[asterisk-dev] 1.6.2 is broken
Inbox
venefax at gmail.com
Wed Jun 24 12:19:33 CDT 2009
There is one way to know. I will upgrade one server and wait until the
company that we use to monitor our system calls me saying "we cannot get a
response on port 5060"
If it fails again, is there any way I can invite you to take a look before I
go back and reinstall the version that works?
Philip
-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Mark Michelson
Sent: Wednesday, June 24, 2009 12:02 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] 1.6.2 is broken
Inbox wrote:
> I need everybody to know that since version SVN-branch-1.6.2-r197190 the
> 1.62. branch is broken. For more than a week, I have been trying every day
> to upgrade one of my 12 Asterisk boxes, which never go above 100
> simultaneous calls, and every time after a few hours Asterisk stops
talking
> to the network, so I have to go back and use version
> SVN-branch-1.6.2-r197190, which never stops working. I think we should
stop
> here and resolve this. I can replicate it, but then what? I don't know
what
> to do after I get it to fail. The Asterisk is working in the sense that I
> can log in and type "core restart now" It happens after dozen of thousands
> of calls, not immediately, but it always happens. I am probably the only
guy
> who can replicate this since I pass more than 1 million calls daily. I
have
> a load balancer made of OpenSips and 12 Asterisks to do the "real"
> processing.
> The version that works maybe higher than mine. I just had that one working
> fine in another box and went back to use it.
>
>
> Philip
>
"Philip"
I committed a change to chan_sip.c in rev 202340 of the 1.6.2 branch which
fixed
an infinite loop problem that could happen during reinvite glare situations.
It's impossible to tell from the meager amount of information you have
provided,
but it is certainly possible that the fix applied in that revision will fix
the
problem you are seeing.
Mark Michelson
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
More information about the asterisk-dev
mailing list