[asterisk-dev] Help
Chuck Crosby
CCrosby at submeter.com
Wed Jan 25 07:01:57 MST 2006
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Chuck Crosby
IT Coordinator
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> Today's Topics:
>
> 1. IM/Presence with asterisk. asterisk NewBie (roswel ajf)
> 2. Re: IM/Presence with asterisk. asterisk NewBie
> (Matthew Fredrickson)
> 3. Passing AOC information across channels (Koopmann, Jan-Peter)
> 4. Problems submitting a patch (Eric Hartley)
> 5. Re: Problems submitting a patch (Tilghman Lesher)
> 6. svn 1.2 (Martin V?t)
> 7. No audio? Update your Asterisk (Olle E Johansson)
> 8. "Issue #6439 - the "timebomb" bug. Patch by Markster over
> GPRS" (Tamas)
> 9. Re: No audio? Update your Asterisk (Luigi Rizzo)
> 10. Re: "Issue #6439 - the "timebomb" bug. Patch by Markster over
> GPRS" (Luigi Rizzo)
> 11. Re: "Issue #6439 - the "timebomb" bug. Patch by Markster over
> GPRS" (Tamas)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 24 Jan 2006 19:25:52 +0000
> From: "roswel ajf" <roswel_ajf at hotmail.com>
> Subject: [asterisk-dev] IM/Presence with asterisk. asterisk NewBie
> To: asterisk-dev at lists.digium.com
> Message-ID: <BAY101-F22D1485382D0675C8943DEED130 at phx.gbl>
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>
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> ------------------------------
>
> Message: 2
> Date: Tue, 24 Jan 2006 14:40:18 -0600
> From: Matthew Fredrickson <creslin at digium.com>
> Subject: Re: [asterisk-dev] IM/Presence with asterisk. asterisk NewBie
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <897618d09de5bd47a1900c22c9130f80 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>
> On Jan 24, 2006, at 1:25 PM, roswel ajf wrote:
>
> > HI:
> >
> > I want to be an asterisk developer, want to get hands on with
> > IM/Presence work on asterisk. Currently, we already have asterisk
> > running on redhat linux FC3. Please advice.
> >
>
> There is a little bit of work going on in this area, you can find out
> more at issue #5501 on bugs.digium.com.
>
> Matthew Fredrickson
>
>
>
> ------------------------------
>
> Message: 3
> Date: Tue, 24 Jan 2006 22:08:51 +0100
> From: "Koopmann, Jan-Peter" <Jan-Peter.Koopmann at seceidos.de>
> Subject: [asterisk-dev] Passing AOC information across channels
> To: <asterisk-dev at lists.digium.com>
> Message-ID:
> <AEF86EFA5497434190F6D57E2666EA7A41DA00 at ERWIN.intern.seceidos.de>
> Content-Type: text/plain; charset="US-ASCII"
>
> Hi,
>
> I am following up the discussion in
>
> http://bugs.digium.com/view.php?id=6152
>
>
> How are we continueing on this issue? Obviously a lot of people want
> this feature and would like to see better AOC support in all sorts of
> channels, esp. in the ISDN channels. What is necessary to be able to
> pass the AOC information from channel to channel? Would it make sense to
> create a new AST_CONTROL frame for this?
>
> @oej: You mentioned this
>
> > Please note that we are moving from a fixed CDR
> > format to a dynamic format with the custom CDR
> > variables. There should be no need to discuss what
> > should be part of the CDR record as soon as the CDR
> > drivers actually support custom CDR fields, then it's
> > up to the sysadmin to configure what goes into each CDR.
> > The problem now is that the infrastructure is there, but
> > only one CDR driver actually supports it.
>
>
> I still do not understand how I get the AOC information from chan_zap in
> the dialplan so I could possibly do a
>
> Set(CDR(aoc)=whatever)
>
> to store it in a custom CDR value?
>
>
> Kind regards,
> JP
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 24 Jan 2006 19:08:50 -0500
> From: Eric Hartley <ehartley at thetangentgroup.com>
> Subject: [asterisk-dev] Problems submitting a patch
> To: asterisk-dev at lists.digium.com
> Message-ID: <43D6C192.3040501 at thetangentgroup.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> I recently submitted a patch I wrote to bugs.digium.com and did get some
> feedback that it should go in another file, which I corrected. Then I
> got more feedback that I have formatting errors. I've looked at what I
> wrote and it looks to me like the original code in the file. The
> guidelines I found online look like I wrote it correctly. Can anyone
> point me to updated guidelines or show me what's up with this so i can
> fix it.
>
> By the way, the bug in question is # 0006304.
>
> Thanks,
> Eric
>
>
>
> ------------------------------
>
> Message: 5
> Date: Tue, 24 Jan 2006 23:13:04 -0600
> From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
> Subject: Re: [asterisk-dev] Problems submitting a patch
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <200601242313.04857.tilghman at mail.jeffandtilghman.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On Tuesday 24 January 2006 18:08, Eric Hartley wrote:
> > I recently submitted a patch I wrote to bugs.digium.com and did get
> > some feedback that it should go in another file, which I corrected.
> > Then I got more feedback that I have formatting errors. I've
> > looked at what I wrote and it looks to me like the original code in
> > the file. The guidelines I found online look like I wrote it
> > correctly. Can anyone point me to updated guidelines or show me
> > what's up with this so i can fix it.
> >
> > By the way, the bug in question is # 0006304.
>
> The most frequent formatting error is that of lines indented by
> spaces, instead of by tabs. Towards the end of your patch, you have
> several lines that were indented with spaces.
>
> --
> Tilghman
>
>
> ------------------------------
>
> Message: 6
> Date: Wed, 25 Jan 2006 10:18:48 +0100
> From: Martin V?t <vit at lam.cz>
> Subject: [asterisk-dev] svn 1.2
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <43D74278.40707 at lam.cz>
> Content-Type: text/plain; charset=ISO-8859-2; format=flowed
>
> Hello, i've problem with calls with the lates svn 1.2 version.
> 1.2.0 is ok.
>
> how to debug this?
>
> Jan 25 09:07:23 DEBUG[3602] channel.c: Avoiding initial deadlock for
> 'IAX2/lam-nt-2'
> Jan 25 09:07:25 DEBUG[3614] channel.c: Dropping voice to exceptionally
> long queue on IAX2/lam-nt-2
> Jan 25 09:07:25 DEBUG[3614] channel.c: Dropping voice to exceptionally
> long queue on IAX2/lam-nt-2
> Jan 25 09:07:25 DEBUG[3614] channel.c: Dropping voice to exceptionally
> long queue on IAX2/lam-nt-2
> Jan 25 09:07:25 DEBUG[3614] channel.c: Dropping voice to exceptionally
> long queue on IAX2/lam-nt-2
> Jan 25 09:07:25 DEBUG[3614] channel.c: Dropping voice to exceptionally
> long queue on IAX2/lam-nt-2
> Jan 25 09:07:25 DEBUG[3614] channel.c: Dropping voice to exceptionally
> long queue on IAX2/lam-nt-2
>
>
> ------------------------------
>
> Message: 7
> Date: Wed, 25 Jan 2006 10:59:00 +0100
> From: Olle E Johansson <oej at edvina.net>
> Subject: [asterisk-dev] No audio? Update your Asterisk
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>, Asterisk Developers Mailing List
> <asterisk-dev at lists.digium.com>
> Message-ID: <43D74BE4.7050908 at edvina.net>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> This morning we discovered a serious bug that stopped all bridged audio
> in our Asterisk servers. Mark found the problem and soon fixed it.
>
> If you get this problem today, please update your Asterisk server. A fix
> has been commited to the subversion repository for 1.2 as well as trunk.
>
> A fixed 1.2.3 release will be published on ftp.digium.com as soon as we
> can find a release engineer (consider the time zone problem).
>
> A big thank you to everyone in the IRC channel that helped us locate
> this issue and to Mark that fixed it so quickly.
>
> /Olle
>
>
> ------------------------------
>
> Message: 8
> Date: Wed, 25 Jan 2006 11:33:14 +0100
> From: Tamas <jalsot at gmail.com>
> Subject: [asterisk-dev] "Issue #6439 - the "timebomb" bug. Patch by
> Markster over GPRS"
> To: asterisk-dev at lists.digium.com
> Message-ID: <43D753EA.4030402 at gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hello,
>
> anybody knows any explanation of this issue? When it came into the
> system what was the problem?
> Unfortunately some of our call centers 'died' yesterday evening and we
> have to make some explanation to the customer.
>
> The given patch solved the problem.
>
> Thanks in advance,
> Tamas
>
>
> ------------------------------
>
> Message: 9
> Date: Wed, 25 Jan 2006 03:13:19 -0800
> From: Luigi Rizzo <rizzo at icir.org>
> Subject: Re: [asterisk-dev] No audio? Update your Asterisk
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Cc: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <20060125031319.A5384 at xorpc.icir.org>
> Content-Type: text/plain; charset=us-ascii
>
> On Wed, Jan 25, 2006 at 10:59:00AM +0100, Olle E Johansson wrote:
> > This morning we discovered a serious bug that stopped all bridged audio
> > in our Asterisk servers. Mark found the problem and soon fixed it.
> >
> > If you get this problem today, please update your Asterisk server. A fix
> > has been commited to the subversion repository for 1.2 as well as trunk.
> >
> > A fixed 1.2.3 release will be published on ftp.digium.com as soon as we
> > can find a release engineer (consider the time zone problem).
> >
> > A big thank you to everyone in the IRC channel that helped us locate
> > this issue and to Mark that fixed it so quickly.
>
> a good way to identify these bugs would be to change
> ast_timediff_ms() to log values that are clearly out
> of range, and return a properly saturated value
> in those cases.
> Code below - i have no web access at the moment,
> but tried it and it spotted the bug before it was fixed.
> (adding code to print a backtrace would even
> point straight to the list of offending calls,
> i have code for that as well if someone is interested).
>
> cheers
> luigi
>
> AST_INLINE_API(
> int ast_tvdiff_ms(struct timeval end, struct timeval start),
> {
> /* the offset by 1,000,000 below is intentional...
> it avoids differences in the way that division
> is handled for positive and negative numbers, by ensuring
> that the divisor is always positive
> */
> int a = end.tv_sec - start.tv_sec;
> const int lim = 1<<21; /* max 21 bits, so the *1000 scaling will still fit
> in a 32-bit int */
> if (a > lim || a < -lim) {
> ast_log(LOG_WARNING, "tvdiff too large, saturating %d\n", a);
> a = (a > lim) ? lim : -lim;
> }
> return (a * 1000) +
> (((1000000 + end.tv_usec - start.tv_usec) / 1000) - 1000);
> }
> )
>
>
> ------------------------------
>
> Message: 10
> Date: Wed, 25 Jan 2006 04:06:38 -0800
> From: Luigi Rizzo <rizzo at icir.org>
> Subject: Re: [asterisk-dev] "Issue #6439 - the "timebomb" bug. Patch
> by Markster over GPRS"
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <20060125040638.A5969 at xorpc.icir.org>
> Content-Type: text/plain; charset=us-ascii
>
> On Wed, Jan 25, 2006 at 11:33:14AM +0100, Tamas wrote:
> > Hello,
> >
> > anybody knows any explanation of this issue? When it came into the
> > system what was the problem?
>
> the problem was an overflow in the computation of
> a difference between two struct timeval, and it occurred
> because one of the parameter was set to 0 instead of
> a time in the vicinity of the current date.
> Probably 64-bit machines were not affected.
> The bug would come out with a period of roughly every 2^22 seconds,
> which is not _that_ long. We were just lucky (or unlucky)
> that it did not come out earlier.
> On the other hand, for similar bugs, a missile (Ariane ?)
> came down a few years ago, so we are not the first nor
> the last to experience these things...
>
> cheers
> luigi
>
> > Unfortunately some of our call centers 'died' yesterday evening and we
> > have to make some explanation to the customer.
> >
> > The given patch solved the problem.
> >
> > Thanks in advance,
> > Tamas
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> ------------------------------
>
> Message: 11
> Date: Wed, 25 Jan 2006 13:11:09 +0100
> From: Tamas <jalsot at gmail.com>
> Subject: Re: [asterisk-dev] "Issue #6439 - the "timebomb" bug. Patch
> by Markster over GPRS"
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Message-ID: <43D76ADD.8040108 at gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hello,
>
> I found the issue, bug was not #6439, but #6349
>
> Sorry for bits....
>
> Regards,
> Tamas
>
> Tamas wrote:
> > Hello,
> >
> > anybody knows any explanation of this issue? When it came into the
> > system what was the problem?
> > Unfortunately some of our call centers 'died' yesterday evening and we
> > have to make some explanation to the customer.
> >
> > The given patch solved the problem.
> >
> > Thanks in advance,
> > Tamas
> >
>
>
>
> ------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> End of asterisk-dev Digest, Vol 18, Issue 74
> ********************************************
>
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