asterisk users - Jan 2006 - No audio? Update your Asterisk

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

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);
}
)

hello,

Could you give us the path to the patch quickly ?

Harry Gaillac
--- Olle E Johansson <oej@edvina.net> a ?crit :
> 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
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com
> --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 


	

	
		
___________________________________________________________________________ 
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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.
>
What versions of Asterisk does this affect?

Darren

hello,

Could provide us how to fix this serious bug my server
is out of order please to post how to solve quickly
this problem .

Harry

--- Olle E Johansson <oej@edvina.net> a ?crit :
> 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
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com
> --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 


	

	
		
___________________________________________________________________________ 
Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs
exceptionnels pour appeler la France et l'international.
T?l?chargez sur http://fr.messenger.yahoo.com

Was this in 1.2.2 or was it introduced in a subsequent change?

--On Wednesday, January 25, 2006 10:59 AM +0100 Olle E Johansson 
<oej@edvina.net> 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.
>
> /Olle
> _______________________________________________
> --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

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);
}
)
_______________________________________________
--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

> 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.
> >
> What versions of Asterisk does this affect?
> 
> Darren
> _______________________________________________
Hopefully not "stable" ;-)

All versions released in the last 2 weeks i think.

take the newest versions from svn or the ftp. (1.2.3 is released).

Cheers,

Zoa


Darren Ellis wrote:
> 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.
>>
> What versions of Asterisk does this affect?
>
> Darren
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

unsubscribe 

-----Original Message-----
From: asterisk-dev-bounces@lists.digium.com
[mailto:asterisk-dev-bounces@lists.digium.com] On Behalf Of Ed Greenberg
Sent: Wednesday, January 25, 2006 6:30 AM
To: Asterisk Developers Mailing List; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-dev] No audio? Update your Asterisk

Was this in 1.2.2 or was it introduced in a subsequent change?

--On Wednesday, January 25, 2006 10:59 AM +0100 Olle E Johansson
<oej@edvina.net> 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.
>
> /Olle
> _______________________________________________
> --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



_______________________________________________
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asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Zoa wrote:
> All versions released in the last 2 weeks i think.
>
> take the newest versions from svn or the ftp. (1.2.3 is released).
>
Thanks Zoa,

/emergency mode off

FYI, just upgraded from 1.2.2 to 1.2.3 and audio problems in sip
channels gone way.

Thanks a lot,

--
Antonio Jos? dos Santos Brand?o
Virgos Tecnologia da Informa??o
www.virgos.com.br - S?o Carlos,SP


On 1/25/06, Olle E Johansson <oej@edvina.net>
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.
>
> /Olle
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Yeah, got two whacked 1.2.2 servers here. Updated to 1.2.3 from ftp source,
and a-ok.

HOWEVER, echo cancellation seems to be non-existant in our TDM card?!? I
recompiled/installed zaptel 1.2.2, no effect. Even tried different
cancellers. Anyone else experience this after upgrading to 1.2.3?


Sincerely,

Brent A. Torrenga
brent.torrenga@torrenga.com

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

219.836.8918x325 Voice
219.836.1138 Facsimile
www.torrenga.com

> For what it's worth, I've been messing around with my install all
> night and haven't had a single issue.  Asterisk@Home 2.2, Asterisk
> version 1.2.1.  Even set the date ahead, still no problems.  Could be
> a fluke, I'm interested if anyone else is using 1.2.1 and has these
> issues, but for now I'm sticking with what I have.
Just to expand on this a little, I installed 1.2.2 as soon as it was
released, and ran it on a production server quite happily until this
morning, when I suddenly lost audio on certain calls.  Restarts did not fix
the problem, which really confused me.  I saw the upgrade to 1.2.3 and
installed it right away.  Everything was fixed with 1.2.3.

On another production server, I've been running 1.2.1 for a few weeks with
no problems.

- Noah

To check if it's the same problem, set your system clock back 2 weeks.
If it gets better, then the upgrade didn't take. If it doesn't get
better, it's something else.

--Rob
> -----Original Message-----
> Very greatful to find this I have upgraded to 1.2.3 but
> still have no sip-sip audio!