void MainWin::on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id,
pjsip_rx_data *rdata)
{
PJ_UNUSED_ARG(acc_id);
PJ_UNUSED_ARG(rdata);
if (currentCall_ != -1) {
pjsua_call_answer(call_id, PJSIP_SC_BUSY_HERE, NULL, NULL);
return;
}
emit signalNewCall(call_id, true);
pjsua_call_info ci;
char status[80];
pjsua_call_get_info(call_id, &ci);
snprintf(status, sizeof(status), "Incoming call from %.*s",
(int)ci.remote_info.slen, ci.remote_info.ptr);
showStatus(status);
}
void QPhoneHandler::onAccept()
{
pjsua_call_answer(callId, PJSIP_SC_OK, nullptr, nullptr);
}
void QPhoneHandler::onCancel()
{
pjsua_call_answer(callId, PJSIP_SC_TEMPORARILY_UNAVAILABLE, nullptr, nullptr);
}
static LRESULT CALLBACK WndProc (HWND hWnd, UINT message, WPARAM wParam, LPARAM lParam)
{
int wmId, wmEvent;
switch (message) {
case WM_KEYUP:
if (wParam==114) {
wParam = ID_MENU_CALL;
} else if (wParam==115) {
if (g_current_call == PJSUA_INVALID_ID)
wParam = ID_EXIT;
else
wParam = ID_MENU_DISCONNECT;
} else
break;
case WM_COMMAND:
wmId = LOWORD (wParam);
wmEvent = HIWORD (wParam);
if (wmId == ID_BTN_ACTION)
wmId = g_current_action;
switch (wmId) {
case ID_MENU_CALL:
if (g_current_call != PJSUA_INVALID_ID) {
MessageBox (NULL, TEXT ("Can not make call"),
TEXT ("You already have one call active"), MB_OK);
}
pj_str_t dst_uri;
wchar_t text[256];
char tmp[256];
pj_status_t status;
GetWindowText (hwndURI, text, PJ_ARRAY_SIZE (text));
pj_unicode_to_ansi (text, pj_unicode_strlen (text),
tmp, sizeof (tmp));
dst_uri.ptr = tmp;
dst_uri.slen = pj_ansi_strlen (tmp);
status = pjsua_call_make_call (g_current_acc,
&dst_uri, 0, NULL,
NULL, &g_current_call);
if (status != PJ_SUCCESS)
OnError (TEXT ("Unable to make call"), status);
break;
case ID_MENU_ANSWER:
if (g_current_call == PJSUA_INVALID_ID)
MessageBox (NULL, TEXT ("Can not answer"),
TEXT ("There is no call!"), MB_OK);
else
pjsua_call_answer (g_current_call, 200, NULL, NULL);
break;
case ID_MENU_DISCONNECT:
if (g_current_call == PJSUA_INVALID_ID)
MessageBox (NULL, TEXT ("Can not disconnect"),
TEXT ("There is no call!"), MB_OK);
else
pjsua_call_hangup (g_current_call, PJSIP_SC_DECLINE, NULL, NULL);
break;
case ID_EXIT:
DestroyWindow (hWnd);
break;
default:
return DefWindowProc (hWnd, message, wParam, lParam);
}
break;
case WM_CREATE:
OnCreate (hWnd);
break;
case WM_DESTROY:
OnDestroy();
CommandBar_Destroy (hwndCB);
PostQuitMessage (0);
break;
case WM_TIMER:
pjsua_handle_events (1);
break;
default:
return DefWindowProc (hWnd, message, wParam, lParam);
}
return 0;
}
/*
* main()
*
* argv[] contains the registration information.
*/
int main(int argc, char *argv[]) {
pj_status_t status;
int loglevel = 0;
if (argc < 4) {
usage(argv[0]);
exit(1);
}
if (argc > 5){
sscanf(argv[5], "%d", &loglevel);
}
pjsua_acc_id acc_id;
/* Create pjsua first! */
status = pjsua_create();
if (status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status);
/* Init pjsua */
{
pjsua_config cfg;
pjsua_logging_config log_cfg;
pjsua_media_config media_cfg;
// This will prevent the SIP stack from trying to switch to TCP.
// It will prevent the stack from giving "Transport unavailable" errors.
// http://trac.pjsip.org/repos/wiki/Using_SIP_TCP
pjsip_cfg()->endpt.disable_tcp_switch = PJ_TRUE;
pjsua_config_default(&cfg);
cfg.cb.on_incoming_call = &on_incoming_call;
cfg.cb.on_call_media_state = &on_call_media_state;
cfg.cb.on_call_state = &on_call_state;
cfg.cb.on_call_tsx_state = &on_call_tsx_state;
pjsua_logging_config_default(&log_cfg);
log_cfg.console_level = loglevel; // 0 = Mute console, 3 = somewhat useful, 4 = overly verbose.
pjsua_media_config_default(&media_cfg);
status = pjsua_init(&cfg, &log_cfg, &media_cfg);
if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
}
/* Add UDP transport. */
{
pjsua_transport_config cfg;
pjsua_transport_config_default(&cfg);
sscanf(argv[4], "%d", &cfg.port);
status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);
if (status != PJ_SUCCESS) error_exit("Error creating transport", status);
printf("Created UDP transport on port %d.\n", cfg.port);
}
/* 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. */
{
char reg_uri_buf[80] = "sip:";
char uri_buf[80] = "sip:";
pjsua_acc_config cfg;
char* username = argv[1];
char* password = argv[2];
char* domain = argv[3];
strcat (uri_buf, username);
strcat (reg_uri_buf, domain);
strcat (uri_buf, "@");
strcat (uri_buf, domain);
pjsua_acc_config_default(&cfg);
cfg.id = pj_str(uri_buf);
printf("Registering: %.*s.\n", (int)cfg.id.slen, cfg.id.ptr);
cfg.reg_uri = pj_str(reg_uri_buf);
cfg.cred_count = 1;
cfg.cred_info[0].realm = pj_str(domain);
cfg.cred_info[0].scheme = pj_str("Digest");
cfg.cred_info[0].username = pj_str(username);
cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
cfg.cred_info[0].data = pj_str(password);
cfg.register_on_acc_add = PJ_FALSE;
status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
if (status != PJ_SUCCESS) error_exit("Error adding account", status);
or_event(OR_READY, "Agent initialized..");
}
/* Start by muting the audio device. This a passive agent, and sound
is just wasted CPU time and uwanted side effects. */
pjsua_set_null_snd_dev();
fflush(stdout);
/* Wait until user press "q" to quit. */
for (;;) {
char option[256];
if (fgets(option, sizeof (option), stdin) == NULL) {
or_status (OR_ERROR, "EOF while reading stdin, will quit now..");
break;
}
/* Dial command. */
if (option[0] == 'd') {
pj_str_t uri = pj_str(&option[1]);
status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);
if (status != PJ_SUCCESS) {
or_status (OR_ERROR, "Could not make call");
}
else {
or_status (OR_OK, "Dialling...");
}
}
/* Register */
else if (option[0] == 'r') {
register_account(acc_id);
}
/* Unregister */
else if (option[0] == 'u') {
unregister_account(acc_id);
}
/* Enable autoanswer */
else if (option[0] == 'a') {
if (option[1] == '1') {
autoanswer[0] = true;
or_status(OR_OK, "Autoanswer enabled.");
} else {
or_status(OR_ERROR, "Invalid account.");
}
}
/* Disable autoanswer (manual answer) */
else if (option[0] == 'm') {
autoanswer[0] = false;
or_status(OR_OK, "Autoanswer disabled.");
}
/* Pickup incoming call, unsupported for now. */
else if (option[0] == 'p') {
if (current_call != PJSUA_INVALID_ID) {
pjsua_call_answer(current_call, 200, NULL, NULL);
or_status(OR_OK, "Call picked up.");
} else {
or_status(OR_ERROR, "No call to pick up.");
}
}
/* Hang up current call */
else if (option[0] == 'H') {
if (current_call != PJSUA_INVALID_ID) {
pjsua_call_hangup (current_call, 0,NULL, NULL);
or_status (OR_OK, "Hanging up current call...");
} else {
or_status(OR_ERROR, "No call to hang up.");
}
}
/* Full hangup.. */
else if (option[0] == 'h') {
pjsua_call_hangup_all();
or_status (OR_OK, "Hanging up all calls...");
}
/* Status */
else if (option[0] == 's') {
or_dump_status();
}
/* Exit application. */
else if (option[0] == 'q') {
break;
}
else {
or_status (OR_ERROR, "Unknown command:");
}
}
pjsua_destroy();
or_status (OR_OK, "Exiting...");
return 0;
}
void SipAccount::onIncomingCall(pjsua_call_id call_id, pjsip_rx_data *rdata) {
Logger::debug("SipAccount::onIncomingCall(call_id=%d)...", call_id);
PJ_UNUSED_ARG(rdata);
pj_status_t status = pjsua_call_set_user_data(call_id, this);
if (status != PJ_SUCCESS) {
Logger::error("pjsua_acc_set_user_data() failed (%s)", Helper::getPjStatusAsString(status).c_str());
}
pjsua_call_info ci;
pjsua_call_get_info(call_id, &ci);
#if 0
Logger::debug("local_info %s", pj_strbuf(&ci.local_info));
Logger::debug("local_contact %s", pj_strbuf(&ci.local_contact));
Logger::debug("remote_info %s", pj_strbuf(&ci.remote_info));
Logger::debug("remote_contact %s", pj_strbuf(&ci.remote_contact));
Logger::debug("call_id %s", pj_strbuf(&ci.call_id));
#endif
std::string display, number;
if (!getNumber(&ci.remote_info, &display, &number)) {
Logger::warn("invalid URI received '%s'", pj_strbuf(&ci.remote_info));
return;
}
std::string msg;
bool block = false;
if (number == "anonymous" or number == "") {
block = m_pPhone->isAnonymousNumberBlocked(&m_settings.base, &msg);
} else {
block = m_pPhone->isNumberBlocked(&m_settings.base, number, &msg);
}
Logger::notice(msg.c_str());
#if 0
// 302 redirect
Use pjsua_call_hangup() and put the destination URL in the Contact
header of the pjsua_msg_data.
pj_pool_t* pool = pjsua_pool_create("", 512, 512);
pjsua_msg_data msgData;
pjsua_msg_data_init(&msgData);
pj_str_t tmp;
pjsip_generic_string_hdr* hdr =
pjsip_generic_string_hdr_create(pool, pj_cstr(&tmp, "Contact"), pj_cstr(&tmp, "URI ...TODO"));
pj_list_push_back(&msgData.hdr_list, hdr);
// codes: http://de.wikipedia.org/wiki/SIP-Status-Codes
// enum pjsip_status_code...
pjsua_call_hangup(call_id, PJSIP_SC_MOVED_TEMPORARILY, NULL, &msgData);
pj_pool_release(pool);
#endif
if (block) {
// answer incoming calls with 200/OK, then we hangup in onCallState...
pj_status_t status = pjsua_call_answer(call_id, 200, NULL, NULL);
if (status != PJ_SUCCESS) {
Logger::warn("pjsua_call_answer() failed (%s)", Helper::getPjStatusAsString(status).c_str());
}
}
}
