Dave Alan Caruana
2003-Jul-31 08:32 UTC
[Asterisk-Users] 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
Dave Alan Caruana
2003-Aug-04 16:55 UTC
[Asterisk-Users] SIP calls cause segmentation fault
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