[asterisk-users] 1.2.12.1 crashing

[Matt Florell]

If you downgrade, let us know if it fixes things for you.

It's strange that there were so many changes in the 1.2 SVN branch
after 1.2.7.1 that seem to be complete changes in how some things
operate(like the transcoding optimization mess for Asterisk 1.2.11 and
1.2.12 that was fixed in 1.2.12.1). I wish that such radical changes
would not be made in a release branch at the expense of reliabitily.

MATT---

On 10/12/06, Matt <mhoppes at gmail.com> wrote:
> Same thing here.. I can't pin it down.. other then I can make it
> happen by using ChanSpy.  This is an e-mail I got from our CSR Manager
> today..
>
> Technician is talking to the customer.
> The call drops.
> The technician still receives calls as if they were logged-in to the
> queue, but the callcenter website shows them not logged-in.
>
> I've heard this from several different technicians on different days.
> Something is up.
>
> hehe :)  I think I'll be downgrading to 1.2.7 Monday or tomorrow.
>
>
>
> On 10/12/06, Matt Florell <astmattf at gmail.com> wrote:
> > We have seen more random crashing on 1.2.12.1 as well as compared to 1.2.7.1
> >
> > It's not that bad, we only have one crash per week out of 6 servers
> > that are on 1.2.12.1, but it is much more than when we were running on
> > 1.2.7.1.
> >
> > It's never the same reason when we do a gdb backtrace on it though so
> > I have no idea what could be causing it.
> >
> >
> > MATT---
> >
> > On 10/12/06, Matt <mhoppes at gmail.com> wrote:
> > > Hi,
> > > We just upgraded from 1.2.7 to 1.2.12.1.   Everything is fine, except
> > > that asterisk seems to just crash at random.   Often I can make it
> > > crash by using the ChanSpy function (which we use to monitor agents).
> > > Sometimes it will just crash on its own.
> > >
> > > The reason we were initially running 1.2.7 was because of the
> > > stability it gave us (weeks without a restart).
> > >
> > > We upgraded to 1.2.12.1 because it seemed to resolve an issue we were
> > > having with some 800 # original being brought to us via sip
> > > (occassionally the calls would have 1 way audio when put on phone
> > > hold).  1.2.12.1 fixed this, however it is randomly and sporatically
> > > crashing.   Advise?
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