[Asterisk-Users] SIP calls cause Segmentation Fault

[Dave Alan Caruana]

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
> >
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>