[Asterisk-Users] SIP calls cause Segmentation Fault

[Mark Spencer]

First off, there was a bug that caused this problem that has been fixed
about two days ago.  If your CVS is older than that, update.

Generally, be sure to run asterisk with the "-g" flag so it dumps core,
then do (in /usr/src/asterisk):

gdb ./asterisk /path/to/core.<pid>
(gdb) bt

and send me the result.

Mark


On Fri, 1 Aug 2003, Dave Alan Caruana wrote:

> Mark,
>
> the server has already been installed at a client and
> the only access to internet I have is from behind a NAT
> therefore I cannot give you access to log into the server.
>
> Also, I do not have an IRC client on the machine,
> and the closest windows machine is 4 floors away.
>
> What is the procedure to extract debug data I can
> send you please ?
>
> thanks
> Dave
>
> ----- Original Message -----
> From: "Mark Spencer" <markster at digium.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Thursday, July 31, 2003 8:11 PM
> Subject: Re: [Asterisk-Users] SIP calls cause Segmentation Fault
>
>
> > Yes, find me on #asterisk so I can login.  Be sure you're generating cores
> > and running on very latest CVS.
> >
> > Mark
> >
> > On Thu, 31 Jul 2003, Dave Alan Caruana wrote:
> >
> > > I have an asterisk installation at a client, it's quite simple.
> > > Basically it's an asterisk downloaded from CVS about
> > > a week ago, with 3 Zaptel FXO cards (the digium ones)
> > > and 10 Grandstream Budgettone SIP phones ...
> > >
> > > Every now and then, especially when a call is ringing
> > > and not picked up immediately, Asterisk quits with
> > > a segmentation fault error. IT seems quite inexplicable,
> > > my dialplan is a modification of the sample one that
> > > came with Asterisk, and I haven't touched that many
> > > other conf files actually.
> > >
> > > Any way I can get this debugged?
> > >
> > > cheers
> > > Dave
> > >
> > >
> > >
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> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
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> >
>
>
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