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Code Issues Releases Wiki Activity

96_SIP: add repeat function (Forum: #613729)

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git-svn-id: https://svn.fhem.de/fhem/trunk@13858 2b470e98-0d58-463d-a4d8-8e2adae1ed80
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Wzut 2017-03-31 05:40:05 +00:00
parent 0301beccec
commit 39cea8420b
1 changed files with 293 additions and 141 deletions
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392
fhem/FHEM/96_SIP.pm
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@ -45,6 +45,7 @@ package main;
use strict; use strict;
use warnings; use warnings;
use POSIX qw( strftime );
use Net::SIP qw//; use Net::SIP qw//;
use Net::SIP::Packet; use Net::SIP::Packet;
use IO::Socket; use IO::Socket;
@ -53,7 +54,7 @@ use Net::Domain qw(hostname hostfqdn);
use Blocking; # http://www.fhemwiki.de/wiki/Blocking_Call use Blocking; # http://www.fhemwiki.de/wiki/Blocking_Call
#use Data::Dumper; #use Data::Dumper;
my $sip_version ="V1.48 / 25.03.17"; my $sip_version ="V1.5 / 31.03.17";
my $ua; # SIP user agent my $ua; # SIP user agent
my @fifo; my @fifo;
@ -78,28 +79,30 @@ sub SIP_Initialize($$)
$hash->{GetFn} = "SIP_Get"; $hash->{GetFn} = "SIP_Get";
$hash->{NotifyFn} = "SIP_Notify"; $hash->{NotifyFn} = "SIP_Notify";
$hash->{AttrFn} = "SIP_Attr"; $hash->{AttrFn} = "SIP_Attr";
$hash->{AttrList} = "sip_watch_listen ". $hash->{AttrList} = "sip_watch_listen ". #
"sip_ringtime ". "sip_ringtime ". #
"sip_waittime ". "sip_waittime ". #
"sip_ip ". "sip_ip ". #
"sip_port ". "sip_port ". #
"sip_user ". "sip_user ". #
"sip_registrar ". "sip_registrar ". #
"sip_from ". "sip_from ". #
"sip_call_audio_delay:0,0.25,0.5,0.75,1,1.25,1.5,1.75,2,2.25,2.5,2.75,3 ". "sip_call_audio_delay:0,0.25,0.5,0.75,1,1.25,1.5,1.75,2,2.25,2.5,2.75,3 ". #
"sip_audiofile_call ". "sip_audiofile_call ". #
"sip_audiofile_dtmf ". "sip_audiofile_dtmf ". #
"sip_audiofile_ok ". "sip_audiofile_ok ". #
"sip_audiofile_wfp ". "sip_audiofile_wfp ". # CE
"sip_dtmf_size:1,2,3,4 ". "sip_dtmf_size:1,2,3,4 ". #
"sip_dtmf_send:audio,rfc2833 ". "sip_dtmf_send:audio,rfc2833 ". #
"sip_dtmf_loop:once,loop ". "sip_dtmf_loop:once,loop ". #
"sip_listen:none,dtmf,wfp ". "sip_listen:none,dtmf,wfp,echo ". #
"sip_filter ". "sip_filter ". #
"sip_blocking ". "sip_blocking ". #
"T2S_Device ". "sip_elbc:no,yes ". #
"T2S_Timeout ". "sip_force_interval ". #
"audio_converter:sox,ffmpeg ". "T2S_Device ". #
"T2S_Timeout ". #
"audio_converter:sox,ffmpeg ". #
"disabled:0,1 ".$readingFnAttributes; "disabled:0,1 ".$readingFnAttributes;
} }
@ -122,6 +125,7 @@ sub SIP_Define($$)
$attr{$name}{sip_dtmf_size} = '2' unless (exists($attr{$name}{sip_dtmf_size})); $attr{$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_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_dtmf_send} = 'audio' unless (exists($attr{$name}{sip_dtmf_send}));
$attr{$name}{sip_elbc} = 'no' unless (exists($attr{$name}{sip_elbc}));
$attr{$name}{sip_from} = 'sip:'.$attr{$name}{sip_user}.'@'.$attr{$name}{sip_registrar} unless (exists($attr{$name}{sip_from})); $attr{$name}{sip_from} = 'sip:'.$attr{$name}{sip_user}.'@'.$attr{$name}{sip_registrar} unless (exists($attr{$name}{sip_from}));
unless (exists($attr{$name}{sip_ip})) unless (exists($attr{$name}{sip_ip}))
@ -323,18 +327,19 @@ sub SIP_Register($$$)
} }
close($leg); close($leg);
$leg = $ip.":".$port;
my $registrar = AttrVal($name,"sip_registrar","fritz.box"); my $registrar = AttrVal($name,"sip_registrar","fritz.box");
my $user = AttrVal($name,"sip_user","620");
$leg = $ip.":".$port; my $from = AttrVal($name,"sip_from","sip:".$user."@".$registrar);
# create new agent # create new agent
$ua = Net::SIP::Simple->new( $ua = Net::SIP::Simple->new(
registrar => $registrar, registrar => $registrar,
domain => $registrar, domain => $registrar,
leg => $leg, leg => $leg,
from => AttrVal($name,"sip_from",'sip:620@fritz.box'), from => $from,
auth => [ AttrVal($name,"sip_user","620") , SIP_readPassword($name) ]); auth => [ $user , SIP_readPassword($name) ]);
# Register agent # Register agent
# optional registration # optional registration
@ -342,13 +347,11 @@ sub SIP_Register($$$)
$sub_register = sub $sub_register = sub
{ {
my $expire = $ua->register(registrar => $registrar ) || return "registration failed: ".$ua->error; my $expire = $ua->register(registrar => $registrar ) || return "registration failed: ".$ua->error;
my $parent = getppid(); my $cmd = "ps -e | grep '".$hash->{parent}." '";
Log3 $name,5,"$logname, my parent is $parent";
my $cmd = "ps -e | grep '".$parent." '";
my $result = qx($cmd); my $result = qx($cmd);
if (index($result,"perl") == -1) if (index($result,"perl") == -1)
{ {
Log3 $name, 2 , $logname.", cant find my parent ".$parent." in process list !"; Log3 $name, 2 , $logname.", can´t find my parent ".$hash->{parent}." in process list !";
die; die;
} }
Log3 $name,4,"$logname, register new expire : ".localtime(time()+$expire); Log3 $name,4,"$logname, register new expire : ".localtime(time()+$expire);
@ -373,27 +376,42 @@ sub SIP_CALLStart($)
{ {
my ($arg) = @_; my ($arg) = @_;
return unless(defined($arg)); return unless(defined($arg));
my ($name,$nr,$ringtime,$msg,$force) = split("\\|",$arg); my ($name,$nr,$ringtime,$msg,$repeat) = split("\\|",$arg);
my $hash = $defs{$name}; my $hash = $defs{$name};
my $logname = $name."[".$$."]"; my $logname = $name."[".$$."]";
$hash->{telnetPort}= undef; $hash->{telnetPort}= undef;
$ua = undef;
my $rtp_done = 0; my $rtp_done = 0;
my $dtmf = 'ABCD*#123--4567890';
my $port = AttrVal($name,"sip_port","5060");
my $delay = AttrVal($name,"sip_call_audio_delay",0); # Verzoegerung in 1/4 Sekunden Schritten
my $fi = 0;
#$repeat = 0 if (!$repeat);
my $packets = int($delay)*50;
my $timeout = 0;
my $final; my $final;
my $peer_hangup; my $peer_hangup;
my $peer_hangup2;
my $stopvar; my $stopvar;
my $state; my $state;
my $no_answer; my $no_answer;
my $dtmf = 'ABCD*#123--4567890';
my $port = AttrVal($name,"sip_port","5060");
my $call; my $call;
my $codec; my $codec;
my $call_established;
my @files;
my $anz;
my $sound_of_silence = sub
{
return unless $packets-- > 0;
return chr(0) x 160; # 160 bytes for PCMU/8000 = 1/50 Sekunde Sound
};
$ua = undef; $hash->{parent} = getppid();
Log3 $name,4,"$logname, my parent is ".$hash->{parent};
$hash->{telnetPort} = SIP_telnetPort(); $hash->{telnetPort} = SIP_telnetPort();
return $name."|no telnet port without password found" if (!$hash->{telnetPort}); return $name."|no telnet port without password found" if (!$hash->{telnetPort});
my $error = SIP_Register($hash,$port,"calling"); my $error = SIP_Register($hash,$port,"calling");
return $name."|0|CallRegister: $error" if ($error); return $name."|0|CallRegister: $error" if ($error);
@ -403,68 +421,130 @@ sub SIP_CALLStart($)
$codec = "PCMU/8000" if ($msg =~ /\.ul(.+)$/); $codec = "PCMU/8000" if ($msg =~ /\.ul(.+)$/);
return $name."|0|CallStart: please use filetype .alaw (for a-law) or .ulaw (for u-law)" if !defined($codec); return $name."|0|CallStart: please use filetype .alaw (for a-law) or .ulaw (for u-law)" if !defined($codec);
Log3 $name,4,"$logname, CallStart msg : $msg - $codec $force"; push @files,$sound_of_silence if ($delay);
for(my $i=0; $i<=$repeat; $i++) { push @files,$msg; }
$anz = @files;
Log3 $name,4,"$logname, CallStart with $anz files - first file : $files[0] - $codec , repeat $repeat";
$call = $ua->invite( $nr, $call = $ua->invite( $nr,
init_media => $ua->rtp('send_recv', $msg), init_media => $ua->rtp('send_recv', $files[0]),
cb_rtp_done => \$rtp_done, cb_rtp_done => \$rtp_done,
cb_final => sub { my ($status,$self,%info) = @_; cb_final => sub { my ($status,$self,%info) = @_;
$final = $info{code}; $final = $info{code};
my $delay = AttrVal($name,"sip_call_audio_delay",0); # Verzoegerung in 1/4 Sekunden Schritten Log3 $name,4,"$logname, cb_final - status : $status" if (!defined($final));
if (($status eq "FAIL") && defined($final) && $delay) Log3 $name,4,"$logname, cb_final - status : $status - final : $final" if (defined($final));
if (($status eq "FAIL") && defined($final))
{ {
if (int($final) == 481) if (int($final) == 481) { SIP_telnet($hash,"set $name call_state ringing\nexit\n"); }
{ elsif (int($final) == 486) { $fi=1 ; } # Anruf abgelehnt
Log3 $name,4, $logname.", waiting $delay seconds to play audio"; }
select(undef, undef, undef, $delay); elsif (($status eq "OK") && !defined($final) && !$call_established) # der Angrufene hat abgenommen
{
Log3 $name,4, $logname.", call established";
SIP_telnet($hash,"set $name call_state established\nexit\n");
$call_established++; # nur 1x , bei mehr als einem File kommen wir ofters hier vorbei
} }
} # der Anrufer hat abgenommen
}, },
recv_bye => \$peer_hangup, recv_bye => \$peer_hangup,
cb_noanswer => \$no_answer, #ring_time => 5,
#cb_noanswer => \$no_answer, klappt hier nicht wir gehen ueber add_timer
rtp_param => [8, 160, 160/8000, $codec]) || return $name."|0|invite failed: ".$ua->error; rtp_param => [8, 160, 160/8000, $codec]) || return $name."|0|invite failed: ".$ua->error;
} }
else else
{ {
$dtmf = (substr($msg,0,1) eq "-") ? substr($msg,1) : $dtmf; $dtmf = (substr($msg,0,1) eq "-") ? substr($msg,1) : $dtmf;
Log3 $name,4,"$name, CallStart DTMF : $dtmf $force"; Log3 $name,4,"$logname, CallStart DTMF : $dtmf";
$delay = 0; # wenn delay sein muss dann ueber DTMF ----
$repeat = 0; # keine Wiederholungen
$call = $ua->invite($nr, $call = $ua->invite($nr,
init_media => $ua->rtp( 'recv_echo',undef,0 ), init_media => $ua->rtp( 'recv_echo',undef,0 ),
rtp_param => [0, 160, 160/8000, 'PCMU/8000'], rtp_param => [0, 160, 160/8000, 'PCMU/8000'],
cb_final => sub { my ($status,$self,%info) = @_; $final = $info{code};}, cb_final => sub { my ($status,$self,%info) = @_;
cb_noanswer => \$no_answer, $final = $info{code};
recv_bye => \$peer_hangup) || return $name."|0|invite failed ".$ua->error."|".$force; Log3 $name,4,"$logname, cb_final - Status : $status" if (!defined($final));
Log3 $name,4,"$logname, cb_final - status : $status - final : $final" if (defined($final));
if (($status eq "FAIL") && defined($final))
{
if (int($final) == 481) { SIP_telnet($hash,"set $name call_state ringing\nexit\n"); }
elsif (int($final) == 486) { $fi=1 };
}
elsif (($status eq "OK") && !defined($final)) # der Angrufene hat abgenommen
{
Log3 $name,4, $logname.", call established";
SIP_telnet($hash,"set $name call_state established\nexit\n");
}
},
#cb_noanswer => \$no_answer,
#ring_time => 5, siehe oben -> add_timer
cb_cleanup => sub {0},
recv_bye => \$peer_hangup) || return $name."|0|invite failed ".$ua->error;
if (AttrVal($name,"sip_dtmf_send","audio") eq "audio") if (AttrVal($name,"sip_dtmf_send","audio") eq "audio")
{ $call->dtmf( $dtmf, methods => 'audio', duration => 500, cb_final => \$rtp_done); } { $call->dtmf( $dtmf, methods => 'audio', duration => 500, cb_final => \$rtp_done); }
else { $call->dtmf( $dtmf, cb_final => \$rtp_done); } else { $call->dtmf( $dtmf, cb_final => \$rtp_done); }
} }
return "$name|0|invite call failed ".$call->error."|".$force if ($call->error); return "$name|0|invite call failed ".$call->error if ($call->error);
SIP_telnet($hash,"set $name call_state calling $nr\nexit\n");
Log3 $name,4,"$logname, calling : $nr"; Log3 $name,4,"$logname, calling : $nr";
SIP_telnet($hash,"set $name call_state calling $nr\nexit\n");
return "$name|0|no answer" if ($no_answer); #return "$name|1|no answer" if ($no_answer);
$ua->add_timer($ringtime,\$stopvar); $ua->add_timer($ringtime,\$stopvar);
$ua->loop( \$stopvar,\$peer_hangup,\$rtp_done); $ua->loop( \$stopvar,\$peer_hangup,\$rtp_done,\$fi );
$timeout = 1 if defined($stopvar); # hat der bereits zugeschlagen ?
Log3 $name,5,"$logname, 0. Ende des ersten Loops";
Log3 $name,5,"$logname, 1. rtp_done : $rtp_done" if defined($rtp_done);
Log3 $name,5,"$logname, 2. fi : $fi" if defined($fi);
Log3 $name,5,"$logname, 3. timeout : $timeout";
Log3 $name,5,"$logname, 4. peer_hangup : $peer_hangup" if defined($peer_hangup);
# Lebt der Call noch und gibt es ueberhaupt etwas zum wiederholen ?
while ( !$peer_hangup && !$peer_hangup2 && !$fi && !$stopvar && $msg && ($anz > 1))
{
shift(@files); # done with file
@files || last; # raus hier sobald kein File mehr da ist
Log3 $name,4,"$logname, next file : $files[0]" if defined($files[0]);
Log3 $name,3,"$logname, opps no file" if !defined($files[0]);
# re-invite on current call for next file
$rtp_done = undef; # wichtig ! u.U. haengen wir hier fest wenn der Anrufer jetzt auflegt
select(undef, undef, undef, 0.1); # minimale pause
$call->reinvite(
init_media => $ua->rtp('send_recv', $files[0]),
#rtp_param => [0, 160, 160/8000, 'PCMU/8000'], unebeding weglassen ! fuehrt zu Verzerrungen bei der Wiedergabe :(
cb_rtp_done => \$rtp_done,
recv_bye => \$peer_hangup2, # FIXME: do we need to repeat this? Wzut : I think so ...
) || return $name."|0|reinvite failed: ".$ua->error;
$ua->loop( \$rtp_done,\$peer_hangup2,\$peer_hangup,\$stopvar );
}
$timeout = 1 if defined($stopvar); # nach eventuellen reinvte nochmal testen
# timeout or dtmf done, hang up # timeout or dtmf done, hang up
if ( $stopvar || $rtp_done ) if ( $timeout || $rtp_done)
{ {
$stopvar = undef; $stopvar = undef;
$call->bye( cb_final => \$stopvar ); $call->bye( cb_final => \$stopvar );
$ua->loop( \$stopvar ); $ua->loop( \$stopvar );
} }
Log3 $name,5,"$logname, RTP done : $rtp_done" if defined($rtp_done); Log3 $name,5,"$logname, RTP done : $rtp_done" if defined($rtp_done);
Log3 $name,5,"$logname, Hangup : $peer_hangup" if defined($peer_hangup); Log3 $name,5,"$logname, Hangup : $peer_hangup" if defined($peer_hangup);
Log3 $name,5,"$logname, Stopvar : $stopvar" if defined($stopvar); Log3 $name,5,"$logname, Hangup2 : $peer_hangup2" if defined($peer_hangup2);
Log3 $name,5,"$logname, Timeout : $timeout";
Log3 $name,5,"$logname, Final : $final" if defined($final); Log3 $name,5,"$logname, Final : $final" if defined($final);
if (defined($rtp_done)) if (defined($rtp_done))
{ {
if ($rtp_done eq "OK") {return $name."|1|ok|$force";} # kein Audio if ($rtp_done eq "OK") {return $name."|1|ok";} # kein Audio
else else
{ {
if (defined($final)) if (defined($final))
@ -472,17 +552,21 @@ sub SIP_CALLStart($)
my $txt; my $txt;
$txt = "canceled" if (int($final) == 486); $txt = "canceled" if (int($final) == 486);
$txt = "no answer" if (int($final) == 487); $txt = "no answer" if (int($final) == 487);
$final = $txt if defined($txt); $txt = "declined" if (int($final) == 603);
return $name."|1|$txt" if $txt;
} }
else {return $name."|1|ok|$force" if ($rtp_done !=0);} else {return $name."|1|ok" if ($rtp_done !=0);}
} }
} }
# immer noch kein richtiger Test zur Rueckgabe ?
$final = "unknown" if !defined($final); $final = "unknown" if (!defined($final) && !$timeout);
$final .= " peer hangup" if defined($peer_hangup); $final = "timeout" if (!defined($final) && $timeout);
$final = "peer hangup" if defined($peer_hangup);
$final = "peer_hangup" if defined($peer_hangup2); # ts,ts hat der doch glatt im reinvite noch abgebochen
return $name."|1|".$final."|$force"; return $name."|1|".$final;
} }
sub SIP_CALLDone($) sub SIP_CALLDone($)
@ -494,8 +578,8 @@ sub SIP_CALLDone($)
my $hash = $defs{$r[0]}; my $hash = $defs{$r[0]};
my $error = (defined($r[1])) ? $r[1] : "0"; my $error = (defined($r[1])) ? $r[1] : "0";
my $final = (defined($r[2])) ? $r[2] : "???"; my $final = (defined($r[2])) ? $r[2] : "???";
my $force = (defined($r[3])) ? $r[3] : "";
my $name = $hash->{NAME}; my $name = $hash->{NAME};
my @a;
Log3 $name, 4,"$name, CALLDone -> $string"; Log3 $name, 4,"$name, CALLDone -> $string";
@ -521,36 +605,51 @@ sub SIP_CALLDone($)
readingsEndUpdate($hash, 1); readingsEndUpdate($hash, 1);
} }
if ($force && defined($hash->{CALL}) && (lc($final) ne "ok")) my (undef,$nr,$ringtime,$msg,$repeat,$force) = split("\\|",$hash->{CALL}); # zerlegen wir den Original Call
if ($force && (lc($final) ne "ok"))
{ {
my (undef,$nr,$ringtime,$msg,$force) = split("\\|",$hash->{CALL}); $repeat = "0" if (!$repeat);
#$nr =~ s/[^0-9]//g;
my $nr2 = $nr; my $nr2 = $nr;
$nr2 =~ tr/0-9//cd; $nr2 =~ tr/0-9//cd;
CommandDefine(undef, "at_forcecall_".$nr2." at +00:01:00 set $name call $nr $ringtime $msg $force");
$attr{"at_forcecall_".$nr2}{room} = AttrVal($name,"room","Unsorted"); my $t = int(AttrVal($name,"sip_force_interval",60));
delete $hash->{CALL}; my $time_s = strftime("\%H:\%M:\%S", gmtime($t));
$error = CommandDefine(undef, "at_forcecall_".$nr2." at +".$time_s." set $name call $nr $ringtime $msg *$repeat &");
if (!$error) { $attr{"at_forcecall_".$nr2}{room} = AttrVal($name,"room","Unsorted"); }
else { Log3 $name,2,"$name, $error"; }
} }
my $call = shift @fifo; my $nextcall = shift @fifo; # sind da noch Calls in der Queue ?
if ($call) if ($nextcall)
{ {
my @a = split(" ",$call); @a = split(" ",$nextcall);
SIP_Set($hash,@a); $error = SIP_Set($hash,@a);
} Log3 $name,3,"$name, error setting nextcall $nextcall -> $error" if ($error);
return undef;
} else { Log3 $name,5,"$name, fifo is empty"; }
if (exists($hash->{'.elbc'}))
{
@a = (undef,"listen");
Log3 $name,4,"$name, try restarting listen process after call ends";
$error = SIP_Set($hash,@a);
Log3 $name,3,"$name, error restarting listen -> $error" if ($error);
delete $hash->{'.elbc'};
} else { Log3 $name,5,"$name, no elbc"; }
delete $hash->{CALL};
return undef; return undef;
} }
##################################### #####################################
sub SIP_Set($@) sub SIP_Set($@)
{ {
my ($hash, @a) = @_; my ($hash, @a) = @_;
my $name = $hash->{NAME}; my $name = $hash->{NAME};
my $cmd = (defined($a[1])) ? $a[1] : "??"; my $cmd = (defined($a[1])) ? $a[1] : "?";
my $subcmd; my $subcmd;
return join(" ", sort keys %sets) if ($cmd eq "?"); return join(" ", sort keys %sets) if ($cmd eq "?");
@ -574,25 +673,43 @@ sub SIP_Set($@)
return "missing target call number" if (!$nr); return "missing target call number" if (!$nr);
return "invalid max time : $ringtime" unless $ringtime =~ m/^\d+$/; return "invalid max time : $ringtime" unless $ringtime =~ m/^\d+$/;
my $anz = @a;
$anz--; # letztes Element
my $force = ($a[$anz] eq "&") ? "&" : "";
Log3 $name,3,"$name, force call" if ($force);
if (exists($hash->{CPID})) if (exists($hash->{CPID}))
{ {
#return "there is already a call activ for target $nr" if (defined($hash->{lastnr}) && ($hash->{lastnr} eq $nr)); #return "there is already a call activ for target $nr" if (defined($hash->{lastnr}) && ($hash->{lastnr} eq $nr));
my $call = join(" ",@a); my $call = join(" ",@a);
push (@fifo,$call); push (@fifo,$call);
Log3 $name ,4,"$name, add $call to fifo"; Log3 $name ,4,"$name, add call -> $call to fifo we will do it later";
return undef; return undef;
} }
my $anz = @a;
$anz--; # letztes Element
my $force = ($a[$anz] eq "&") ? 1 : 0;
$anz-- if ($force); # checken wir dann noch auf repeat
my $repeat;
if (substr($a[$anz],0,1) ne "*")
{ $repeat = 0; }
else
{
$repeat = $a[$anz];
$repeat =~ s/^\*//;
$repeat ++; $repeat --;
} # * weg , Rest als Int
Log3 $name,3,"$name, force call" if ($force);
Log3 $name,4,"$name, msg will be repeat $repeat times" if ($repeat);
if (exists($hash->{LPID}) && (AttrVal($name,"sip_elbc","no") ne "no"))
{
Log3 $name,4,"$name, listen process $hash->{LPID} must be killed befor starting call !";
BlockingKill($hash->{helper}{LISTEN_PID});
delete $hash->{helper}{LISTEN_PID};
delete $hash->{LPID};
$hash->{'.elbc'} = 1; # haben wir gerade einen listen Prozess abgeschossen ?
}
if ($msg) if ($msg)
{ {
#Log3 $name, 4, $name.", sending $msg to $nr, ringtime: $ringtime";
if (substr($msg,0,1) eq "-") if (substr($msg,0,1) eq "-")
{ {
Log3 $name, 4, $name.", message DTMF = $msg"; Log3 $name, 4, $name.", message DTMF = $msg";
@ -605,16 +722,20 @@ sub SIP_Set($@)
$a[1] = "tts"; # Kommando des Set Befehls $a[1] = "tts"; # Kommando des Set Befehls
$a[2] =~ s/^\!//; # das ! muss weg $a[2] =~ s/^\!//; # das ! muss weg
pop @a if ($force); # das & muss ggf. auch noch weg pop @a if ($force); # das & muss ggf. auch noch weg
pop @a if ($repeat); # das * muss ggf. auch weg
# die nachsten vier brauchen wir unbedingt fuer T2S
$hash->{callnr} = $nr;
$hash->{ringtime} = $ringtime;
$hash->{forcecall} = $force;
$hash->{repeat} = $repeat;
my $ret = SIP_create_T2S_File($hash,@a); # na dann lege schon mal los my $ret = SIP_create_T2S_File($hash,@a); # na dann lege schon mal los
return $ret if defined($ret); # Das ging leider schief return $ret if defined($ret); # Das ging leider schief
readingsSingleUpdate($hash,"call_state","waiting T2S",0); readingsSingleUpdate($hash,"call_state","waiting T2S",0);
$hash->{callnr} = $nr; Log3 $name,1,"$name :set repeat $repeat";
$hash->{ringtime} = $ringtime;
$hash->{forcecall} = $force;
RemoveInternalTimer($hash); RemoveInternalTimer($hash);
# geben wir T2S mal ein paar Sekunden # geben wir T2S mal ein paar Sekunden
InternalTimer(gettimeofday()+int(AttrVal($name,"T2S_Timeout",5)), "SIP_wait_for_t2s", $hash); InternalTimer(gettimeofday()+int(AttrVal($name,"T2S_Timeout",5)), "SIP_wait_for_t2s", $hash);
@ -628,19 +749,22 @@ sub SIP_Set($@)
else else
{ {
Log3 $name, 3, $name.", message $msg NOT found !"; Log3 $name, 3, $name.", message $msg NOT found !";
$hash->{repeat} = 0;
$hash->{forcecall} = 0;
$msg = ""; $msg = "";
} }
} }
else { Log3 $name, 4, $name.", calling $nr, ringtime: $ringtime , no message"; } else { Log3 $name, 4, $name.", calling $nr, ringtime: $ringtime , no message"; }
my $arg = "$name|$nr|$ringtime|$msg|$force"; my $arg = "$name|$nr|$ringtime|$msg|$repeat"; # da muss force nicht mit
#BlockingCall($blockingFn, $arg, $finishFn, $timeout, $abortFn, $abortArg); #BlockingCall($blockingFn, $arg, $finishFn, $timeout, $abortFn, $abortArg);
$hash->{helper}{CALL_PID} = BlockingCall("SIP_CALLStart",$arg, "SIP_CALLDone") unless(exists($hash->{helper}{CALL_PID})); $hash->{helper}{CALL_PID} = BlockingCall("SIP_CALLStart",$arg, "SIP_CALLDone") unless(exists($hash->{helper}{CALL_PID}));
if($hash->{helper}{CALL_PID}) if($hash->{helper}{CALL_PID})
{ {
$hash->{CPID} = $hash->{helper}{CALL_PID}{pid}; $hash->{CPID} = $hash->{helper}{CALL_PID}{pid};
$hash->{CALL} = $arg; $hash->{CALL} = $arg."|$force"; # hier retten wir aber force
Log3 $name, 4, "$name, call -> ".$hash->{CALL};
Log3 $name, 5, "$name, call has pid ".$hash->{CPID}; Log3 $name, 5, "$name, call has pid ".$hash->{CPID};
readingsBeginUpdate($hash); readingsBeginUpdate($hash);
readingsBulkUpdate($hash, "call_state","invite"); readingsBulkUpdate($hash, "call_state","invite");
@ -666,7 +790,7 @@ sub SIP_Set($@)
{ {
my $type = AttrVal($name,"sip_listen","none"); my $type = AttrVal($name,"sip_listen","none");
return "there is already a listen process running with pid ".$hash->{LPID} if exists($hash->{LPID}); return "there is already a listen process running with pid ".$hash->{LPID} if exists($hash->{LPID});
return "please set attr sip_listen to dtmf or wfp first" if (AttrVal($name,"sip_listen","none") eq "none"); return "please set attr sip_listen to dtmf or wfp or echo first" if (AttrVal($name,"sip_listen","none") eq "none");
my $error = SIP_try_listen($hash); my $error = SIP_try_listen($hash);
if ($error) if ($error)
{ {
@ -767,8 +891,8 @@ sub SIP_ListenStart($)
return unless(defined($name)); return unless(defined($name));
my $logname = $name."[".$$."]"; my $logname = $name."[".$$."]";
my $hash = $defs{$name}; # $hash / $name gueltig in diesem Block my $hash = $defs{$name}; # $hash / $name gueltig in diesem Block
my $parent = getppid(); $hash->{parent} = getppid();
Log3 $name,3,"$logname, my parent is $parent"; Log3 $name,4,"$logname, my parent is ".$hash->{parent};
$hash->{telnetPort} = undef; $hash->{telnetPort} = undef;
my $dtmfloop; # Ende-Flag für die DTMF-Schleife my $dtmfloop; # Ende-Flag für die DTMF-Schleife
@ -1078,9 +1202,26 @@ sub SIP_ListenStart($)
rtp_param => [8, 160, 160/8000, 'PCMA/8000'] rtp_param => [8, 160, 160/8000, 'PCMA/8000']
); # options are invite and hangup ); # options are invite and hangup
} }
elsif (AttrVal($name,"sip_listen", "none") eq "echo")
{
$ua->listen(
filter => \&$sub_filter,
cb_create => \&$sub_create,
cb_invite => sub {
if (!$block_it)
{
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
}
},
init_media => $ua->rtp( 'recv_echo',undef,0 ),
rtp_param => [8, 160, 160/8000, 'PCMA/8000'],
recv_bye => \&$sub_bye,
);
}
else { return $name."|end"; } else { return $name."|end"; }
$ua->loop;
$ua->loop;
return $name."|end"; # hier sollten wir eigentlich nie himkommen ! return $name."|end"; # hier sollten wir eigentlich nie himkommen !
} }
@ -1299,11 +1440,16 @@ sub SIP_wait_for_t2s($)
} }
# nun aber calling # nun aber calling
$hash->{repeat} = "*".$hash->{repeat};
$hash->{forcecall} = ($hash->{forcecall}) ? "&" : "";
my @a = ($name,"call",$hash->{callnr}, $hash->{ringtime},$msg,$hash->{repeat},$hash->{forcecall}) ;
my $test = join(" ",@a);
Log3 $name ,1,"$name, Test : $test";
my @a = ($name,"call",$hash->{callnr}, $hash->{ringtime},$msg,$hash->{forcecall}) ;
delete($hash->{callnr}); delete($hash->{callnr});
delete($hash->{ringtime}); delete($hash->{ringtime});
delete($hash->{forcecall}); delete($hash->{forcecall});
delete($hash->{repeat});
my $ret = SIP_Set($hash , @a); my $ret = SIP_Set($hash , @a);
Log3 $name,3,"$name, T2S Call : $ret" if defined($ret); Log3 $name,3,"$name, T2S Call : $ret" if defined($ret);
return undef; return undef;
@ -1332,7 +1478,9 @@ sub SIP_telnet($$)
my ($hash,$cmd) = @_; my ($hash,$cmd) = @_;
my $name = $hash->{NAME}; my $name = $hash->{NAME};
my $logname = $name."[".$$."]"; my $logname = $name."[".$$."]";
Log3 $name, 5, "$logname, telnet : $cmd"; my $txt = $cmd;
$txt =~ s/\n/ /g;
Log3 $name, 5, "$logname, telnet : $txt";
my $sock = IO::Socket::INET->new( my $sock = IO::Socket::INET->new(
PeerHost => "127.0.0.1", PeerHost => "127.0.0.1",
@ -1663,13 +1811,32 @@ sub SIP_readPassword($)
<a name="SIPattr"></a> <a name="SIPattr"></a>
<b>Attributes</b> <b>Attributes</b>
<ul> <ul>
<li><a href="#sip_audiofile_wfp">sip_audiofile_wfp</a><br> <li><a name="#sip_user">sip_user</a><br>
Audiofile das nach dem Command <b>fetch</b> abgespielt wird. Das Audiofile kann mit dem externen Programm sox erzeugt werden :<br> User Name des SIP-Clients. Default ist 620 (Fritzbox erstes SIP Telefon)
</li>
<li><a name="#sip_registrar">sip_registrar</a><br>
Hostname oder IP-Addresse des SIP-Servers mit dem sich das Modul verbinden soll. (Default fritz.box)
</li>
<li><a name="#sip_from">sip_from</a><br>
SIP-Client-Info. Syntax : sip:sip_user@sip_registrar Default ist sip:620@fritz.box
</li>
<li><a name="#sip_ip">sip_ip</a><br>
Die IP-Addresse von FHEM im Heimnetz. Solange das Attribut nicht gesetzt ist versucht das Modul diese beim Start zu ermitteln.
</li>
<li><a name="#sip_port">sip_port</a><br>
Port der vom Modul genutzt wird. Default ist 5060 und wird automatisch um 10 erh&oml;ht wenn der Port nicht frei sein sollte.
</li>
<li><b>Audiofiles</b>
Audiofiles k&ouml;nnen einfach 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;.al<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<br> Unterst&uuml;tzt werden nur die beiden RAW Audio Formate a-law und u-law !<br>
Statt eines echten Audiofiles kann auch eine Text2Speech Nachricht eingetragen werden.<br> Statt eines echten Audiofiles kann auch eine Text2Speech Nachricht eingetragen werden.<br>
Bsp : attr mySIP sip_audiofile_call !Hier ist dein FHEM Server Bsp : attr mySIP sip_audiofile_call !Hier ist dein FHEM Server
</li> </li>
<li><a href="#sip_audiofile_wfp">sip_audiofile_wfp</a><br>
Audiofile das nach dem Command <b>fetch</b> abgespielt wird.
</li>
<li><a href="#sip_audiofile_call">sip_audiofile_call</a></li> <li><a href="#sip_audiofile_call">sip_audiofile_call</a></li>
Audiofile das dem Angerufenen bei set call vorgespielt wird. Audiofile das dem Angerufenen bei set call vorgespielt wird.
<li><a href="#sip_audiofile_dtmf">sip_audiofile_dtmf</a></li> <li><a href="#sip_audiofile_dtmf">sip_audiofile_dtmf</a></li>
@ -1677,24 +1844,9 @@ sub SIP_readPassword($)
<li><a href="#sip_audiofile_ok">sip_audiofile_ok</a></li> <li><a href="#sip_audiofile_ok">sip_audiofile_ok</a></li>
Audiofile das bei erkannter DTMF Sequenz abgespielt wird. Audiofile das bei erkannter DTMF Sequenz abgespielt wird.
<li><a href="#sip_listen">sip_listen</a> (none , dtmf, wfp)</li> <li><a href="#sip_listen">sip_listen</a> (none , dtmf, wfp)</li>
<li><a name="#sip_from">sip_from</a><br>
Meine SIP-Client-Info. Default ist sip:620@fritz.box
</li>
<li><a name="#sip_ip">sip_ip</a><br>
Die IP-Addresse des FHEM-Servers.
</li>
<li><a name="#sip_port">sip_port</a><br>
Port der für den SIP-Client genutzt wird. Default ist 5060 und wird automatisch um 10 erhöht wenn der Port nicht frei ist.
</li>
<li><a name="#sip_registrar">sip_registrar</a><br>
Hostname oder IP-Addresse des SIP-Servers mit dem sich der Client verbindet. Default ist fritz.box.
</li>
<li><a name="#sip_ringtime">sip_ringtime</a><br> <li><a name="#sip_ringtime">sip_ringtime</a><br>
Klingelzeit für eingehende Anrufe bei listen_for_dtmf Klingelzeit für eingehende Anrufe bei listen_for_dtmf
</li> </li>
<li><a name="#sip_user">sip_user</a><br>
User Name des SIP-Clients. Default ist 620.
</li>
<li><a name="#sip_dtmf_size">sip_dtmf_size</a><br> <li><a name="#sip_dtmf_size">sip_dtmf_size</a><br>
1 bis 4 , default 2 Legt die L&auml;ge des erwartenden DTMF Events fest. 1 bis 4 , default 2 Legt die L&auml;ge des erwartenden DTMF Events fest.
</li> </li>
@ -1702,16 +1854,16 @@ sub SIP_readPassword($)
once oder loop , default once ... once oder loop , default once ...
</li> </li>
<li><a name="#sip_waittime">sip_waittime</a><br> <li><a name="#sip_waittime">sip_waittime</a><br>
Maximale Wartezeit im Status listen_for_wfp bis das Gespräch automatisch angenommen wird. Maximale Wartezeit im Status listen_for_wfp bis das Gespr&auml;ch automatisch angenommen wird.
</li> </li>
<li>T2S_Device<br> <li>T2S_Device<br>
Name des Text2Speech Devices Name des Text2Speech Devices (Wird nur ben&ouml;tigt wenn Sprachnachrichten statt Audiofiles verwendet werden)
</li> </li>
<li>T2S_Timeout<br> <li>T2S_Timeout<br>
Wartezeit wie lange max auf Text2Speech gewartet wird. Wartezeit in Sekunden wie lange maximal auf Text2Speech gewartet wird.
</li> </li>
<li>audo_converter<br>sox oder ffmpeg, default sox<br> <li>audo_converter<br>sox oder ffmpeg, default sox<br>
f&uml;r Text2Speech unbedingt erforderlich !<br> Ist f&uml;r Text2Speech unbedingt erforderlich um die mp3 Dateien in Raw Audio umzuwandeln.<br>
Installation z.B. mit sudo apt-get install sox und noch die mp3 Unterst&uuml;tzung mit sudo apt-get install libsox-fmt-mp3 Installation z.B. mit sudo apt-get install sox und noch die mp3 Unterst&uuml;tzung mit sudo apt-get install libsox-fmt-mp3
</li> </li>
</ul> </ul>