc_id;
pj_status_t status;
pjsua_transport_id transport_id =-1;
//pjsua_transport_config tcp_cfg;
/* Create pjsua first! */
status = pjsua_create();
if (status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status);
/* If argument is specified, it's got to be a valid SIP URL */
if (argc > 1) {
status = pjsua_verify_sip_url(argv[1]);
if (status != PJ_SUCCESS) error_exit("Invalid URL in argv", status);
}
/* Init pjsua */
default_config(&app_config);
status =parse_args(argc,argv,&app_config,&uri_arg);
if(status!=PJ_SUCCESS)
return status;
app_config.cfg.cb.on_incoming_call = &on_incoming_call;
app_config.cfg.cb.on_call_media_state = &on_call_media_state;
app_config.cfg.cb.on_call_state = &on_call_state;
//pjsua_logging_config_default(&log_cfg);
app_config.log_cfg.console_level = 4;
status = pjsua_init(&app_config.cfg, &app_config.log_cfg,
&app_config.media_cfg);
if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
#ifdef STEREO_DEMO
stereo_demo();
#endif
/* Optionally registers WAV file */
for (i=0; i<1; ++i) {
pjsua_player_id wav_id;
unsigned play_options = 0;
if (app_config.auto_play_hangup)
play_options |= PJMEDIA_FILE_NO_LOOP;
status = pjsua_player_create(&app_config.wav_files[i], play_options,
&wav_id);
if (status != PJ_SUCCESS)
printf("Unable to create player\n");
//goto on_error;
if (app_config.wav_id == PJSUA_INVALID_ID) {
app_config.wav_id = wav_id;
app_config.wav_port = pjsua_player_get_conf_port(app_config.wav_id);
if (app_config.auto_play_hangup) {
pjmedia_port *port;
pjsua_player_get_port(app_config.wav_id, &port);
status = pjmedia_wav_player_set_eof_cb(port, NULL,
&on_playfile_done);
if (status != PJ_SUCCESS)
printf("Unable to get port for wav player\n");
}
}
}
printf("%d..............................................\n",app_config.tone_count);
/* Optionally registers tone players */
for (i=0; i<1; ++i) {
pjmedia_port *tport;
char name[80];
pj_str_t label;
pj_status_t status;
pj_ansi_snprintf(name, sizeof(name), "tone-%d,%d",
app_config.tones[i].freq1,
app_config.tones[i].freq2);
label = pj_str(name);
status = pjmedia_tonegen_create2(app_config.pool, &label,
8000, 1, 160, 16,
PJMEDIA_TONEGEN_LOOP, &tport);
if (status != PJ_SUCCESS) {
pjsua_perror(THIS_FILE, "Unable to create tone generator", status);
//goto on_error;
}
status = pjsua_conf_add_port(app_config.pool, tport,
&app_config.tone_slots[i]);
pj_assert(status == PJ_SUCCESS);
status = pjmedia_tonegen_play(tport, 1, &app_config.tones[i], 0);
pj_assert(status == PJ_SUCCESS);
}
/* Create ringback tones */
if (app_config.no_tones == PJ_FALSE) {
unsigned i, samples_per_frame;
pjmedia_tone_desc tone[RING_CNT+RINGBACK_CNT];
pj_str_t name;
samples_per_frame = app_config.media_cfg.audio_frame_ptime *
app_config.media_cfg.clock_rate *
app_config.media_cfg.channel_count / 1000;
/* Ringback tone (call is ringing) */
name = pj_str("ringback");
status = pjmedia_tonegen_create2(app_config.pool, &name,
app_config.media_cfg.clock_rate,
app_config.media_cfg.channel_count,
samples_per_frame,
16, PJMEDIA_TONEGEN_LOOP,
&app_config.ringback_port);
if (status != PJ_SUCCESS)
printf("Ringback tone can not create\n");
//goto on_error;
pj_bzero(&tone, sizeof(tone));
for (i=0; i<RINGBACK_CNT; ++i) {
tone[i].freq1 = RINGBACK_FREQ1;
tone[i].freq2 = RINGBACK_FREQ2;
tone[i].on_msec = RINGBACK_ON;
tone[i].off_msec = RINGBACK_OFF;
}
tone[RINGBACK_CNT-1].off_msec = RINGBACK_INTERVAL;
pjmedia_tonegen_play(app_config.ringback_port, RINGBACK_CNT, tone,
PJMEDIA_TONEGEN_LOOP);
status = pjsua_conf_add_port(app_config.pool, app_config.ringback_port,
&app_config.ringback_slot);
if (status != PJ_SUCCESS)
printf("unable to add ringback port\n");
//goto on_error;
/* Ring (to alert incoming call) */
name = pj_str("ring");
status = pjmedia_tonegen_create2(app_config.pool, &name,
app_config.media_cfg.clock_rate,
app_config.media_cfg.channel_count,
samples_per_frame,
16, PJMEDIA_TONEGEN_LOOP,
&app_config.ring_port);
if (status != PJ_SUCCESS)
printf("Ring can not create\n");
//goto on_error;
for (i=0; i<RING_CNT; ++i) {
tone[i].freq1 = RING_FREQ1;
tone[i].freq2 = RING_FREQ2;
tone[i].on_msec = RING_ON;
tone[i].off_msec = RING_OFF;
}
tone[RING_CNT-1].off_msec = RING_INTERVAL;
pjmedia_tonegen_play(app_config.ring_port, RING_CNT,
tone, PJMEDIA_TONEGEN_LOOP);
status = pjsua_conf_add_port(app_config.pool, app_config.ring_port,
&app_config.ring_slot);
if (status != PJ_SUCCESS)
printf("Unable to add Ring port\n");
//goto on_error;
}
/* Add UDP transport. */
{
//pjsua_transport_config cfg;
//pjsua_transport_config_default(&cfg);
//cfg.port = 5060;
status = pjsua_transport_create(PJSIP_TRANSPORT_UDP,
&app_config.udp_cfg, &transport_id);
if (status != PJ_SUCCESS) error_exit("Error creating transport",
status);
}
/* Set sound device latency */
pjmedia_snd_set_latency(app_config.capture_lat,
app_config.playback_lat);
/* Use null sound device? */
#ifndef STEREO_DEMO
if (app_config.null_audio) {
status = pjsua_set_null_snd_dev();
if (status != PJ_SUCCESS)
return status;
}
#endif
if (app_config.capture_dev != PJSUA_INVALID_ID ||
app_config.playback_dev != PJSUA_INVALID_ID)
{
status = pjsua_set_snd_dev(app_config.capture_dev,
app_config.playback_dev);
if (status != PJ_SUCCESS)
// goto on_error;
printf("Unable to open playback device\n");
}
/* Initialization is done, now start pjsua */
status = pjsua_start();
if (status != PJ_SUCCESS) error_exit("Error starting pjsua", status);
/* Register to SIP server by creating SIP account. */
{
pjsua_acc_config cfg;
pjsua_acc_config_default(&cfg);
cfg.id = pj_str("sip:" SIP_USER "@" SIP_DOMAIN);
cfg.reg_uri = pj_str("sip:" SIP_DOMAIN);
cfg.cred_count = 1;
cfg.cred_info[0].realm = pj_str("*");
cfg.cred_info[0].scheme = pj_str("digest");
cfg.cred_info[0].username = pj_str(SIP_USER);
cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
cfg.cred_info[0].data = pj_str(SIP_PASSWD);
status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
if (status != PJ_SUCCESS) error_exit("Error adding account", status);
}
printf("%d......................",app_config.wav_count);
/* If URL is specified, make call to the URL. */
if (argc > 1) {
pj_str_t uri = pj_str(argv[1]);
status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);
if (status != PJ_SUCCESS) error_exit("Error making call", status);
}
for (;;) {
char option[10];
puts("Press 'h' to hangup all calls, 'q' to quit");
fgets(option, sizeof(option), stdin);
if (option[0] == 'q'){
pjsua_destroy();
break;
}
if (option[0] == 'h')
pjsua_call_hangup_all();
}
but still i am not getting ring, i think i am not getting correct value of
tone_count and wav_count,
will you help for this....
Regards
New_bie
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090116/85cd6114/attachment-0001.html>
