[asterisk-dev] Help

Malcolm Davenport malcolmd at digium.com
Wed Jan 25 08:26:37 MST 2006 

Done,

Cheers.


On Wed, 2006-01-25 at 09:01 -0500, Chuck Crosby wrote:
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> Chuck Crosby
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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>
> > Content-Type: text/plain; charset="us-ascii"
> > 
> > An HTML attachment was scrubbed...
> > URL:
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> http://lists.digium.com/pipermail/asterisk-dev/attachments/20060124/2659e317/attachment-0001.htm
> > 
> > ------------------------------
> > 
> > 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 --
> > 
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> > 
> > 
> > End of asterisk-dev Digest, Vol 18, Issue 74
> > ********************************************
> > 
> 
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