[Asterisk-Users] Bugfix for CVS-HEAD-06/26/04-21:56:45
reseaux
reseauxit at yahoo.it
Fri Jul 2 06:20:03 MST 2004
Dear Ted
my problem is not related to Voip Provider i use two * box:
- Box1 is my local network PBX with 2 BG phone , TDM400 FXS and E100P
If i can from BG phone to TDM400 FXS i have oneway audio also if i call to
E100P this happen with yestarday CVS HEAD, but if rollback to stable branch
everything work (also with CVS HEAD of last week).
Thanks in advance
Dimitri
On Friday 02 July 2004 01:19 am, programmer_ted wrote:
> bkw: now, I'm pretty sure it's BroadVoice (or other SIP providers)
> related. The only people having trouble appear to be using a SIP provider
> for Asterisk. I'm drawing this conclusion from two factors: Chris and I
> are both using BroadVoice, and it appears that those people not using SIP
> providers don't experience the issue.
>
> At 10:36 AM 7/1/2004, you wrote:
> >Ya I tried to duplicate this problem but couldn't... So It think it's a
> >problem elsewhere but we shall see once mark looks at it.
> >
> >bkw
> >
> > > -----Original Message-----
> > > From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> > > admin at lists.digium.com] On Behalf Of programmer_ted
> > > Sent: Thursday, July 01, 2004 12:19 PM
> > > To: asterisk-users at lists.digium.com
> > > Subject: Re: [Asterisk-Users] Bugfix for CVS-HEAD-06/26/04-21:56:45
> > >
> > > 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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