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Code Issues Releases Wiki Activity

96_SIP: add direct support for Text2Speech (Forum #604112)

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git-svn-id: https://svn.fhem.de/fhem/trunk@13693 2b470e98-0d58-463d-a4d8-8e2adae1ed80
This commit is contained in:
Wzut 2017-03-13 19:14:59 +00:00
parent ed378ad815
commit 483505600e
1 changed files with 453 additions and 85 deletions
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538
fhem/FHEM/96_SIP.pm
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@ -46,14 +46,15 @@ use strict;
use warnings;
use Net::SIP qw//;
use Net::SIP::Packet;
use IO::Socket;
use Socket;
use Net::Domain qw( hostfqdn );
use Blocking; # http://www.fhemwiki.de/wiki/Blocking_Call
#use Data::Dumper;
my $sip_version ="V1.31 / 28.02.17";
my $ua; # SIP user agent
my $sip_version ="V1.41 / 13.03.17";
my $ua; # SIP user agent
my %sets = (
"call" => "",
@ -73,7 +74,9 @@ sub SIP_Initialize($$)
$hash->{ShutdownFn} = "SIP_Undef";
$hash->{SetFn} = "SIP_Set";
$hash->{GetFn} = "SIP_Get";
$hash->{AttrList} = "sip_waits ".
$hash->{NotifyFn} = "SIP_Notify";
$hash->{AttrFn} = "SIP_Attr";
$hash->{AttrList} = "sip_watch_listen ".
"sip_ringtime ".
"sip_waittime ".
"sip_ip ".
@ -81,9 +84,17 @@ sub SIP_Initialize($$)
"sip_user ".
"sip_registrar ".
"sip_from ".
"sip_audiofile ".
"sip_audiofile_call ".
"sip_audiofile_dtmf ".
"sip_audiofile_ok ".
"sip_audiofile_wfp ".
"sip_dtmf_size:1,2,3,4 ".
"sip_listen:none,dtmf,wfp ".
"sip_dtmf_send:audio,rfc2833 ".
"sip_dtmf_loop:once,loop ".
"sip_listen:none,dtmf,wfp ".
"T2S_Device ".
"T2S_Timeout ".
"audio_converter:sox,ffmpeg ".
"disabled:0,1 ".$readingFnAttributes;
}
@ -97,13 +108,17 @@ sub SIP_Define($$)
$hash->{STATE} = "defined";
$hash->{VERSION} = $sip_version;
$attr{$name}{sip_ringtime} = '10' unless (exists($attr{$name}{sip_ringtime}));
$hash->{".reset"} = 0;
$attr{$name}{sip_ringtime} = '3' unless (exists($attr{$name}{sip_ringtime}));
$attr{$name}{sip_user} = '620' unless (exists($attr{$name}{sip_user}));
$attr{$name}{sip_ip} = $addr unless (exists($attr{$name}{sip_ip}));
$attr{$name}{sip_port} = '5060' unless (exists($attr{$name}{sip_port}));
$attr{$name}{sip_registrar} = 'fritz.box' unless (exists($attr{$name}{sip_registrar}));
$attr{$name}{sip_listen} = 'none' unless (exists($attr{$name}{sip_listen}));
$attr{$name}{sip_dtmf_size} = '2' unless (exists($attr{$name}{sip_dtmf_size}));
$attr{$name}{sip_dtmf_loop} = 'once' unless (exists($attr{$name}{sip_dtmf_loop}));
$attr{$name}{sip_dtmf_send} = 'audio' unless (exists($attr{$name}{sip_dtmf_send}));
$attr{$name}{sip_from} = 'sip:'.$attr{$name}{sip_user}.'@'.$attr{$name}{sip_registrar} unless (exists($attr{$name}{sip_from}));
RemoveInternalTimer($hash);
@ -111,6 +126,71 @@ sub SIP_Define($$)
return undef;
}
sub SIP_Notify($$)
{
# $hash is my entry, $dev_hash is the entry of the changed device
my ($hash, $dev_hash) = @_;
return undef if ($dev_hash->{NAME} ne AttrVal($hash->{NAME},"T2S_Device",""));
SIP_wait_for_tts($hash) if (defined($hash->{callnr}) && defined($hash->{ringtime}));
return undef;
}
sub SIP_Attr (@)
{
my ($cmd, $name, $attrName, $attrVal) = @_;
my $hash = $defs{$name};
if ($cmd eq "set")
{
if (substr($attrName ,0.4) eq "sip_")
{
$_[3] = $attrVal;
$hash->{".reset"} = 1 if ($hash->{LPID} && ($attrName ne "sip_audiofile_call") && ($attrName ne "sip_dtmf_send"));
}
elsif (($attrName eq "disable") && ($attrVal == 1))
{
readingsSingleUpdate($hash,"state","disabled",1);
$_[3] = $attrVal;
$hash->{".reset"} = 1 if $hash->{LPID};
}
elsif ($attrName eq "audio_converter")
{
my $res = qx(which $attrVal);
$res =~ s/\n//;
$hash->{AC} = ($res) ? $res : undef;
}
elsif ($attrName eq "T2S_Device")
{
$_[3] = $attrVal;
$hash->{NOTIFYDEV} = $attrVal;
}
}
elsif ($cmd eq "del")
{
if (substr($attrName,0.4) eq "sip_")
{
$_[3] = $attrVal;
$hash->{".reset"} = 1 if ($hash->{LPID} && ($attrName ne "sip_audiofile_call") && ($attrName ne "sip_dtmf_send"));
}
elsif ($attrName eq "audio_converter")
{
$_[3] = $attrVal;
delete $hash->{AC};
}
elsif ($attrName eq "T2S_Device")
{
$_[3] = $attrVal;
delete $hash->{NOTIFYDEV};
}
}
SIP_updateConfig($hash) if ($hash->{".reset"});
return undef;
}
sub SIP_updateConfig($)
{
# this routine is called 5 sec after the last define of a restart
@ -133,7 +213,7 @@ sub SIP_updateConfig($)
RemoveInternalTimer($hash);
if(defined($hash->{LPID}))
{
Log3 $name,4, "$name, Listen Kill PID : ".$hash->{LPID};
Log3 $name,4, "$name, Listen Kill PID : ".$hash->{LPID};
BlockingKill($hash->{helper}{LISTEN_PID});
delete $hash->{helper}{LISTEN_PID};
delete $hash->{LPID};
@ -155,6 +235,18 @@ sub SIP_updateConfig($)
return undef;
}
my $t2s = AttrVal($name,"T2S_Device",undef);
$hash->{NOTIFYDEV} = $t2s if defined($t2s);
if (AttrVal($name,"audio_converter","") && defined($t2s))
{
my $converter = AttrVal($name,"audio_converter","");
my $res = qx(which $converter);
$res =~ s/\n//;
$hash->{AC} = ($res) ? $res : undef;
}
if (AttrVal($name,"sip_listen", "none") ne "none")
{
@ -174,9 +266,9 @@ sub SIP_updateConfig($)
}
sub SIP_Register($$)
sub SIP_Register($$$)
{
my ($hash,$port) = @_;
my ($hash,$port,$type) = @_;
my $name = $hash->{NAME};
my $ip = AttrVal($name,"sip_ip","");
return "missing attr sip_ip" if (!$ip);
@ -218,6 +310,7 @@ sub SIP_Register($$)
$sub_register = sub {
my $expire = $ua->register(registrar => $registrar ) || return "registration failed: ".$ua->error;
Log3 $name,4,"$name, register new expire : ".localtime(time()+$expire);
SIP_telnet($hash,"set $name state $type\nexit\n");
# need to refresh registration periodically
$ua->add_timer( $expire/2, $sub_register );
};
@ -250,41 +343,51 @@ sub SIP_CALLStart($)
my $dtmf = 'ABCD*#123--4567890';
my $port = AttrVal($name,"sip_port","5060");
my $call;
my $codec;
$ua = undef;
sleep 1;
my $error = SIP_Register($hash,$port);
$hash->{telnetPort} = SIP_telnetPort();
return $name."|no telnet port without password found" if (!$hash->{telnetPort});
my $error = SIP_Register($hash,$port,"calling");
return $name."|0|CallRegister: $error" if ($error);
if ((substr($msg,0,1) ne "-") && $msg)
{
Log3 $name,4,"$name, CallStart msg : $msg";
$call = $ua->invite( $nr,
init_media => $ua->rtp('send_recv', $msg),
$codec = "PCMA/8000" if ($msg =~ /\.al(.+)$/);
$codec = "PCMU/8000" if ($msg =~ /\.ul(.+)$/);
return $name."|0|CallStart: please use filetype .alaw (for a-law) or .ulaw (for u-law)" if !defined($codec);
Log3 $name,4,"$name, CallStart msg : $msg - $codec";
$call = $ua->invite( $nr,
init_media => $ua->rtp('send_recv', $msg),
cb_rtp_done => \$rtp_done,
cb_final => sub { my ($status,$self,%info) = @_; $final = $info{code};},
recv_bye => \$peer_hangup,
cb_final => sub { my ($status,$self,%info) = @_; $final = $info{code};},
recv_bye => \$peer_hangup,
cb_noanswer => \$no_answer,
asymetric_rtp => 0,
rtp_param => [8, 160, 160/8000, 'PCMA/8000']) || return $name."|0|invite failed: ".$ua->error;
rtp_param => [8, 160, 160/8000, $codec]) || return $name."|0|invite failed: ".$ua->error;
}
else
{
$dtmf = (substr($msg,0,1) eq "-") ? substr($msg,1) : $dtmf;
Log3 $name,4,"$name, CallStart DTMF : $dtmf";
$call = $ua->invite($nr,
init_media => $ua->rtp( 'recv_echo',undef,0 ),
cb_final => sub { my ($status,$self,%info) = @_; $final = $info{code};},
init_media => $ua->rtp( 'recv_echo',undef,0 ),
rtp_param => [0, 160, 160/8000, 'PCMU/8000'],
cb_final => sub { my ($status,$self,%info) = @_; $final = $info{code};},
cb_noanswer => \$no_answer,
recv_bye => \$peer_hangup) || return $name."|0|invite failed ".$ua->error;
$call->dtmf( $dtmf, cb_final => \$rtp_done);
recv_bye => \$peer_hangup) || return $name."|0|invite failed ".$ua->error;
if (AttrVal($name,"sip_dtmf_send","audio") eq "audio")
{ $call->dtmf( $dtmf, methods => 'audio', duration => 500, cb_final => \$rtp_done); }
else { $call->dtmf( $dtmf, cb_final => \$rtp_done); }
}
return "$name|0|invite call failed ".$call->error if ($call->error);
$hash->{telnetPort} = SIP_telnetPort();
SIP_telnet($hash,"set $name call_state calling $nr\nexit\n") if $hash->{telnetPort} ;
SIP_telnet($hash,"set $name call_state calling $nr\nexit\n");
Log3 $name,4,"$name, calling : $nr";
return "$name|0|no answer" if ($no_answer);
@ -300,7 +403,6 @@ sub SIP_CALLStart($)
$ua->loop( \$stopvar );
}
# $state = $rtp_done if (!$state);
Log3 $name,5,"$name, RTP done : $rtp_done" if defined($rtp_done);
Log3 $name,5,"$name, Hangup : $peer_hangup" if defined($peer_hangup);
Log3 $name,5,"$name, Stopvar : $stopvar" if defined($stopvar);
@ -308,7 +410,6 @@ sub SIP_CALLStart($)
if (defined($rtp_done))
{
#print Dumper($stopvar);
if ($rtp_done eq "OK") {return $name."|1|ok";} # kein Audio
else
{
@ -353,6 +454,7 @@ sub SIP_CALLDone($)
readingsBulkUpdate($hash, "call","done");
readingsBulkUpdate($hash, "last_error",$final);
readingsBulkUpdate($hash, "call_state","fail");
readingsBulkUpdate($hash, "state",$hash->{'.oldstate'}) if defined($hash->{'.oldstate'});
readingsEndUpdate($hash, 1);
}
else
@ -360,6 +462,7 @@ sub SIP_CALLDone($)
readingsBeginUpdate($hash);
readingsBulkUpdate($hash, "call","done");
readingsBulkUpdate($hash, "call_state",lc($final));
readingsBulkUpdate($hash, "state",$hash->{'.oldstate'}) if defined($hash->{'.oldstate'});
readingsEndUpdate($hash, 1);
}
@ -393,26 +496,72 @@ sub SIP_Set($@)
if ($cmd eq "call")
{
my $nr = (defined($a[2])) ? $a[2] : "";
my $ringtime = (defined($a[3])) ? $a[3] : AttrVal($name, "sip_ringtime", "10");
my $msg = (defined($a[4])) ? $a[4] : AttrVal($name, "sip_audiofile", "");
my $ringtime = (defined($a[3])) ? $a[3] : 30;
my $msg = (defined($a[4])) ? $a[4] : AttrVal($name, "sip_audiofile_call", "");
return "there is already a call activ with pid ".$hash->{CPID} if exists($hash->{CPID});
return "missung call number" if (!$nr);
return "missing call number" if (!$nr);
my $arg;
if ($msg)
{
Log3 $name, 4, $name.", sending file $msg to $nr, ringtime: $ringtime";
#Log3 $name, 4, $name.", sending $msg to $nr, ringtime: $ringtime";
if((-e $msg) || (substr($msg,0,1) eq "-")) { Log3 $name, 4, $name.", $msg found"; }
else
if (substr($msg,0,1) eq "-")
{
Log3 $name, 4, $name.", message DTMF = $msg";
}
elsif (substr($msg,0,1) eq "!") # Text2Speech Text ?
{
$msg =~ s/^\!//; # das ! muss weg
my $t2s_name = AttrVal($name,"T2S_Device",undef);
return "attr T2S_Device not set !" if !defined($t2s_name);
my $t2s_hash = $defs{$t2s_name};
return "T2S_Device $t2s_name not found" if !defined($t2s_hash);
return "attr audio_converter not set" if !AttrVal($name,"audio_converter","");
return "external sox or ffmpeg programm not found, please install sox or ffmpeg first and set attr audio_converter" if !defined($hash->{AC});
my $t2s_file = ReadingsVal($t2s_name,"lastFilename",undef);
Log3 $name,3,"$name, Reading lastFilename not found at device $t2s_name, are you using a old version ?" if !defined($t2s_file);
readingsSingleUpdate($t2s_hash,"lastFilename","---",0);
shift @a;
shift @a;
$a[0] = $t2s_name; # ist aber egal wird eh verworfen
$a[1] = "tts"; # Kommando des Set Befehls
$a[2] = $msg;
my $ret = Text2Speech_Set($t2s_hash, @a); # na dann lege schon mal los
if (defined($ret))
{
Log3 $name,3,"$name, T2S error : $ret";
readingsSingleUpdate($hash,"last_error",$ret,0);
return $ret; # Das ging leider schief
}
readingsSingleUpdate($hash,"call_state","waiting T2S",0);
$hash->{callnr} = $nr;
$hash->{ringtime} = $ringtime;
RemoveInternalTimer($hash);
# geben wir TTS mal ein paar Sekunden
InternalTimer(gettimeofday()+int(AttrVal($name,"T2S_Timeout",5)), "SIP_wait_for_tts", $hash, 0);
return undef;
}
elsif (-e $msg)
{
Log3 $name, 4, $name.", message $msg found";
return "unknown message type, please use only .alaw or .ulaw" if (($msg !~ /\.al(.+)$/) && ($msg !~ /\.ul(.+)$/));
}
else
{
Log3 $name, 3, $name.", message $msg NOT found !";
$msg = "";
}
}
else { Log3 $name, 4, $name.", calling $nr, ringtime: $ringtime"; }
else { Log3 $name, 4, $name.", calling $nr, ringtime: $ringtime , no message"; }
$arg = "$name|$nr|$ringtime|$msg";
my $arg = "$name|$nr|$ringtime|$msg";
#BlockingCall($blockingFn, $arg, $finishFn, $timeout, $abortFn, $abortArg);
$hash->{helper}{CALL_PID} = BlockingCall("SIP_CALLStart",$arg, "SIP_CALLDone") unless(exists($hash->{helper}{CALL_PID}));
@ -424,6 +573,7 @@ sub SIP_Set($@)
readingsBulkUpdate($hash, "call_state","invite");
readingsBulkUpdate($hash, "call",$nr);
readingsEndUpdate($hash, 1);
$hash->{'.oldstate'} = ReadingsVal($name,"state",undef);
return undef;
}
else
@ -469,7 +619,7 @@ sub SIP_Set($@)
}
elsif ($cmd eq "reset")
{
$hash->{".reset"} =1;
$hash->{".reset"} = 1;
SIP_updateConfig($hash);
return undef;
}
@ -480,7 +630,12 @@ sub SIP_Set($@)
# den Rest als ein String
$subcmd = join(" ",@a);
if ($cmd eq "caller")
if ($cmd eq "state")
{
readingsSingleUpdate($hash, "state",$subcmd,1);
return undef;
}
elsif ($cmd eq "caller")
{
readingsSingleUpdate($hash, "caller",$subcmd,1);
return undef;
@ -533,10 +688,18 @@ sub SIP_ListenStart($)
my $hash = $defs{$name}; # $hash / $name gueltig in diesem Block
$hash->{telnetPort} = undef;
my $dtmfloop; # Ende-Flag für die DTMF-Schleife
my $okloop; # Ende-Flag für die OK-Ansage
my $okloopbye = 0; # Ende-Flag für recv_bye währne der OK-Ansage
my $byebye = 0; # Anrufer hat aufgelegt
my $packets = 50;
my $sub_create;
my $sub_invite;
my $sub_filter;
my $sub_bye;
my $sub_dtmf;
my $send_something;
$hash->{telnetPort} = SIP_telnetPort();
@ -545,24 +708,44 @@ sub SIP_ListenStart($)
my $port = AttrVal($name,"sip_port","5060");
$port += 10;
$ua = undef;
my $error = SIP_Register($hash,$port);
my $error = SIP_Register($hash,$port,"listen_".AttrVal($name,"sip_listen",""));
return $name."|ListenRegister: $error" if ($error);
my $msg1 = AttrVal($name, "sip_audiofile_dtmf", "");
my $msg2 = AttrVal($name, "sip_audiofile_ok", "");
my $msg3 = AttrVal($name, "sip_audiofile_wfp", "");
$msg1 = SIP_check_file($hash,$msg1) if ($msg1);
$msg2 = SIP_check_file($hash,$msg2) if ($msg2);
$msg3 = SIP_check_file($hash,$msg3) if ($msg3);
my $msg = AttrVal($name, "sip_audiofile", "");
$hash->{dtmf} = 0;
$hash->{dtmf_event} = "";
$hash->{old} ="-";
$sub_dtmf = sub {
$send_something = sub {
return unless $packets-- > 0;
my $buf = sprintf "%010d",$packets;
$buf .= "1234567890" x 15;
return $buf; # 160 bytes for PCMU/8000
};
$sub_dtmf = sub
{
my ($event,$dur) = @_;
Log3 $name,5,"$name : DTMF Event : $event";
if ($event eq "#")
Log3 $name,5,"$name : DTMF Event : $event - $dur ms";
return if (int($dur) < 100);
if (($event eq "#") && !$hash->{dtmf})
{
$hash->{dtmf} = 1;
$hash->{old} = $event;
return;
$hash->{dtmf} = 1;
$hash->{dtmf_event} = "";
$hash->{old} = $event;
return;
}
if (($event ne $hash->{old}) && $hash->{dtmf})
{
@ -577,14 +760,25 @@ sub SIP_ListenStart($)
$hash->{dtmf} = 0;
$hash->{dtmf_event} = "";
$hash->{old} ="-";
$dtmfloop = 1;
}
}
return;
return;
};
$sub_invite = sub {
$sub_create = sub
{
my ($call,$request,$leg,$from) = @_;
my $method = $request->method;
my $response = $request->create_response( '180','Ringing' );
$call->{endpoint}->new_response( $call->{ctx},$response,$leg,$from );
1;
};
$sub_invite = sub
{
my ($a,$b,$c,$d) = @_;
my $waittime = AttrVal($name, "sip_waittime", "10");
my $waittime = int(AttrVal($name, "sip_waittime", 10));
my $action;
my $i;
@ -600,12 +794,12 @@ sub SIP_ListenStart($)
SIP_telnet($hash,"set $name caller_state fetching\nexit\n");
last;
}
#$call->bye();
}
return 0;
};
$sub_filter = sub {
$sub_filter = sub
{
my ($a,$b) = @_;
my ($caller,undef) = split("\;", $a);
$caller =~ s/\"//g;
@ -618,11 +812,13 @@ sub SIP_ListenStart($)
return 1;
};
$sub_bye = sub {
$sub_bye = sub
{
my ($event) = @_;
Log3 $name, 5, "$name, SIP_bye : $event";
#print Dumper($event);
SIP_telnet($hash, "set $name caller none\nset $name caller_state hangup\nexit\n") ;
$byebye = 1;
return 1;
};
@ -631,28 +827,97 @@ sub SIP_ListenStart($)
if (AttrVal($name,"sip_listen", "none") eq "dtmf")
{
$hash->{dtmf} = 0;
$ua->listen(init_media => ($msg) ? $ua->rtp('send_recv',$msg) : $ua->rtp('recv_echo'), # echo everything back
cb_dtmf => \&$sub_dtmf ,
$dtmfloop = 0; # Ende-Flag für die DTMF-Schleife
$okloop = 0; # Ende-Flag für die OK-Ansage
$okloopbye = 0; # Ende-Flag für recv_bye währne der OK-Ansage
$byebye = 0; # Anrufer hat aufgelegt
while(1)
{
my $call;
$ua->listen (cb_create => \&$sub_create,
cb_invite => sub {
SIP_telnet($hash,"set $name caller_state ringing\nexit\n");
sleep int(AttrVal($name, "sip_ringtime", 3)); #Anrufer hört das typische Klingeln wenn die Gegenseite nicht abnimmt
},
filter => \&$sub_filter,
recv_bye => \&$sub_bye,
asymetric_rtp => 0,
rtp_param => [8, 160, 160/8000, 'PCMA/8000'])
}
cb_established => sub {
(my $status,$call) = @_;
SIP_telnet($hash,"set $name caller_state established\nexit\n");
return 1;
} # sobald invite verlassen wird, wird in cb_established verzweigt
);
$ua->loop(\$call);
# Der SIP-Client ist jetzt im echo-Modus und zwar so lange, bis der Anrufer auflegt,
# das bekommen wir durch recv_bye mit
my $dtmf_loop = 1; # für jeden Anruf neu setzen
while ($dtmf_loop) # Schleife für Code-Ansage, DTMF-Erkennung, okay-Ansage
{
$dtmfloop = 0;
$okloop = 0;
$okloopbye = 0;
$call->reinvite(
init_media => $ua->rtp('send_recv',($msg1) ? $msg1 : $send_something),
rtp_param => [8, 160, 160/8000, 'PCMA/8000'],
cb_rtp_done => sub { $packets = 25; },
cb_dtmf => \&$sub_dtmf,
recv_bye => \&$sub_bye,
);
$ua->loop(\$dtmfloop, \$byebye);
if (!$byebye)
{ # Anrufer hat nicht aufgelegt
$call->reinvite(
init_media => $ua->rtp('send_recv',($msg2) ? $msg2 : $send_something),
rtp_param => [8, 160, 160/8000, 'PCMA/8000'],
cb_rtp_done => sub { select(undef, undef, undef, 0.1); $okloop = 1; $packets = 50;},
recv_bye => sub { $okloopbye = 1; },
cb_cleanup => sub {0},
);
$ua->loop(\$okloop,\$okloopbye); # ohne diese loop endet der Anruf sofort
}
else { $dtmf_loop = 0; } # Schleife beenden, Anrufer hat aufgelegt
if ( ( defined $okloopbye && $okloopbye ) || $byebye )
{ # wenn jemand mitten im "okay" auflegt
$dtmf_loop = 0; # beende die inner loop
$byebye = 1;
}
else { $dtmf_loop = (AttrVal($name,"sip_dtmf_loop","once")) ? 0 : 1;
SIP_telnet($hash, "set $name caller none\nset $name caller_state hangup\nexit\n") if(!$dtmf_loop);
} # führt ggf. zum Schleifenende
} # end inner loop
if (!$byebye)
{ # Anrufer hat nicht aufgelegt und nur ein DTMF angefordert
my $hanguploop;
$call->bye( cb_final => \$hanguploop );
$ua->loop( \$hanguploop );
}
} # while(1)
}
elsif (AttrVal($name,"sip_listen", "none") eq "wfp")
{
$ua->listen(
cb_invite => \&$sub_invite,
filter => \&$sub_filter,
recv_bye => \&$sub_bye,
init_media => ($msg) ? $ua->rtp('send_recv',$msg) : $ua->rtp('recv_echo'),
asymetric_rtp => 0,
rtp_param => [8, 160, 160/8000, 'PCMA/8000']
cb_create => \&$sub_create,
cb_invite => \&$sub_invite,
filter => \&$sub_filter,
recv_bye => \&$sub_bye,
init_media => $ua->rtp('send_recv',($msg3) ? $msg3 : $send_something),
cb_rtp_done => sub { $packets = 50;},
rtp_param => [8, 160, 160/8000, 'PCMA/8000']
); # options are invite and hangup
}
else { return $name."|end"; }
$ua->loop;
$ua->cleanup;
return $name."|end";
}
else { return $name."|end"; }
$ua->loop;
return $name."|end"; # hier sollten wir eigentlich nie himkommen !
}
@ -680,7 +945,7 @@ sub SIP_ListenDone($)
readingsEndUpdate($hash, 1 );
Log3 $name, 3 , "$name, listen error -> $ret";
return if(IsDisabled($name));
InternalTimer(gettimeofday()+AttrVal($name, "sip_waits", 60), "SIP_try_listen", $hash, 0);
InternalTimer(gettimeofday()+AttrVal($name, "sip_watch_listen", 60), "SIP_try_listen", $hash, 0);
}
else
{
@ -696,7 +961,7 @@ sub SIP_try_listen($)
{
my ($hash) = @_;
my $name = $hash->{NAME};
my $waits = AttrVal($name, "sip_waits", 60);
my $waits = AttrVal($name, "sip_watch_listen", 60);
$hash->{helper}{LISTEN_PID} = BlockingCall("SIP_ListenStart",$name, "SIP_ListenDone") unless(exists($hash->{helper}{LISTEN_PID}));
@ -706,8 +971,8 @@ sub SIP_try_listen($)
Log3 $name, 4 , $name.", Listen new PID : ".$hash->{LPID};
RemoveInternalTimer($hash);
InternalTimer(gettimeofday()+$waits, "SIP_watch_listen", $hash, 0); # starte die Überwachung
my $state = "listen_for_".AttrVal($name,"sip_listen",undef);
readingsSingleUpdate($hash, "state", $state, 1);
#my $state = "listen_for_".AttrVal($name,"sip_listen",undef);
#readingsSingleUpdate($hash, "state", $state, 1);
return 0;
}
else
@ -729,7 +994,6 @@ sub SIP_watch_listen($)
return if (IsDisabled($name));
return if (!defined($hash->{LPID}));
my $waits = AttrVal($name, "sip_waits", 60);
my $cmd = "ps -e | grep '".$hash->{LPID}." '";
my $result = qx($cmd);
@ -743,10 +1007,76 @@ sub SIP_watch_listen($)
}
else { Log3 $name, 5 , $name.", listen prozess ".$hash->{LPID}." found"; }
InternalTimer(gettimeofday()+$waits, "SIP_watch_listen", $hash, 0);
InternalTimer(gettimeofday()+60, "SIP_watch_listen", $hash, 0);
return;
}
sub SIP_wait_for_tts($)
{
my ($hash) = @_;
my $name = $hash->{NAME};
RemoveInternalTimer($hash);
my $t2s_name = AttrVal($name,"T2S_Device",undef);
my $file = ReadingsVal($t2s_name,"lastFilename","---");
my $msg = "";
if ($file ne "---")
{
Log3 $name,4,"$name, new TTS file $file";
my $out = $file;
$out =~ s/mp3/alaw/;
if (-e $out)
{
Log3 $name,5,"$name, not converted using $out from cache";
$msg = $out;
}
else
{
my $ret;
my $cmd;
my $converter = AttrVal($name,"audio_converter","sox");
if (($converter eq "sox") && defined($hash->{AC}))
{
$cmd = $hash->{AC}." ".$file." -t raw -r 8000 -c 1 -e a-law ".$out;
Log3 $name,5,"$name, $cmd";
$_ = qx($cmd);
}
elsif (($converter eq "ffmpeg") && defined($hash->{AC}))
{
$cmd = $hash->{AC}." -v quiet -y -i ".$file." -f alaw -ar 8000 ".$out;
Log3 $name,5,"$name, $cmd";
$_ = qx($cmd);
}
if ($_)
{
Log3 $name,4,"$name, $converter : $_ , $?";
readingsSingleUpdate($hash,"last_error","$converter error $_",1);
}
else { $msg = $out; }
}
}
else
{
Log3 $name,3,"$name, timeout waiting for T2S";
readingsSingleUpdate($hash,"call_state","TTS timeout",1);
}
# nun aber calling
my @a = ($name,"call",$hash->{callnr}, $hash->{ringtime},$msg) ;
delete($hash->{callnr});
delete($hash->{ringtime});
my $ret = SIP_Set($hash , @a);
Log3 $name,3,"$name, TTS Call : $ret" if defined($ret);
# haben wir vllt. den Timer missbraucht ?
SIP_watch_listen($hash) if (defined($hash->{LPID}));
return undef;
}
sub SIP_telnetPort()
{
foreach my $d (sort keys %defs)
@ -863,6 +1193,30 @@ sub SIP_readPassword($)
#####################################
sub SIP_check_file($$)
{
my ($hash,$file) = @_;
my $name = $hash->{NAME};
if (substr($file,0,1) eq "!")
{
Log3 $name,3,"$name, Text2Speech is not supported for listen, ignoring it";
return "";
}
if (!-e $file)
{
Log3 $name,3,"$name, audio file $file not found, ignoring it";
return "";
}
if (($file !~ /\.al(.+)$/) && ($file !~ /\.ul(.+)$/))
{
Log3 $name,3,"$name, audio file $file not type .alaw or .ulaw, ignoring it";
return "";
}
return $file;
}
1;
=pod
@ -906,7 +1260,7 @@ sub SIP_readPassword($)
<li>
<code>set &lt;name&gt; call &lt;number&gt [&lt;ringtime&gt] [&lt;message&gt]</code><br>
Start a call to the given number.<br>
Optionally you can supply a ringtime. If not given the value from attribute sip_ringtime is taken. Default is 10.
Optionally you can supply a max time. Default is 10.
Optionally you can supply a message which is either a full path to an audio file or a relativ path starting from the home directory of the fhem.pl.
</li>
<li>
@ -923,11 +1277,14 @@ sub SIP_readPassword($)
<a name="SIPattr"></a>
<b>Attributes</b>
<ul>
<li><a href="#sip_audiofile">sip_audiofile</a><br>
<li><a href="#sip_audiofile_wfp">sip_audiofile_wfp</a><br>
Audio file that will be played after <b>fetch</b> command. The audio file has to be generated via <br>
sox &lt;file&gt;.wav -t raw -r 8000 -c 1 -e a-law &lt;file&gt;.alaw<br>
sox &lt;file&gt;.wav -t raw -r 8000 -c 1 -e a-law &lt;file&gt;.al<br>
since only raw audio format is supported.
</li>
<li><a href="#sip_audiofile_call">sip_audiofile_call</a></li>
<li><a href="#sip_audiofile_dtmf">sip_audiofile_dtmf</a></li>
<li><a href="#sip_audiofile_ok">sip_audiofile_ok</a></li>
<li><a href="#sip_listen">sip_listen</a>(none , dtmf , wfp)</li>
<li><a name="#sip_from">sip_from</a><br>
My sip client info, defaults to sip:620@fritz.box
@ -942,17 +1299,20 @@ sub SIP_readPassword($)
Hostname or IP address of the SIP server you are connecting to, defaults to fritz.box.
</li>
<li><a name="#sip_ringtime">sip_ringtime</a><br>
Calltime for outgoing calls. Please dont use it now, it will be changed in a later version !
Ringtime for incomming calls (dtmf &wfp)
</li>
<li><a name="#sip_user">sip_user</a><br>
User name of the SIP client, defaults to 620.
</li>
<li><a name="#sip_waittime">sip_waittime</a><br>
Maximum waiting time in state listen_for_nwfp it will wait to pick up the call.
Maximum waiting time in state listen_for_wfp it will wait to pick up the call.
</li>
<li><a name="#sip_dtmf_size">sip_dtmf_size</a><br>
1 to 4 , default is 2 ...
</li>
<li><a name="#sip_dtmf_loop">sip_dtmf_loop</a><br>
once or loop , default once ...
</li>
</ul>
<br>
@ -997,7 +1357,7 @@ sub SIP_readPassword($)
<li>
<code>set &lt;name&gt; call &lt;nummer&gt [&lt;ringtime&gt] [&lt;nachricht&gt]</code><br>
Startet einen Anruf an die angegebene Nummer.<br>
Optional kann die ringtime angegeben werden. Wird keine angegeben zieht das Attribut sip_ringtime. Default ist 10.<br>
Optional kann die maximale Zeit angegeben werden. Default ist 10.<br>
Optional kann eine Nachricht in Form eines Audiofiles angegeben werden . Das File ist mit dem vollen Pfad oder dem relativen ab dem Verzeichnis mit fhem.pl anzugeben..
</li>
<li>
@ -1013,11 +1373,15 @@ sub SIP_readPassword($)
<a name="SIPattr"></a>
<b>Attributes</b>
<ul>
<li><a href="#sip_audiofile">sip_audiofile</a><br>
<li><a href="#sip_audiofile_wfp">sip_audiofile_wfp</a><br>
Audiofile das nach dem Command <b>fetch</b> abgespielt wird. Das Audiofile kann mit dem externen Programm sox erzeugt werden :<br>
sox &lt;file&gt;.wav -t raw -r 8000 -c 1 -e a-law &lt;file&gt;.alaw<br>
sox &lt;file&gt;.wav -t raw -r 8000 -c 1 -e a-law &lt;file&gt;.al<br>
da nur das raw audio format unterstützt wird.
</li>
</li>
<li><a href="#sip_audiofile_call">sip_audiofile_call</a></li>
<li><a href="#sip_audiofile_dtmf">sip_audiofile_dtmf</a></li>
<li><a href="#sip_audiofile_ok">sip_audiofile_ok</a></li>
<li><a href="#sip_listen">sip_listen</a> (none , dtmf, wfp)</li>
<li><a name="#sip_from">sip_from</a><br>
Meine SIP-Client-Info. Default ist sip:620@fritz.box
@ -1032,7 +1396,7 @@ sub SIP_readPassword($)
Hostname oder IP-Addresse des SIP-Servers mit dem sich der Client verbindet. Default ist fritz.box.
</li>
<li><a name="#sip_ringtime">sip_ringtime</a><br>
Klingelzeit für ausgehende Anrufe.
Klingelzeit für eingehende Anrufe.(dtmf & wfp)
</li>
<li><a name="#sip_user">sip_user</a><br>
User Name des SIP-Clients. Default ist 620.
@ -1040,6 +1404,10 @@ sub SIP_readPassword($)
<li><a name="#sip_dtmf_size">sip_dtmf_size</a><br>
1 bis 4 , default 2 Legt die L&auml;ge des erwartenden DTMF Events fest.
</li>
<li><a name="#sip_dtmf_loop">sip_dtmf_loop</a><br>
once oder loop , default once ...
</li>
<li><a name="#sip_waittime">sip_waittime</a><br>
Maximale Wartezeit im Status listen_for_wfp bis das Gespräch automatisch angenommen wird.
</li>