[Asterisk-Users] Bugfix for CVS-HEAD-06/26/04-21:56:45

Chris Saw chriss at watertech.com
Wed Jun 30 12:02:11 MST 2004 

I am encountering the same issues EXACTLY with a GrandStream BT101. I had to
downgrade to the CVS as of 6/7 in order to get the grandstreams to work. It
seems that a change thereafter is causing this issue...

it only happens when called from PSTN -> Asterisk. Calls through asterisk
between phones don't show this and calls from the phone -> PSTN don't do
this either...

    -Chris

----- Original Message -----
From: "programmer_ted" <ted at fusionapple.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, June 30, 2004 11:39 AM
Subject: [Asterisk-Users] Bugfix for CVS-HEAD-06/26/04-21:56:45


> Hiya,
>
> I sent this bugfix to the asterisk-dev mailing list, and modified it as I
> noticed side effects, but now it appears to be finished.  Nobody seemed to
> notice it there, so I thought I'd post here, as it seems to be something
> that will be needed as people update to the latest CVS version.  So...read
> on :)
>
> Ted
> programmer_ted at hotmail.com
>
> P.S. Read to the very end.  The original bugfix has an annoying side
effect.
>
>
> >>>>>Hi,
> >>>>>
> >>>>>My friend and I were getting a warning when calling his Sipura from a
> >>>>>PSTN line (connecting to Asterisk through BroadVoice), that said:
> >>>>>
> >>>>>Asked to transmit frame type 64, while native formats is 4
(read/write
> >>>>>= 4/4)
> >>>>>
> >>>>>and was followed by a hangup (type 64 is 16-bit Signed Linear PCM,
> >>>>>type 4 is G711u).  I found that many people have had similar issues,
> >>>>>but these were never resolved.  So, I figured that because Asterisk
is
> >>>>>open-source, I'd dive into the code and try to fix the bug.
> >>>>>
> >>>>>After a couple of hours of familiarizing myself with the Asterisk
code
> >>>>>and tracing the problem, I found that for some reason the tone
> >>>>>generator, which uses 16-bit Signed Linear PCM, was still being
> >>>>>allocated and playtones_generator (indications.c) was still getting
> >>>>>called, regardless that the Sipura doesn't take SLINEAR data (in my
> >>>>>case, it accepts G711u).  So, I ended up adding an if conditional to
> >>>>>the beginning of the playtones_alloc function (indications.c) to
check
> >>>>>if SLINEAR was supported by the channel, and if not, return 0 (which,
> >>>>>when received by the ast_activate_generator function (channel.c),
> >>>>>causes the channel generatordata to remain empty, effectively
stopping
> >>>>>the SLINEAR data from being sent in the most nonintrusive way
possible).
> >>>>>
> >>>>>NOTICE: this bugfix will work for similar issues involving format 64
> >>>>>(16-bit Signed Linear PCM) being sent even if channel capabilities
> >>>>>don't allow it, if the generator is involved - it's not limited to my
> >>>>>situation (dialing the Sipura from Asterisk).
> >>>>>
> >>>>>This patch should be applied to indications.c under the main asterisk
> >>>>>source directory (usually /usr/src/asterisk):
> >>>>>
> >>>>>68a69
> >>>>> >       if (!(chan->nativeformats & AST_FORMAT_SLINEAR)) return 0;
> >>>>>
> >>>>>Oh, and finally, here's a shameless plug to a good friend's website
> >>>>>(which includes a VOIP forum!): http://outcast.ws
> >>>>>
> >>>>>Comments?  Questions?  :)
> >>>>
> >>>>Just a quick update.  I was looking things over again and it appears
> >>>>this fix also disables the generator when I'm calling in on PSTN over
> >>>>BroadVoice (when dialing the Sipura), not just disabling it for the
> >>>>Sipura.  This basically disables the dialing sound while waiting for
> >>>>the Sipura to pick up.  I have an idea that I should have used
> >>>>chan->capabilities rather than chan->nativeformats, but it's too late
> >>>>to check at the moment.  I'll try it out first thing tomorrow and
> >>>>update you guys, but for now, that's one drawback of using this fix.
> >>>
> >>>I thought it over a little bit more and the optimum solution would be
to
> >>>just translate the SLINEAR data to a format that is recognized by
> >>>whoever is receiving the data, thus eliminating all drawbacks.  I'm
> >>>going to try using capabilities rather than nativeformats as a quick
> >>>workaround (after debugging to see if it'll work), and then work on
> >>>adding the translating code to sip_write.  Actually, thinking about it
> >>>again, it'd probably be best to just translate at the
> >>>playtones_generator function.  I'll keep you guys updated.
> >>>
> >>>...snipped non-relevant signature info etc...
> >>
> >>Learning as I go.  It appears I don't have access to the capabilities
> >>value from the ast_channel structure.  I'm just gonna go ahead and have
> >>the SLINEAR data translate to the channel's writeformat.
> >
> >Ok, as I thought, PSTN over BroadVoice does not understand SLINEAR
> >natively, which is why the dialing sound was also disabled when I dialed
> >the Sipura.  I added some code to playtones_alloc (indications.c) so that
> >the write format is only set to SLINEAR if it's supported, and added some
> >code to playtones_generator to translate from SLINEAR to the channel's
> >writeformat if SLINEAR isn't supported natively by the channel.  Of
> >course, I also had to include the translate.h header.
> >
> >Conclusion: playtones_generator now works regardless of SLINEAR support
by
> >the channel, as long as a translator path can be found from SLINEAR to
the
> >channel's writeformat.  If SLINEAR is supported, no translation takes
> >place.  This should fix some bugs where format 64 is being sent
regardless
> >of codec allow settings in the configuration files.
> >
> >Apply this patch to indications.c:
> >
> >28a29
> > > #include <asterisk/translate.h>         /* Needed for bugfix */
> >75c76
> ><       if (ast_set_write_format(chan, AST_FORMAT_SLINEAR)) {
> >---
> > >       if ((chan->nativeformats & AST_FORMAT_SLINEAR) &&
> > ast_set_write_format(chan, AST_FORMAT_SLINEAR)) {
> >128c129,142
> ><       ast_write(chan, &ps->f);
> >---
> > >
> > >       // Now, we have a finished SLINEAR frame that we need to
> > translate, IF
> > >       // the channel doesn't support SLINEAR.  Otherwise, we need to
just
> > >       // write the SLINEAR frame.
> > >       if (!(chan->nativeformats & AST_FORMAT_SLINEAR)) {
> > >               struct ast_trans_pvt* transPath =
> > ast_translator_build_path(chan->writeformat, AST_FORMAT_SLINEAR);
> > >               struct ast_frame* transFrame = ast_translate(transPath,
> > &ps->f, 0);
> > >               if (transFrame) {
> > >                       ast_write(chan, transFrame);
> > >                       ast_frfree(transFrame);
> > >               }
> > >               ast_translator_free_path(transPath);
> > >       }
> > >       else ast_write(chan, &ps->f);
> >
> >Hopefully, this fixes the problem for good.
>
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