asterisk users - Jul 2003 - SIP calls cause Segmentation Fault

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

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@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

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@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Adam Donnison                                  email: adam@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

does anyone of the programmers know if this has been
fixed in a more recent CVS version? should I redownload
and recompile?

cheers
Dave

----- Original Message ----- 
From: "Adam Donnison" <adam@saki.com.au>
To: <asterisk-users@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@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> -- 
> Adam Donnison                                  email: adam@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
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

Yes, latest CVS fixes this.

Adam

Dave Alan Caruana wrote:
> does anyone of the programmers know if this has been
> fixed in a more recent CVS version? should I redownload
> and recompile?
> 
> cheers
> Dave
> 
-- 
Adam Donnison                                  email: adam@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