myUtils.pm Frage

lammy · 22 Dezember 2016, 19:15:58

Hi Leute,

ich habe ein paar Probleme mit dem nachfolgenden Quellcode und den Fehlern in der Log Datei. Da ich mit Perl noch nicht so vertraut bin und es leider an der Syntax liegt hoffe ich das jemand Zeit hat der es aus dem FF kann und mir sagt was ich falsch gemacht habe. Ich habe schon ein paar "", (), {}, '' irgend wo mal gesetzt aber komme einfach nicht zum Ziel

Was soll das Script machen...
ich möchte über eine DOIF Funktion den Befehl

Code Auswählen

({voipcall("Ein Feuer ist ausgebrochen",'017*****')}) starten.
Anschließend soll das Script mit dem Programm espeak auf dem Linuxhost mit dem Text "Ein Feuer ist ausgebrochen" eine wave Datei erzeugen, diese soll anschließend mit dem Programm sox in eine neue brauchbare VOIP wave Datei umgewandelt werden, damit zum Schluss, dass Programm sipcmd die neue wave Datei nutzen kann um einen Anruf zu tätigen.

Herzlichen Dank schon mal

myUtils.pm

Code Auswählen

######## VoIP Call mit sipcmd starten ############
sub
voipcall
{
  my $text = shift;
  my $callnumber = shift;
  my $return_sipcmd = "";
  my $return_espeak = "";
  my $return_sox = "";
  my $name = "FhemVoIP";
  my $konto = "999999";
  my $passwrd = "*******";
  my $provider = "sip.voip2gsm.de";
  my $audiofilefolder = "/opt/fhem/voip/messages/";
  my $audiofile = int(rand(9999-1));
  my $audiofileextentin = "_mono";
  Log 1, "voipcall callnumber: $callnumber";
  Log 1, "voipcall Text: $text";
  Log 1, "voipcall Filename: $audiofile";
  Log 1, "voipcall Debug Filename/Folder/Extention: $audiofilefolder$audiofile";
  espeak -vde+m4  "$text" -b 1 -w $audiofilefolder$audiofile;
  #$return_espeak .= qx(espeak -v'de+m4'  "$text" -b '1' -w '$audiofilefolder$audiofile');
  sox $audiofilefolder$audiofile -b 16 -c 1 -r 8000 $audiofilefolder$audiofile$audiofileextentin;
  #$return_sox .= qx(sox $audiofilefolder$audiofile -b 16 -c 1 -r 8000 $audiofilefolder$audiofile$audiofileextentin);
  sipcmd -u '$konto' -c '$passwrd' -P sip -w '$provider' -x "c'$callnumber';ws1000;v$audiofilefolder$audiofile$audiofileextentin;h";
  #$return_sipcmd .= qx(sipcmd -u '$konto' -c '$passwrd' -P sip -w '$provider' -x "c'$callnumber';ws1000;v$audiofilefolder$audiofile$audiofileextentin;h");
  #$ret =~ s,[\r\n]*,,g;
  Log 1, "voipcall returned: $return_sipcmd";
}

Logdatei:

Code Auswählen

syntax error at ./FHEM/99_myUtils.pm line 35, near "m4  "$text"" 
syntax error at ./FHEM/99_myUtils.pm line 37, near "$audiofile -b "

Devender · 22 Dezember 2016, 20:57:15

Entferne mal bitte deine Userdaten zum sipaccount

Code Auswählen


my $name = "FhemVoIP";
  my $konto = "30xxxx2";
  my $passwrd = "uxxxxkx";
  my $provider = "sip.vo

Puschel74 · 22 Dezember 2016, 21:06:30

Auch die Handynummer würde ich in keinem Forum posten:

Zitat({voipcall("Ein Feuer ist ausgebrochen",'0xxxxxxxxxx')})

Das Passwort zum Account würde ich persönlich spätestens jetzt ändern

Aber ich versuch mal was zum Problem:
Du rufst eine sub mit 2 Parameter auf aber die sub erwartet keine.
1.

Code Auswählen

sub
voipcall ($$)
{

für die 2 Parameter.
2. - Zeile 35 (diese hier?)

Code Auswählen

espeak -vde+m4 "$text" -b 1 -w $audiofilefolder$audiofile;
gehört zwischen -vde und +m4 nicht ein Leerzeichen?
3. - Zeile 37 (diese hier vermutlich)

Code Auswählen

sox $audiofilefolder$audiofile -b 16 -c 1 -r 8000 $audiofilefolder$audiofile$audiofileextentin;
gehört nach sox zwischen $audiofilefolder und $audiofile nicht auch ein Leerzeichen?
Grad gesehen - gehören die Variablen nicht alle mit einem Leerzeichen getrennt

Verbunden werden sie in Perl zumindest so nicht - zum verbinden von Variablen braucht es in Perl noch den Punkt .

espeak und sox sagt mir grad nichts und ich bin im Moment auch zu faul um Google zu befragen (vielleicht mach ich das noch wenn ich Zeit habe).

Edith: Nummer vom TE "angepasst"

lammy · 22 Dezember 2016, 22:01:55

Hi Danke für die Anregunngen mit den REAL Daten

Habe schon lange nichts mehr in einem Forum gepostet....

1. ($$) habe ich hin zu gefügt

2. Die Programmzeilen habe ich aus der Shell raus kopiert wo ich sie vorher getestet habe, daher laufen sie so wie sie im Code stehen wenn man sie in eine Shell eingibt.

Ich habe den Code jetzt zum laufen gebracht bis auf "sipcmd" vieleicht habt ihr dazu noch eine Idee.

Quellcode:

Code Auswählen

##############################################
# $Id: 99_myUtils.pm 11984 2016-08-19 12:47:50Z rudolfkoenig $
package main;

use strict;
use warnings;
use POSIX;

sub
myUtils_Initialize($$)
{
  my ($hash) = @_;
}

######## VoIP Call mit sipcmd starten ############
sub
voipcall($$)
{
  my $text = shift;
  my $callnumber = shift;
  my $return_sipcmd = "";
  my $return_espeak = "";
  my $return_sox = "";
  my $name = "FhemVoIP";
  my $konto = "USER";
  my $passwrd = "pass";
  my $provider = "host";
  my $audiofilefolder = "/opt/fhem/voip/messages/";
  my $audiofile = int(rand(9999-1));
  my $dateityp = ".wav";
  my $audiofileextentin = "_mono";
  Log 1, "voipcall callnumber: $callnumber";
  Log 1, "voipcall Text: $text";
  Log 1, "voipcall Filename: $audiofile$dateityp";
  Log 1, "voipcall Debug Filename/Folder/Extention: $audiofilefolder$audiofile$dateityp";

  ####### espeak auführen #######
  #system "espeak '-vde+m4'  '$text' '-b 1' '-w $audiofilefolder$audiofile$dateityp'";
  $return_espeak .= qx(espeak '-vde+m4'  '$text' '-b 1' '-w $audiofilefolder$audiofile$dateityp');
  Log 1, "voipcall espeak returned: $return_espeak";

  ####### sox auführen #######
  #system "sox '$audiofilefolder$audiofile$dateityp' '-b 16' '-c 1' '-r 8000' '$audiofilefolder$audiofile$audiofileextentin$dateityp'";
  $return_sox .= qx(sox '$audiofilefolder$audiofile$dateityp' '-b 16' '-c 1' '-r 8000' '$audiofilefolder$audiofile$audiofileextentin$dateityp');
  Log 1, "voipcall sox returned: $return_sox";
  
  ####### sipcm auführen #######
[color=red][u][b]  $return_sipcmd .= qx(sipcmd '-u $konto' '-c $passwrd' '-P sip' '-w $provider' '-x c$callnumber;ws1000;$audiofilefolder$audiofile$audiofileextentin$dateityp;h');[/b][/u][/color]
  #$ret =~ s,[\r\n]*,,g;
  Log 1, "voipcall sipcmd returned: $return_sipcmd";
  #system "rm '$audiofilefolder$audiofile$dateityp'";
  #system "rm '$audiofilefolder$audiofile$dateityp$audiofileextentin'";
  Log 1, "voipcall: nicht benoetigte Dateien wurden geloescht";

}

1;

Fehlercode:

Code Auswählen

2016.12.22 21:58:17 1: voipcall sipcmd returned: Starting sipcmd
invalid protocol

Devender · 22 Dezember 2016, 22:31:38

Zitat von: lammy am 22 Dezember 2016, 22:01:55
[/code]

Fehlercode:
Code Auswählen Erweitern
2016.12.22 21:58:17 1: voipcall sipcmd returned: Starting sipcmd invalid protocol

Anhand der Meldung würde er jetzt versuchen dein Usernamen als protocol zu identifizieren.
ich würde das -P sip mal ganz an den Anfang setzen.

So steht es zumindest in den Beispielen im Netz:

Code Auswählen


./sipcmd -P sip -u [username] -c [password] -w [server] -x "c;w200;d12345"

Ist aber nur geraten bzw. anhand der Meldung abgeleitet.

lammy · 23 Dezember 2016, 00:50:02

Hi,

danke für den Tip. Habe es geschafft.
Ich würde ganz gerne nur noch die Rückmeldung bekommen das auch ein Anruf rausgegangen ist. Sprich in die Log geschrieben haben.
Anbei ist ein Auszug von dem sipcmd wie es aktuell in die Config reinballert.
Man müsste nach "## Hangup ##" Wenn das vorkommt dann war der call i.O. Jemand ne Idee??

Code Auswählen

016.12.23 00:35:50 1: voipcall sipcmd returned: Starting sipcmdin debug modeManagerInitinitialising SIP endpoint...TestChanAudioTestChanAudioListening for SIP signalling on 0.0.0.0:5060SIP listener upregistered as sip:3000.02@sip.voip2gsm.deCreated LocalEndPointMain## Call ##TestPhone::Main: calling "01718637118" using gateway "sip.voip2gsm.de" at Fri Dec 23 00:35:18 2016Setting up a call to: sip:01718637118@sip.voip2gsm.deLocalEndpoint::MakeConnectionLocalEndpointCreateConnectionLocalConnectionOnIncomingConnection: token=Lcdf475db2connection set up to sip:01718637118@sip.voip2gsm.deTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 0TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 1TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 1TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 1TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 1TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 1TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 1TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 1TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 1TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 1TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 1TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 2TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 2TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 2TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 2TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 2TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 2TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 2TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 2TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 2TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 2TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 3TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 3TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 3TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 3TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 3TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 3TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 3TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 3TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 3TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 3TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 4TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 4TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 4TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 4TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 4TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 4TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 4TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 4TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 4TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 5TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 5TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 5TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 5TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 5TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 5TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 5TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 5TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 5TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 5TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 6TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 6TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 6TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 6TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 6TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 6TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 6TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 6TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 6TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 6TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 7TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 7TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 7TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 7TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 7TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 7TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 7TestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 7OnOpenMediaStreamrecording media from sipCreateMediaStreamTestChannel[ Call[C597bd9c51]-EP<local>[Lcdf475db2] - 0x7f9014031cb0 ]OnOpenMediaStreamstreaming media to localTestChannel::WriteTestChannel::WriteCreateMediaStreamTestChannel[ Call[C597bd9c51]-EP<local>[Lcdf475db2] - 0x7f901404b130 ]OnOpenMediaStreamrecording media from localOnOpenMediaStreamstreaming media to sipTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 7TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 7TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 8TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 8TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 8TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 8TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 8TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 8TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 8TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 8TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 8TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 8TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 9TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 9TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 9TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 9TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 9TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 9TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 9TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 9TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 9TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 9TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 10TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 10TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 10TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 10TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 10TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 10TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 10TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 10TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 10TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 10TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 11TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestPhone::Main: calling "sip:01718637118@sip.voip2gsm.de" for 11TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteOnEstablished## Wait: waiting for 1000ms ##OnEstablishedOnEstablishedCallIn call with sip:01718637118@85.25.203.230:5060 using local:fhem token=[C597bd9c51]TestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteWait: wait done## Voice audiofile=/opt/fhem/voip/messages/6448_mono.wav ##PlaybackAudioFilePlaybackAudioFile: opening file "/opt/fhem/voip/messages/6448_mono.wav" as WAVPlaybackAudioTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteTestChannel::WriteStopAudioPlaybackTestChanAudio::PlaybackAudio: play back done 1## Hangup ##Hangup: at Fri Dec 23 00:35:34 2016TestChannel::WriteTestChannel::WriteOnClosedMediaStreamClose [ Call[C597bd9c51]-EP<local>[Lcdf475db2] - 0x7f9014031cb0 ]TestChanAudio::CloseChannelStopAudioPlaybackStopAudioRecordingOnClosedMediaStreamOnClosedMediaStreamClose [ Call[C597bd9c51]-EP<local>[Lcdf475db2] - 0x7f901404b130 ]TestChanAudio::CloseChannelStopAudioPlaybackStopAudioRecordingOnClosedMediaStreamOnReleased: reason: EndedByLocalUserOnReleased: reason: EndedByLocalUserOnClearedCall~LocalConnectionTestPhone::Main: shutting downTestPhone::Main: exiting...Exiting...~Manager

Hier der funktionierende Quellcode

Code Auswählen

##############################################
# $Id: 99_myUtils.pm 11984 2016-08-19 12:47:50Z rudolfkoenig $
package main;

use strict;
use warnings;
use POSIX;

sub
myUtils_Initialize($$)
{
  my ($hash) = @_;
}

######## VoIP Call mit sipcmd starten ############
######## Beispiel:
######## ({voipcall("Guten Tag hier ist Hans Wurst",'08003301000')})
sub
voipcall($$)
{
  my $text = shift;
  my $callnumber = shift;
  my $return_sipcmd = "";
  my $return_espeak = "";
  my $return_sox = "";
  my $ret = "";
  my $name = "FhemVoIP";
  my $konto = "user";
  my $passwrd = "pass";
  my $provider = "host";
  my $protocol = "sip";
  my $audiofilefolder = "/opt/fhem/voip/messages/";
  my $audiofile = int(rand(9999-1));
  my $dateityp = ".wav";
  my $audiofileextentin = "_mono";
  Log 1, "-------- voipcall start --------";
  Log 1, "voipcall callnumber: $callnumber";
  Log 1, "voipcall Text: $text";
  Log 1, "voipcall Filename: $audiofile$dateityp";
  Log 1, "voipcall Debug Filename/Folder/Extention: $audiofilefolder$audiofile$dateityp";

  ####### espeak auführen #######
  $return_espeak .= qx(espeak '-vde+m4'  '$text' '-b 1' '-w $audiofilefolder$audiofile$dateityp');
  Log 1, "voipcall espeak returned: $return_espeak";
  ####### sox auführen #######
  $return_sox .= qx(sox '$audiofilefolder$audiofile$dateityp' '-b 16' '-c 1' '-r 8000' '$audiofilefolder$audiofile$audiofileextentin$dateityp');
  Log 1, "voipcall sox returned: $return_sox";


  ####### sipcm auführen #######
  #system "echo sipcmd -P $protocol -u $konto -c '$passwrd' -w $provider -x 'c$callnumber;ws10000;$audiofilefolder$audiofile$audiofileextentin$dateityp;h'";
  $return_sipcmd .= qx(sipcmd -P $protocol -u $konto -c '$passwrd' -w $provider -x 'c$callnumber;ws1000;v$audiofilefolder$audiofile$audiofileextentin$dateityp;h' &);
  #system "sipcmd -P $protocol -u $konto -c '$passwrd' -w $provider -x 'c$callnumber;ws1000;v$audiofilefolder$audiofile$audiofileextentin$dateityp;h' &";
  $return_sipcmd =~ s,[\r\n]*,,g;    # remove CR from return-string
  Log 1, "voipcall sipcmd returned: $return_sipcmd";
  #system "rm '$audiofilefolder$audiofile$dateityp'";
  #system "rm '$audiofilefolder$audiofile$dateityp$audiofileextentin'";
  Log 1, "voipcall cleaning: nicht benoetigte Dateien wurden geloescht";
  Log 1, "-------- voipcall end --------";
}

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