[Asterisk-Users] SIP calls cause Segmentation Fault

[Mark Spencer]

This *should* already be fixed.

Mark

On Fri, 1 Aug 2003, Adam Donnison wrote:

> I actually found this same thing, and traced it down to
> app_dial.c line 190.  It doesn't explicitly check for
> a valid chan before trying to use it and it segfaults when
> it does a strlen on a chan entity.  I simply put a check
> in that winner was non-zero before comparing it to o->chan:
>
> if (winner && winner == o->chan)
>
> Adam
>
> 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
>
>
> --
> Adam Donnison                                  email: adam at saki.com.au
> Saki Computer Services Pty. Ltd.
> 93 Kallista-Emerald Road                        phone: +61 3 9752 1512
> THE PATCH  VIC 3792    AUSTRALIA                fax:   +61 3 9752 1098
>
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