[Asterisk-Users] i4l - summary of patches?

[Iain McWilliams]

Thanks to both Iain & Oliver, but it looks like the i4l driver wasn't my
problem. I had been testing using a SIP softphone (X-Lite) and that appears
to be the problem. See the

[Asterisk-Users] soft phones -- voice quality tuning

thread for exactly the same problem posted by another user. After I realised
that I set up a test of calling in on one channel and then dialing out with
no quality problems at all.

Thanks again,
Iain

----- Original Message -----
From: "Iain Stevenson" <iain at iainstevenson.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, June 18, 2003 12:17 PM
Subject: Re: [Asterisk-Users] i4l - summary of patches?


>
> You probably want to remove the i4l handling of DTMF and silence
> suppression.  You can do this by commenting out the signal processing
> routines in the kernel i4l code.  This stops wasted work from being done.
> I posted the patch below a while back - not sure it it still works.
>
>   Iain
>
>
>
> --- /build/linux-2.4.20/drivers/isdn/isdn_tty.c Fri Dec 21 17:41:54 2001
> +++ isdn_tty.c  Sat Feb  1 09:14:33 2003
> @@ -133,9 +133,9 @@
>                         if (info->online) {
>                                 r = 0;
>  #ifdef CONFIG_ISDN_AUDIO
> -                               isdn_audio_eval_dtmf(info);
> -                               if ((info->vonline & 1) &&
> (info->emu.vpar[1]))
> -                                       isdn_audio_eval_silence(info);
> +//                             isdn_audio_eval_dtmf(info);
> +//                             if ((info->vonline & 1) &&
> (info->emu.vpar[1]))
> +//                                     isdn_audio_eval_silence(info);
>  #endif
>                                 if ((tty = info->tty)) {
>                                         if (info->mcr & UART_MCR_RTS) {
> @@ -190,10 +190,10 @@
>  #ifdef CONFIG_ISDN_AUDIO
>         ifmt = 1;
>
> -       if ((info->vonline) && (!info->emu.vpar[4]))
> -               isdn_audio_calc_dtmf(info, skb->data, skb->len, ifmt);
> -       if ((info->vonline & 1) && (info->emu.vpar[1]))
> -               isdn_audio_calc_silence(info, skb->data, skb->len, ifmt);
> +//     if ((info->vonline) && (!info->emu.vpar[4]))
> +//             isdn_audio_calc_dtmf(info, skb->data, skb->len, ifmt);
> +//     if ((info->vonline & 1) && (info->emu.vpar[1]))
> +//             isdn_audio_calc_silence(info, skb->data, skb->len, ifmt);
>  #endif
>         if ((info->online < 2)
>  #ifdef CONFIG_ISDN_AUDIO
>
>
>
>
> --On Tuesday, June 17, 2003 11:36 pm +0200 The Traveller
> <traveler at xs4all.nl> wrote:
>
> > Yo Iain,
> >
> > On Tue, Jun 17, 2003 at 21:48:34 +0100, Iain McWilliams wrote:
> >
> >> Hi,
> >>
> >> I'm trying to get asterisk running on kernel 2.4.20 however trawling
> >> through the archives I've found a few references to patches to remove
> >> i4l's dtmf detection, but have been unable to find the patch itself (I
> >> think it is isdn_audio.c). Can anyone point me in the right direction?
> >>
> >> The problem I'm seeing is connecting a SIP softphone (tried a few) to
an
> >> external number via an Hisax type 35 isdn card causes 1-2 sec silences
in
> >> the audio on the SIP phone.
> >>
> >> Any pointers?
> >
> > I recently did this myself.  Here's my patch.  Works for me.  Your
> > mileage may vary.  :-)
> >
> >
> >
> >    Grtz,
> >
> >      Oliver
> >
> >
> > Index: channels/chan_modem_i4l.c
> > ===================================================================
> > RCS file: /usr/cvsroot/asterisk/channels/chan_modem_i4l.c,v
> > retrieving revision 1.2
> > diff -u -r1.2 chan_modem_i4l.c
> > --- channels/chan_modem_i4l.c 27 Apr 2003 21:34:27 -0000 1.2
> > +++ channels/chan_modem_i4l.c 17 Jun 2003 21:52:04 -0000
> > @@ -265,7 +265,7 @@
> >  if (option_debug)
> >  ast_log(LOG_DEBUG, "Ignoring Escaped character '%c' (%d)\n", esc,
> > esc);  return &p->fr;
> > - case '0':
> > + /* case '0':
> >  case '1':
> >  case '2':
> >  case '3':
> > @@ -280,7 +280,7 @@
> >  ast_log(LOG_DEBUG, "DTMF: '%c' (%d)\n", esc, esc);
> >  p->fr.frametype=AST_FRAME_DTMF;
> >  p->fr.subclass=esc;
> > - return &p->fr;
> > + return &p->fr; */
> >  case 0: /* Pseudo signal */
> >  return &p->fr;
> >  default:
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
>
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