How to add new call as conference PJSIP with siphon - ios

Question

Whats my project .
voice calling .
Whats library in my project .
Asterisk server (version 11.0)
pjsip 2.5.1
siphon for UI
My achievement
One-to-One call working fine
My issues :-
I need to implement add new buddy feature so that we can do conference .
Whats My problems
I am unable to get conference Voice call . scenarios is A called to B voice working fine ,but when B add new buddy C then B and C communicate but A and C Unable to communicate .
This my code which i am using for calling One-toOne
if (([[_label text] length] > 0) &&
([phoneCallDelegate respondsToSelector:#selector(dialup:number:)]))
{
_lastNumber = [[NSString alloc] initWithString: [_label text]];
[_label setText:#""];
}
else
{
_lcd.backgroundColor = [UIColor colorWithPatternImage:[UIImage imageNamed:#"lcd_top_simple.png"]];
[_label setText:_lastNumber];
[_lastNumber release];
}
}
Call.m file calling this below method .
status = pjsua_call_make_call(acc_id, &pj_uri, 0, NULL, NULL, call_id);
if (status != PJ_SUCCESS)
{
pjsua_perror(THIS_FILE, "Error making call", status);
}

Answer 1

There are missing some files Which is from "PJSIP" library. Please include the "PJSIP" library after compilation, or you may download the project from this link, this project have compiled "PJSIP" included.

Answer 2

static void on_call_media_state(pjsua_call_id call_id)
{
pjsua_call_info ci;
SiphonApplication *app = (SiphonApplication *)[SiphonApplication sharedApplication];
pjsua_call_get_info(call_id, &ci);
// PJ_LOG(3,(THIS_FILE,"on_call_media_state status %d count %d",
// ci.media_status
// pjmedia_conf_get_connect_count()));
/* FIXME: Stop ringback */
sip_ring_stop([app pjsipConfig]);
/* Connect ports appropriately when media status is ACTIVE or REMOTE HOLD,
* otherwise we should NOT connect the ports.
*/
if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE ||
ci.media_status == PJSUA_CALL_MEDIA_REMOTE_HOLD)
{
// When media is active, connect call to sound device.
pjsua_conf_connect(ci.conf_slot, 0);
pjsua_conf_connect(0, ci.conf_slot);
//pjsua_conf_adjust_rx_level(0, 3.0);
//pjsua_conf_adjust_tx_level(0, 5.0);
}
if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) { // When media is active, connect call to sound device.
pjsua_conf_port_id slotOne = ci.conf_slot;
// pjsua_conf_connect(slotOne, 0);
// pjsua_conf_connect(0, slotOne);
//mergeCalls=true;
int max=pjsua_call_get_count();
if (max==2) {
mergeCalls=true;
}
if (mergeCalls == true) {
pjsua_conf_port_id slotTwo = pjsua_call_get_conf_port(activeCallID);
pjsua_conf_connect(slotOne, slotTwo);
pjsua_conf_connect(slotTwo, slotOne);
// since the "activeCallID" is already talking, its conf_port is already connected to "0" (and vice versa) ...
} else {
activeCallID = call_id;
}
} else if (ci.media_status == PJSUA_CALL_MEDIA_LOCAL_HOLD) {
// … callSuspended(callID);
}
}