call mobile number

[megamarco83]

hi, is it possible to use domoticz to make a call when a switch is on?
i would like to configure my home alarm to call me if there is some issue.
is it possible to do that?
thanks

[waaren]

hi, is it possible to use domoticz to make a call when a switch is on?
i would like to configure my home alarm to call me if there is some issue.
is it possible to do that?
thanks

hi @Marco, Yes is possible. (and probably in more than one way)

My method: using dzVents domoticz.triggerIFTTT (available in dzVents >= 2.4.18 = domoticz >= 4.10603)
to a maker event in IFTTT connected to woopla.
woopla can make calls in your native language (Italian available) but is not a free service.

[DutchHans]

Yes it is... I have it working...
I have changed the code from Binerry.de
Take a look there... In my case the text is formatted, send to google.. a mp3 is returned...a sip call is initiated and the mp3 is played..

Cheers, Hans

[megamarco83]

hi thanks both for the answers.
@Hans could you please share the code that you modify from binerry.de?
thanks

[DutchHans]

Sure.. I am not near my server but I will asap.

Cheers, Hans

[DutchHans]

My modified sipcall.c file

Code: Select all

/*
=================================================================================
 Name        : sipcall.c
 Version     : 0.1 alpha

 Copyright (C) 2012 by Andre Wussow, 2012, [email protected]

 Description :
     Tool for making automated calls over SIP/VOIP with PJSUA library and eSpeak.

 Dependencies:
	- PJSUA API (PJSIP)
	- eSpeak
 
 References  :
 http://www.pjsip.org/
 http://www.pjsip.org/docs/latest/pjsip/docs/html/group__PJSUA__LIB.htm
 http://espeak.sourceforge.net/
 http://binerry.de/post/29180946733/raspberry-pi-caller-and-answering-machine
 
================================================================================
This tool is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.

This tool is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
Lesser General Public License for more details.
================================================================================
*/

// definition of endianess (e.g. needed on raspberry pi)
#define PJ_IS_LITTLE_ENDIAN 1
#define PJ_IS_BIG_ENDIAN 0

// includes
#include <fcntl.h>
#include <signal.h>
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <pjsua-lib/pjsua.h>

// some espeak options
#define ESPEAK_LANGUAGE "en"
#define ESPEAK_AMPLITUDE 100
#define ESPEAK_CAPITALS_PITCH 20
#define ESPEAK_SPEED 150
#define ESPEAK_PITCH 75

// disable pjsua logging
#define PJSUA_LOG_LEVEL 0

// struct for app configuration settings
struct app_config { 
	char *sip_domain;
	char *sip_user;
	char *sip_password;
	char *phone_number;
	char *tts;
	char *tts_file;
	int record_call;
	char *record_file;
	int repetition_limit;
	int silent_mode;
} app_cfg;  

// global helper vars
int call_confirmed = 0;
int media_counter = 0;
int app_exiting = 0;

// global vars for pjsua
pjsua_acc_id acc_id;
pjsua_player_id play_id = PJSUA_INVALID_ID;
pjmedia_port *play_port;
pjsua_recorder_id rec_id = PJSUA_INVALID_ID;

// header for new functions
static void default_configs(void);
void verify_arguments(int argc);
static void handle_help_request(const char* arg);
int check_sip_argument(int arg, int argc, char *argv[]);
int check_call_options(int arg, int argc, char *argv[]);
static void parse_arguments(int argc, char *argv[]);

// header of helper-methods
static void create_player(pjsua_call_id);
static void create_recorder(pjsua_call_info);
static void log_message(char *);
static void make_sip_call();
static void register_sip(void);
static void setup_sip(void);
//static void synthesize_speech(char *);
static void tts_google(char *);
static void usage(int);
static int try_get_argument(int, char *, char **, int, char *[]);

// header of callback-methods
static void on_call_media_state(pjsua_call_id);
static void on_call_state(pjsua_call_id, pjsip_event *);
static pj_status_t on_media_finished(pjmedia_port *, void *);
static void signal_handler(int);

// header of app-control-methods
static void app_exit();
static void error_exit(const char *, pj_status_t);

// main application
int main(int argc, char *argv[])
{
	// first set some default values
	default_configs();

	// verify number of arguments
	verify_arguments(argc);
	
	// parse arguments
	parse_arguments(argc, argv);
	
	if (!app_cfg.sip_domain || !app_cfg.sip_user || !app_cfg.sip_password || !app_cfg.phone_number || !app_cfg.tts)
	{
		// too few arguments specified - display usage info and exit app
		usage(1);
		exit(1);
	}
	
	// print infos
	log_message("SIP Call - Simple TTS-based Automated Calls\n");
	log_message("===========================================\n");
	
	// register signal handler for break-in-keys (e.g. ctrl+c)
	signal(SIGINT, signal_handler);
	signal(SIGKILL, signal_handler);
	
	// synthesize speech 
	//synthesize_speech(app_cfg.tts_file);
        tts_google(app_cfg.tts_file);	
	
	// setup up sip library pjsua
	setup_sip();
	
	// create account and register to sip server
	register_sip();
	
	// initiate call
	make_sip_call();
	
	// app loop
	for (;;) { }
	
	// exit app
	app_exit();
	
	return 0;
}

// helper for displaying usage infos
static void usage(int error)
{
	if (error == 1)
	{
		puts("Error, to few arguments.");
		puts  ("");
	}
    puts  ("Usage:");
    puts  ("  sipcall [options]");
    puts  ("");
    puts  ("Mandatory options:");
    puts  ("  -sd=string    Set sip provider domain.");
	puts  ("  -su=string    Set sip username.");
	puts  ("  -sp=string    Set sip password.");
	puts  ("  -pn=string    Set target phone number to call");
	puts  ("  -tts=string   Text to speak");
    puts  ("");
	puts  ("Optional options:");
	puts  ("  -ttsf=string  TTS speech file name to save text");
	puts  ("  -rcf=string   Record call file name to save answer");
	puts  ("  -mr=int       Repeat message x-times");
	puts  ("  -s=int        Silent mode (hide info messages) (0/1)");
	puts  ("");
	
	fflush(stdout);
}

// helper for parsing command-line-argument
static int try_get_argument(int arg, char *arg_id, char **arg_val, int argc, char *argv[])
{
	int found = 0;
	
	// check if actual argument is searched argument
	if (!strcasecmp(argv[arg], arg_id)) 
	{
		// check if actual argument has a value
		if (argc >= (arg+1))
		{
			// set value
			*arg_val = argv[arg+1];			
			found = 1;
		}
	}	
	return found;
}

// helper for logging messages to console (disabled if silent mode is active)
static void log_message(char *message)
{
	if (!app_cfg.silent_mode)
	{
		fprintf(stderr, message);
	}
}

// helper for setting up sip library pjsua
static void setup_sip(void)
{
	pj_status_t status;
	
	log_message("Setting up pjsua ... ");
	
	// create pjsua  
	status = pjsua_create();
	if (status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status);
	
	// configure pjsua	
	pjsua_config cfg;
	pjsua_config_default(&cfg);
	
	// enable just 1 simultaneous call 
	cfg.max_calls = 1;
		
	// callback configuration		
	cfg.cb.on_call_media_state = &on_call_media_state;
	cfg.cb.on_call_state = &on_call_state;
		
	// logging configuration
	pjsua_logging_config log_cfg;		
	pjsua_logging_config_default(&log_cfg);
	log_cfg.console_level = PJSUA_LOG_LEVEL;
		
	// initialize pjsua 
	status = pjsua_init(&cfg, &log_cfg, NULL);
	if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
	
	// add udp transport
	pjsua_transport_config udpcfg;
	pjsua_transport_config_default(&udpcfg);
		
	udpcfg.port = 5060;
	status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &udpcfg, NULL);
	if (status != PJ_SUCCESS) error_exit("Error creating transport", status);
	
	// initialization is done, start pjsua
	status = pjsua_start();
	if (status != PJ_SUCCESS) error_exit("Error starting pjsua", status);
	
	// disable sound - use null sound device
	status = pjsua_set_null_snd_dev();
	if (status != PJ_SUCCESS) error_exit("Error disabling audio", status);
	
	log_message("Done.\n");
}

// helper for creating and registering sip-account
static void register_sip(void)
{
	pj_status_t status;
	
	log_message("Registering account ... ");
	
	// prepare account configuration
	pjsua_acc_config cfg;
	pjsua_acc_config_default(&cfg);
	
	// build sip-user-url
	char sip_user_url[40];
	sprintf(sip_user_url, "sip:%s@%s", app_cfg.sip_user, app_cfg.sip_domain);
	
	// build sip-provder-url
	char sip_provider_url[40];
	sprintf(sip_provider_url, "sip:%s", app_cfg.sip_domain);
	
	// create and define account
	cfg.id = pj_str(sip_user_url);
	cfg.reg_uri = pj_str(sip_provider_url);
	cfg.cred_count = 1;
	cfg.cred_info[0].realm = pj_str(app_cfg.sip_domain);
	cfg.cred_info[0].scheme = pj_str("digest");
	cfg.cred_info[0].username = pj_str(app_cfg.sip_user);
	cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
	cfg.cred_info[0].data = pj_str(app_cfg.sip_password);
	
	// add account
	status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
	if (status != PJ_SUCCESS) error_exit("Error adding account", status);
	
	log_message("Done.\n");
}

// helper for making calls over sip-account
static void make_sip_call()
{
	pj_status_t status;
	
	log_message("Starting call ... ");
	
	// build target sip-url
	char sip_target_url[40];
	sprintf(sip_target_url, "sip:%s@%s", app_cfg.phone_number, app_cfg.sip_domain);
	
	// start call with sip-url
	pj_str_t uri = pj_str(sip_target_url);
	status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);
	if (status != PJ_SUCCESS) error_exit("Error making call", status);
	
	log_message("Done.\n");
}

// helper for creating call-media-player
static void create_player(pjsua_call_id call_id)
{
	// get call infos
	pjsua_call_info ci; 
	pjsua_call_get_info(call_id, &ci);
	
	pj_str_t name;
	pj_status_t status = PJ_ENOTFOUND;
	
	log_message("Creating player ... ");
	
	// create player for playback media		
	status = pjsua_player_create(pj_cstr(&name, app_cfg.tts_file), 0, &play_id);
	if (status != PJ_SUCCESS) error_exit("Error playing sound-playback", status);
		
	// connect active call to media player
	pjsua_conf_connect(pjsua_player_get_conf_port(play_id), ci.conf_slot);
	
	// get media port (play_port) from play_id
    status = pjsua_player_get_port(play_id, &play_port);
	if (status != PJ_SUCCESS) error_exit("Error getting sound player port", status);
	
	// register media finished callback	
    status = pjmedia_wav_player_set_eof_cb(play_port, NULL, &on_media_finished);
	if (status != PJ_SUCCESS) error_exit("Error adding sound-playback callback", status);
	
	log_message("Done.\n");
}

// helper for creating call-recorder
static void create_recorder(pjsua_call_info ci)
{
	// specify target file
	pj_str_t rec_file = pj_str(app_cfg.record_file);
	pj_status_t status = PJ_ENOTFOUND;
	
	log_message("Creating recorder ... ");
	
	// Create recorder for call
	status = pjsua_recorder_create(&rec_file, 0, NULL, 0, 0, &rec_id);
	if (status != PJ_SUCCESS) error_exit("Error recording answer", status);
	
	// connect active call to call recorder
	pjsua_conf_port_id rec_port = pjsua_recorder_get_conf_port(rec_id);		
	pjsua_conf_connect(ci.conf_slot, rec_port);
	
	log_message("Done.\n");
}

// synthesize speech / create message via espeak
/*static void synthesize_speech(char *file)
{
	log_message("Synthesizing speech ... ");
	
	int speech_status = -1;
	char speech_command[200];
	sprintf(speech_command, "espeak -v%s -a%i -k%i -s%i -p%i -w %s '%s'", ESPEAK_LANGUAGE, ESPEAK_AMPLITUDE, ESPEAK_CAPITALS_PITCH, ESPEAK_SPEED, ESPEAK_PITCH, file, app_cfg.tts);
	speech_status = system(speech_command);
	if (speech_status != 0) error_exit("Error while creating phone text", speech_status);
	
	log_message("Done.\n");
}
*/

static void tts_google(char *file)
{
log_message("Google TTS ... ");
int speech_status = -1;
char wget_command[200];
sprintf(wget_command, "wget -q -U Mozilla -O wget.mp3 \"https://translate.google.com/translate_tts?ie=UTF-8&tl=en&client=tw-ob&q=%s\"", app_cfg.tts);
speech_status = system(wget_command);
if (speech_status != 0) error_exit("Error in Google TTS API", speech_status);

char sox_command[200];
sprintf(sox_command, "sox wget.mp3 -r 22050 -b 16 -c 1 %s", file);
speech_status = system(sox_command);
if (speech_status != 0) error_exit("Error while converting mp3 to wav", speech_status);

log_message("Done.\n");
}


// handler for call-media-state-change-events
static void on_call_media_state(pjsua_call_id call_id)
{
	// get call infos
	pjsua_call_info ci; 
	pjsua_call_get_info(call_id, &ci);
	
	pj_status_t status = PJ_ENOTFOUND;

	// check state if call is established/active
	if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) {
	
		log_message("Call media activated.\n");
		
		// create and start media player
		create_player(call_id);
		
		// create and start call recorder
		if (app_cfg.record_call)
		{
			create_recorder(ci);
		}
	} 
}

// handler for call-state-change-events
static void on_call_state(pjsua_call_id call_id, pjsip_event *e)
{
	// get call infos
	pjsua_call_info ci;
	pjsua_call_get_info(call_id, &ci);
	
	// prevent warning about unused argument e
    PJ_UNUSED_ARG(e);
	
	// check call state
	if (ci.state == PJSIP_INV_STATE_CONFIRMED) 
	{
		log_message("Call confirmed.\n");
		
		call_confirmed = 1;
		
		// ensure that message is played from start
		if (play_id != PJSUA_INVALID_ID)
		{
			pjmedia_wav_player_port_set_pos(play_port, 0);
		}
	}
	if (ci.state == PJSIP_INV_STATE_DISCONNECTED) 
	{
		log_message("Call disconnected.\n");
		
		// exit app if call is finished/disconnected
		app_exit();
	}
}

// handler for media-finished-events
static pj_status_t on_media_finished(pjmedia_port *media_port, void *user_data)
{
	PJ_UNUSED_ARG(media_port);
	PJ_UNUSED_ARG(user_data);
	
	if (call_confirmed)
	{
		// count repetition
		media_counter++;
		
		// exit app if repetition limit is reached
		if (app_cfg.repetition_limit <= media_counter)
		{
			app_exit();
		}
	}
	
	pj_status_t status;
	return status;
}

// handler for "break-in-key"-events (e.g. ctrl+c)
static void signal_handler(int signal) 
{
	// exit app
	app_exit();
}

// clean application exit
static void app_exit()
{
	if (!app_exiting)
	{
		app_exiting = 1;
		log_message("Stopping application ... ");
		
		// check if player/recorder is active and stop them
		if (play_id != -1) pjsua_player_destroy(play_id);
		if (rec_id != -1) pjsua_recorder_destroy(rec_id);
		
		// hangup open calls and stop pjsua
		pjsua_call_hangup_all();
		pjsua_destroy();
		
		log_message("Done.\n");
		
		exit(0);
	}
}

// display error and exit application
static void error_exit(const char *title, pj_status_t status)
{
	if (!app_exiting)
	{
		app_exiting = 1;
		
		pjsua_perror("SIP Call", title, status);
		
		// check if player/recorder is active and stop them
		if (play_id != -1) pjsua_player_destroy(play_id);
		if (rec_id != -1) pjsua_recorder_destroy(rec_id);
		
		// hangup open calls and stop pjsua
		pjsua_call_hangup_all();
		pjsua_destroy();
		
		exit(1);
	}
}

// sets default values for app_cfg
static void default_configs(void)
{
	app_cfg.tts_file = "play.wav";
	app_cfg.record_call = 0;
	app_cfg.repetition_limit = 3;
	app_cfg.silent_mode = 0; 
}

void verify_arguments(int argc)
{
	if (argc == 1)
	{
		// no arguments specified - display usage info and exit app
		usage(1);
		exit(1);
	}
}

// parse and handle arguments
static void parse_arguments(int argc, char *argv[])
{
	int arg;
	for( arg = 1; arg < argc; arg+=2 )
	{
		// check if usage info needs to be displayed
		handle_help_request(argv[arg]);
			
		// check for sip domain, user or password
		if (check_sip_argument(arg, argc, argv) == 1)
		{
			continue;
		}
			
		// check for target phone number
		if (try_get_argument(arg, "-pn", &app_cfg.phone_number, argc, argv) == 1)
		{
			continue;
		}

		// check for text to speak, record call option, message repetition option, and silent mode option
		if (check_call_options(arg, argc, argv) == 1)
		{
			continue;
		}
	}
}

// if usage info is needed, provides it and exits
static void handle_help_request(const char *arg)
{
	if (!strcasecmp(arg, "--help"))
	{
		// display usage info and exit app
		usage(0);
		exit(0);			
	}
}

int check_sip_argument(int arg, int argc, char *argv[])
{
	// check for sip domain
	if (try_get_argument(arg, "-sd", &app_cfg.sip_domain, argc, argv) == 1)
	{
		return 1;
	}

// check for sip user
	if (try_get_argument(arg, "-su", &app_cfg.sip_user, argc, argv) == 1)
	{
		return 1;
	}
			
	// check for sip password
	if (try_get_argument(arg, "-sp", &app_cfg.sip_password, argc, argv) == 1)
	{
		return 1;
	}

	return 0;
}

int check_call_options(int arg, int argc, char *argv[])
{
	// check for text to speak
	if (try_get_argument(arg, "-tts", &app_cfg.tts, argc, argv) == 1)
	{
		return 1;
	}
			
	// check for record call option
	if (try_get_argument(arg, "-ttsf", &app_cfg.tts_file, argc, argv) == 1)
	{
		return 1;
	}
			
	// check for record call option
	if (try_get_argument(arg, "-rcf", &app_cfg.record_file, argc, argv) == 1)
	{
		app_cfg.record_call = 1;
		return 1;
	}
			
	// check for message repetition option
	char *mr;
	if (try_get_argument(arg, "-mr", &mr, argc, argv) == 1)
	{
		app_cfg.repetition_limit = atoi(mr); 
		return 1;
	}
			
	// check for silent mode option
	char *s;
	try_get_argument(arg, "-s", &s, argc, argv);
	if (!strcasecmp(s, "1"))
	{
		app_cfg.silent_mode = 1;
		return 1;
	}
}

My simple LUA file:

Code: Select all

commandArray = {}

receiver = "004912345678"


if devicechanged['CallSwitch'] == 'On' then

Message='This is the home automationsystem from Hans in Germany. Smokedetector has been activated!'


Message = string.gsub(Message, " ", "%%20")


os.execute('(sudo /home/pi/domoticz/scripts/SIP-Pi/sipcall -sd sip.yoursipprovider.nl -su xxxxxx  -sp xxxxxx  -pn '..receiver..'  -tts "'..Message..'"> /dev/null)&')

 commandArray['CallSwitch']='Off'

  end

return commandArray

Hope this helps,
Cheers, Hans