asterisk users - Oct 2009 - After call into console/dsp hangup hear ringing

I am running asterisk 1.4.26.1 and using ALSA not oss
dahdi 2.2.0
and libpri-1.4.10

I am calling into console/dsp I hear the audio just fine then after the 
hangup I hear ringing
on the console/dsp.

Why would that be?

I found this bug for OSS https://issues.asterisk.org/view.php?id=13686
Does the same thing exist in ALSA???

some traces below

Jerry

  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.26.1 currently running on ebox3850 (pid = 3902)
Verbosity was 0 and is now 5
    -- Executing [mediaport_audio_visual at smvoice-mediaport:1] 
Goto("SIP/devcentos5x64_to_ebox3850-0837c170", 
"smvoice-mediaport-audio-visual|s|1") in new stack
    -- Goto (smvoice-mediaport-audio-visual,s,1)
    -- Executing [s at smvoice-mediaport-audio-visual:1] 
ChanIsAvail("SIP/devcentos5x64_to_ebox3850-0837c170",
"Console/Dsp") in
new stack
 << Hangup on console >>
    -- Executing [s at smvoice-mediaport-audio-visual:2] 
GotoIf("SIP/devcentos5x64_to_ebox3850-0837c170",
"0?smvoice-busy|s|1")
in new stack
    -- Executing [s at smvoice-mediaport-audio-visual:3] 
Playback("SIP/devcentos5x64_to_ebox3850-0837c170", "beep")
in new stack
    -- <SIP/devcentos5x64_to_ebox3850-0837c170> Playing 'beep'
(language
'en')
    -- Executing [s at smvoice-mediaport-audio-visual:4] 
Dial("SIP/devcentos5x64_to_ebox3850-0837c170",
"Console/dsp") in new stack
 << Call placed to 'dsp' on console >>
 << Auto-answered >>
    -- Called dsp
    -- ALSA/default answered SIP/devcentos5x64_to_ebox3850-0837c170
 << Hangup on console >>
  == Spawn extension (smvoice-mediaport-audio-visual, s, 4) exited 
non-zero on 'SIP/devcentos5x64_to_ebox3850-0837c170'



[smvoice-mediaport-audio-visual]
exten => s,1,ChanIsAvail(Console/Dsp)
exten => s,n,GotoIf($["${AVAILCHAN}" =
""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/dsp)
exten => s,n,Hangup

Looking at the function below in chan_alsa.c shouldnt't the variable
hookstate be set to 1 for auto answer??? I think this is affecting the 
ringing I hear after the call.

Jerry

static int alsa_call(struct ast_channel *c, char *dest, int timeout)
{
        int res = 3;
        struct ast_frame f = { AST_FRAME_CONTROL };
        ast_mutex_lock(&alsalock);
        ast_verbose(" << Call placed to '%s' on console
>> \n", dest);
        if (autoanswer) {
                ast_verbose(" << Auto-answered >> \n");
                grab_owner();
                if (alsa.owner) {
                        f.subclass = AST_CONTROL_ANSWER;
                        ast_queue_frame(alsa.owner, &f);
                        ast_mutex_unlock(&alsa.owner->lock);
                }
        } else {
                ast_verbose(" << Type 'answer' to answer, or
use
'autoanswer' for future calls >> \n");
                grab_owner();
                if (alsa.owner) {
                        f.subclass = AST_CONTROL_RINGING;
                        ast_queue_frame(alsa.owner, &f);
                        ast_mutex_unlock(&alsa.owner->lock);
                }
                if (write(sndcmd[1], &res, sizeof(res)) < 0) {

I found the problem.

The function send_sound() in chan_alsa.c does this:
               if (FD_ISSET(sndcmd[0], &rfds)) {
                        if (read(sndcmd[0], &cursound, sizeof(cursound)) 
< 0) {
                                ast_log(LOG_WARNING, "read() failed: 
%s\n", strerror(errno));
                        }
                        silencelen = 0;
                        offset = 0;
                        sampsent = 0;


This sets cursound to 0 and cursound should be -1.

If I change cursound here to local_cursound and recompile I do not hear 
the ring after my call.

Jerry

On 5/10/09 3:06 PM, Jerry Geis wrote:
> I found the problem.
>
> The function send_sound() in chan_alsa.c does this:
Best bet is to open an issue on the bugtracker and post your patch

-- 
Cheers,

Matt Riddell
Director
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