[Asterisk-Users] Bugfix for CVS-HEAD-06/26/04-21:56:45
programmer_ted
ted at fusionapple.com
Thu Jul 1 10:18:44 MST 2004
reseaux: update to the latest CVS version (from /usr/src, do cvs update
asterisk), and give me an email address I can send a patched indications.c
to. Then put the file I send you in /usr/src/asterisk (overwrite the
existing indications.c), and do "make; make install" as root (without
asterisk running). That should stop the bug from killing your calls, but
after talking to bkw_, I think the bugfix may just cover up a larger
bug. By the way, would you happen to be using BroadVoice or another SIP
provider for Asterisk?
Holger: I'm going to file a bug report today. I asked a few questions
about it yesterday while talking to bkw_; just didn't have time to do the
report yesterday.
At 04:17 AM 7/1/2004, you wrote:
>Dear Ted
> i have notice the same problem had you reported from monday, i
> have try to
>update to today CVS HEAD but nothing still buggy so i roolback to Stable V1.
>Where i can find the pacth?
>Thanks in advance
>Dimitri
>
>On Wednesday 30 June 2004 08:39 pm, programmer_ted wrote:
> > 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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