[Asterisk-Users] SIP calls cause Segmentation Fault

[Dave Alan Caruana]

Thanks Adam,
I will try it out.

cheers
Dave

----- Original Message ----- 
From: "Adam Donnison" <adam at saki.com.au>
To: <asterisk-users at lists.digium.com>
Sent: Friday, August 01, 2003 1:18 AM
Subject: Re: [Asterisk-Users] SIP calls cause Segmentation Fault


> 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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