Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10) - Solved.

[Jonathan Thurman]

> Huh?
>
>
> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz
>
> is not the same as
>
>
> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.10.tar.gz?
>
> Their sha1 files are identical.
>
> sean
>
>
I believe he means that:


http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz

is not the same as

svn checkout http://svn.digium.com/svn/asterisk/branches/1.6.0



Which is true as there are lots of things that have been fixed in the
Subversion repo.

-Jonathan
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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10) - Solved.

[sean darcy]

On Wed, Jul 15, 2009 at 5:23 PM, Wayne wrote:
> Hi all,
> Just a quickie to say that this has been solved now - real simple -
> downloaded '*current*' rather than the versions from the home page of
> Astrisk.org. (didn't realise there was a 'current' version tbh.
>
> Anyways - I don't get Asterisk seg faulting now when hammering the
> speaker button on my cisco phones :)
>
> Interestingly - I've got another query - but will post another question
> when I've had chance to play more.
>
> Cheers
> Wayne.
>
> Wayne wrote:
>> Hi all,
>> I've just built a new installation of CentOS release 5.3 (Final) and
>> have installed both
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Huh?

http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0-current.tar.gz

is not the same as

http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.0.10.tar.gz ?

Their sha1 files are identical.

sean

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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10) - Solved.

[Wayne]

Hi all,
Just a quickie to say that this has been solved now - real simple - 
downloaded '*current*' rather than the versions from the home page of 
Astrisk.org. (didn't realise there was a 'current' version tbh.

Anyways - I don't get Asterisk seg faulting now when hammering the 
speaker button on my cisco phones :)

Interestingly - I've got another query - but will post another question 
when I've had chance to play more.

Cheers
Wayne.

Wayne wrote:
> Hi all,
> I've just built a new installation of CentOS release 5.3 (Final) and 
> have installed both 
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   


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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Jonathan Thurman]

On Sat, Jul 11, 2009 at 12:09 PM, Wayne  wrote:

>
> Thanks for all for the feedback with this - I'd like to help where I can -
> I'm building another 1.6 system for the office to try out the exchange tie
> in so if the general consensus is SIP is ok - then that's good for me too as
> I only have access to a SIP phone there.
>
> All my phones at home are Skinny so I was trying to kill two birds with one
> stone so to speak (get me on the latest version and play around with
> exchange). My  own 1.2 system is chugging along ok so far and there's no
> 'massive' need to move it over (ok other than servers sat in the lounge
> which the missus has a moan at every so often :-) ).
>
> One thing I am unsure of - how do I get the dumps / information you want in
> a suitable format.I'm still a novice with Linux / Asterisk but I'll gladly
> get anything to help out (just need some pointers in the right direction).
>

Take a look at file doc/backtrace.txt and doc/valgrind.txt.

What is your exact test scenario?  I have updated my test box based on the
latest SVN of 1.6.1 (I use 1.6.1.1 in production) and I have one Cisco 7940
configured for Skinny.  It seems to work just fine, no seg faults.  Have you
tried the latest SVN for 1.6.0?

You should take a look at this issue if you haven't already:
https://issues.asterisk.org/view.php?id=13777

-Jonathan
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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Wayne]

Michiel van Baak wrote:

  On 19:07, Fri 10 Jul 09, Steve Totaro wrote:
  
  
And convert those phones to SIP, forget chan_skinny.

  
  
Opinion time for me as well:
Dont. without bugreports chan_skinny will never be on par with chan_sip.
I know there are some segfaults with it here and there, but it's being
worked on.

  

Thanks for all for the feedback with this - I'd like to help where I
can - I'm building another 1.6 system for the office to try out the
exchange tie in so if the general consensus is SIP is ok - then that's
good for me too as I only have access to a SIP phone there.

All my phones at home are Skinny so I was trying to kill two birds with
one stone so to speak (get me on the latest version and play around
with exchange). My  own 1.2 system is chugging along ok so far and
there's no 'massive' need to move it over (ok other than servers sat in
the lounge which the missus has a moan at every so often :-) ).

One thing I am unsure of - how do I get the dumps / information you
want in a suitable format.I'm still a novice with Linux / Asterisk but
I'll gladly get anything to help out (just need some pointers in the
right direction).

Cheers
Wayne.





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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Michiel van Baak]

On 08:53, Sat 11 Jul 09, Steve Totaro wrote:
> On Sat, Jul 11, 2009 at 3:43 AM, Michiel van Baak wrote:
> 
> > On 19:07, Fri 10 Jul 09, Steve Totaro wrote:
> > >
> > > And convert those phones to SIP, forget chan_skinny.
> >
> > Opinion time for me as well:
> > Dont. without bugreports chan_skinny will never be on par with chan_sip.
> > I know there are some segfaults with it here and there, but it's being
> > worked on.
> >
> > --
> >
> > Michiel van Baak
> > mich...@vanbaak.eu
> > http://michiel.vanbaak.eu
> > GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
> >
> > "Why is it drug addicts and computer aficionados are both called users?"
> >
> >
> > My advice was to use chan_skinny, get a core dump and then convert to SIP.
> Result = chan_skinny gets fixed and OP has a usable phone system.

Indeed.

> 
> What good is a phone system if it core dumps everytime you make a call.

That's why we need bugreports with backtraces so we can fix it.

-- 

Michiel van Baak
mich...@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer aficionados are both called users?"


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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Michiel van Baak]

On 08:58, Sat 11 Jul 09, Steve Totaro wrote:
> On Sat, Jul 11, 2009 at 8:53 AM, Steve Totaro
> wrote:
> 
> > On Sat, Jul 11, 2009 at 3:43 AM, Michiel van Baak 
> > wrote:
> >
> >> On 19:07, Fri 10 Jul 09, Steve Totaro wrote:
> >> >
> >> > And convert those phones to SIP, forget chan_skinny.
> >>
> >> Opinion time for me as well:
> >> Dont. without bugreports chan_skinny will never be on par with chan_sip.
> >> I know there are some segfaults with it here and there, but it's being
> >> worked on.
> >>
> >> --
> >>
> >> Michiel van Baak
> >> mich...@vanbaak.eu
> >> http://michiel.vanbaak.eu
> >> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
> >>
> >> "Why is it drug addicts and computer aficionados are both called users?"
> >>
> >>
> >> My advice was to use chan_skinny, get a core dump and then convert to
> > SIP.  Result = chan_skinny gets fixed and OP has a usable phone system.
> >
> > What good is a phone system if it core dumps everytime you make a call.
> >
> >
> Some really off-topic opinion.
> 
> Stop spending time on chan_skinny and work create chan_nbx and chan_megaco.
> 
> Then you could connect Asterisk to 3Com and NEC products without messing
> with T1s or analog ports.

Contact Digium to find out where to send the hardware and wireshark
protocol dumps so we can get to it.

-- 

Michiel van Baak
mich...@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer aficionados are both called users?"


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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Steve Totaro]

On Sat, Jul 11, 2009 at 8:53 AM, Steve Totaro
wrote:

> On Sat, Jul 11, 2009 at 3:43 AM, Michiel van Baak wrote:
>
>> On 19:07, Fri 10 Jul 09, Steve Totaro wrote:
>> >
>> > And convert those phones to SIP, forget chan_skinny.
>>
>> Opinion time for me as well:
>> Dont. without bugreports chan_skinny will never be on par with chan_sip.
>> I know there are some segfaults with it here and there, but it's being
>> worked on.
>>
>> --
>>
>> Michiel van Baak
>> mich...@vanbaak.eu
>> http://michiel.vanbaak.eu
>> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
>>
>> "Why is it drug addicts and computer aficionados are both called users?"
>>
>>
>> My advice was to use chan_skinny, get a core dump and then convert to
> SIP.  Result = chan_skinny gets fixed and OP has a usable phone system.
>
> What good is a phone system if it core dumps everytime you make a call.
>
>
Some really off-topic opinion.

Stop spending time on chan_skinny and work create chan_nbx and chan_megaco.

Then you could connect Asterisk to 3Com and NEC products without messing
with T1s or analog ports.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Steve Totaro]

On Sat, Jul 11, 2009 at 3:43 AM, Michiel van Baak wrote:

> On 19:07, Fri 10 Jul 09, Steve Totaro wrote:
> >
> > And convert those phones to SIP, forget chan_skinny.
>
> Opinion time for me as well:
> Dont. without bugreports chan_skinny will never be on par with chan_sip.
> I know there are some segfaults with it here and there, but it's being
> worked on.
>
> --
>
> Michiel van Baak
> mich...@vanbaak.eu
> http://michiel.vanbaak.eu
> GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD
>
> "Why is it drug addicts and computer aficionados are both called users?"
>
>
> My advice was to use chan_skinny, get a core dump and then convert to SIP.
Result = chan_skinny gets fixed and OP has a usable phone system.

What good is a phone system if it core dumps everytime you make a call.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Michiel van Baak]

On 19:07, Fri 10 Jul 09, Steve Totaro wrote:
> 
> And convert those phones to SIP, forget chan_skinny.

Opinion time for me as well:
Dont. without bugreports chan_skinny will never be on par with chan_sip.
I know there are some segfaults with it here and there, but it's being
worked on.

-- 

Michiel van Baak
mich...@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer aficionados are both called users?"


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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Jonathan Thurman]

On Fri, Jul 10, 2009 at 4:33 PM, Wayne  wrote:

>
>
> Hi Steve,
> Thanks for the pointers. I must admit - I was leaning towards 1.6 as
> this apparently has support for SIP over TCP (?). My end goal with this
> was to try and get Asterisk talking to Exchange 2007 servers unified
> messaging.
>

While I haven't used the SIP over TCP in production (yet), I find that the
1.6.1 series is stable for our environment.  I don't know about using
Exchange, as we are staying as far from unified messaging as possible (for
political reasons of course...)  I wouldn't install 1.0, so why go back to
1.2 or 1.4.  Just more to learn and relearn.  The important thing is to have
a test environment to get all of the show stopping buts out.


>
> As for chan_skinny - I'm currently using this on an existing 1.2 server
> although from what I've picked up from previous posts (going back a
> while) the inbuilt version is now quite stable and possibly better than
> the older 'chan_skinny' (which I think the development has stopped for
> now?). This is why I opted to use it for the new 1.6 server.
>

What is the main reason for staying with skinny on these phones?  I have
quite a few 7940/7960 converted to SIP that work great.

Next week I will try and duplicate this behavior on my test system with
skinny, but you should get a bug report filed with the core and important
configurations.

-Jonathan
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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Steve Totaro]

On Fri, Jul 10, 2009 at 7:33 PM, Wayne  wrote:

>
>
> Steve Totaro wrote:
> >
> > If you are set on "beta" then read no further then the next line.
> >
> > File a bug report with a core dump.
> >
> > OK opinion time.
> >
> > Your server is more than adequate.
> >
> > For my tastes, you are beyond bleeding edge on the Asterisk front.
> >
> > Simply my opinion but if this is going to be a "real" "production"
> > "server" or something you want to use reliably then I would suggest.
> >
> > 1.4.Latest Zaptel
> > 1.4.19 Asterisk (if infact that is the last version that had
> > chan_zap and not DAHDI)
> > 1.4.Current LibPRI
> >
> >
> > And convert those phones to SIP, forget chan_skinny.
> >
> Hi Steve,
> Thanks for the pointers. I must admit - I was leaning towards 1.6 as
> this apparently has support for SIP over TCP (?). My end goal with this
> was to try and get Asterisk talking to Exchange 2007 servers unified
> messaging.
>
> As for chan_skinny - I'm currently using this on an existing 1.2 server
> although from what I've picked up from previous posts (going back a
> while) the inbuilt version is now quite stable and possibly better than
> the older 'chan_skinny' (which I think the development has stopped for
> now?). This is why I opted to use it for the new 1.6 server.
>
> Still open to an suggestions though :)
>
> Thanks
> Wayne.
>

Second line.

File a bug report.

There are not nearly as many people on 1.6 as 1.2 or 1.4.

I wish I had stats, but many people from the old skool never wanted to go
past 1.2, myself included.  1.4 has proven itself stable in my book with
zaptel, I don't mess with DAHDI.  I still do 1.2 installs for core systems.
1.4 was for app_rpt with the URI (a radio repeater controller that is USB
based).

I think the people using 1.4 have been early adoptors or just started using
asterisk at that version.  They don't even know about Asterisk .3 that only
supported Adtran equipment.

1.6.X is super beta and you are in the new frontier.

Out of the people using 1.6.x, you may be the only person to try
chan_skinny.

You may be the first to find the 1.6.x chan_skinny bug, so start asterisk so
it does a core dump and make a skinny call, then file a bug report.

Hopefully things on bugtracker have changed and it gets attention and not
just closed or ignored.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Wayne]

Steve Totaro wrote:
>
> If you are set on "beta" then read no further then the next line.
>
> File a bug report with a core dump.
>
> OK opinion time.
>
> Your server is more than adequate.
>
> For my tastes, you are beyond bleeding edge on the Asterisk front.
>
> Simply my opinion but if this is going to be a "real" "production"
> "server" or something you want to use reliably then I would suggest.
>
> 1.4.Latest Zaptel
> 1.4.19 Asterisk (if infact that is the last version that had
> chan_zap and not DAHDI)
> 1.4.Current LibPRI
>
>
> And convert those phones to SIP, forget chan_skinny.
>
Hi Steve,
Thanks for the pointers. I must admit - I was leaning towards 1.6 as 
this apparently has support for SIP over TCP (?). My end goal with this 
was to try and get Asterisk talking to Exchange 2007 servers unified 
messaging.

As for chan_skinny - I'm currently using this on an existing 1.2 server 
although from what I've picked up from previous posts (going back a 
while) the inbuilt version is now quite stable and possibly better than 
the older 'chan_skinny' (which I think the development has stopped for 
now?). This is why I opted to use it for the new 1.6 server.

Still open to an suggestions though :)

Thanks
Wayne.




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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Steve Totaro]

On Fri, Jul 10, 2009 at 6:44 PM, Wayne  wrote:

> Sorry to bump my own message - but had a mail server problem so don't
> know if I missed any replys :(
> Ta
> Wayne.
>
>
>
> Wayne wrote:
> > Hi all,
> > I've just built a new installation of CentOS release 5.3 (Final) and
> > have installed both
> > <
> http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz
> >Asterisk
> > 1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe
> > trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing
> > complex - Pentium Dual core 2ghz - 1gb ram - 70gb sata hd).
> >
> > The setup at this point is real simple with one Cisco 7960 phone
> > registering with Asterisk using Skinny.
> >
> > I'm finding that simple things as pressing any of the buttons on the
> > phone is enough to cause Asterisk to randomly restart from a
> > segmentation fault.
> >
> > I've tried this with 1.6.1.1 and, after recompiling and replacing,
> 1.6.0.10.
> >
> > I followed
> >
> http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
> > as a basis for installation leaving out things I didnt want to set up
> > (odbc / web admin ).
> >
> > The only thing that didn't seem to go too well was the setup Dahdi
> > (dahdi-linux-2.2.0.1). Although I can do a 'make' and 'make install',
> > 'make config' didnt work and there are no etc/dahdi/ directory to change
> > any config files (as suggested by the guide). This may not be related
> > but just in case I thought I would mention it.
> >
> >
> > This is from the console after pressing the 'speaker' button a couple of
> > times.
> >
> >  /usr/sbin/safe_asterisk: line 146: 21513 Segmentation fault  (core
> > dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
> > ${ASTARGS} >&/dev/${TTY} < /dev/${TTY}
> > Asterisk ended with exit status 139
> > Asterisk exited on signal EXITSTATUS-128.
> > Automatically restarting Asterisk.
> >
> >
> > If I don't use the phone, Asterisk will stay running.
> > I can dial the 1000 test extension along with the 500 inter-asterisk
> > test, these seem to work as expected as long as I dial the number and
> > hit 'dial' on the phone rather than selecting the line and trying to
> > dial each digit in turn. If I try that then at some random point (but
> > not always) Asterisk will fault.
> >
> > The firmware version on the phone is 7.2 to which I've had this phone
> > and several others running off a 1.2 setup for years (using
> > chan_skinny?) but thought it time to update Asterisk.
> >
> >
> > Anyone have any pointers please on what to check next?
> >
> > Thanks,
> > Wayne
> >
> >
>

If you are set on "beta" then read no further then the next line.

File a bug report with a core dump.

OK opinion time.

Your server is more than adequate.

For my tastes, you are beyond bleeding edge on the Asterisk front.

Simply my opinion but if this is going to be a "real" "production" "server"
or something you want to use reliably then I would suggest.

1.4.Latest Zaptel
1.4.19 Asterisk (if infact that is the last version that had chan_zap and
not DAHDI)
1.4.Current LibPRI
-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Steve Totaro]

On Fri, Jul 10, 2009 at 7:06 PM, Steve Totaro <
stot...@totarotechnologies.com> wrote:

>
>
> On Fri, Jul 10, 2009 at 6:44 PM, Wayne  wrote:
>
>> Sorry to bump my own message - but had a mail server problem so don't
>> know if I missed any replys :(
>> Ta
>> Wayne.
>>
>>
>>
>> Wayne wrote:
>> > Hi all,
>> > I've just built a new installation of CentOS release 5.3 (Final) and
>> > have installed both
>> > <
>> http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz
>> >Asterisk
>> > 1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe
>> > trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing
>> > complex - Pentium Dual core 2ghz - 1gb ram - 70gb sata hd).
>> >
>> > The setup at this point is real simple with one Cisco 7960 phone
>> > registering with Asterisk using Skinny.
>> >
>> > I'm finding that simple things as pressing any of the buttons on the
>> > phone is enough to cause Asterisk to randomly restart from a
>> > segmentation fault.
>> >
>> > I've tried this with 1.6.1.1 and, after recompiling and replacing,
>> 1.6.0.10.
>> >
>> > I followed
>> >
>> http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation
>> > as a basis for installation leaving out things I didnt want to set up
>> > (odbc / web admin ).
>> >
>> > The only thing that didn't seem to go too well was the setup Dahdi
>> > (dahdi-linux-2.2.0.1). Although I can do a 'make' and 'make install',
>> > 'make config' didnt work and there are no etc/dahdi/ directory to change
>> > any config files (as suggested by the guide). This may not be related
>> > but just in case I thought I would mention it.
>> >
>> >
>> > This is from the console after pressing the 'speaker' button a couple of
>> > times.
>> >
>> >  /usr/sbin/safe_asterisk: line 146: 21513 Segmentation fault  (core
>> > dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
>> > ${ASTARGS} >&/dev/${TTY} < /dev/${TTY}
>> > Asterisk ended with exit status 139
>> > Asterisk exited on signal EXITSTATUS-128.
>> > Automatically restarting Asterisk.
>> >
>> >
>> > If I don't use the phone, Asterisk will stay running.
>> > I can dial the 1000 test extension along with the 500 inter-asterisk
>> > test, these seem to work as expected as long as I dial the number and
>> > hit 'dial' on the phone rather than selecting the line and trying to
>> > dial each digit in turn. If I try that then at some random point (but
>> > not always) Asterisk will fault.
>> >
>> > The firmware version on the phone is 7.2 to which I've had this phone
>> > and several others running off a 1.2 setup for years (using
>> > chan_skinny?) but thought it time to update Asterisk.
>> >
>> >
>> > Anyone have any pointers please on what to check next?
>> >
>> > Thanks,
>> > Wayne
>> >
>> >
>>
>
> If you are set on "beta" then read no further then the next line.
>
> File a bug report with a core dump.
>
> OK opinion time.
>
> Your server is more than adequate.
>
> For my tastes, you are beyond bleeding edge on the Asterisk front.
>
> Simply my opinion but if this is going to be a "real" "production" "server"
> or something you want to use reliably then I would suggest.
>
> 1.4.Latest Zaptel
> 1.4.19 Asterisk (if infact that is the last version that had chan_zap and
> not DAHDI)
> 1.4.Current LibPRI
>

And convert those phones to SIP, forget chan_skinny.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)

[Wayne]

Sorry to bump my own message - but had a mail server problem so don't 
know if I missed any replys :(
Ta
Wayne.



Wayne wrote:
> Hi all,
> I've just built a new installation of CentOS release 5.3 (Final) and 
> have installed both 
> Asterisk
>  
> 1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe 
> trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing 
> complex - Pentium Dual core 2ghz - 1gb ram - 70gb sata hd).
>
> The setup at this point is real simple with one Cisco 7960 phone 
> registering with Asterisk using Skinny.
>
> I'm finding that simple things as pressing any of the buttons on the 
> phone is enough to cause Asterisk to randomly restart from a 
> segmentation fault.
>
> I've tried this with 1.6.1.1 and, after recompiling and replacing, 1.6.0.10.
>
> I followed 
> http://www.voip-info.org/wiki/view/CentOS+5.2+and+Asterisk+1.6.x+installation 
> as a basis for installation leaving out things I didnt want to set up 
> (odbc / web admin ).
>
> The only thing that didn't seem to go too well was the setup Dahdi 
> (dahdi-linux-2.2.0.1). Although I can do a 'make' and 'make install', 
> 'make config' didnt work and there are no etc/dahdi/ directory to change 
> any config files (as suggested by the guide). This may not be related 
> but just in case I thought I would mention it.
>
>
> This is from the console after pressing the 'speaker' button a couple of 
> times.
>
>  /usr/sbin/safe_asterisk: line 146: 21513 Segmentation fault  (core 
> dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
> ${ASTARGS} >&/dev/${TTY} < /dev/${TTY}
> Asterisk ended with exit status 139
> Asterisk exited on signal EXITSTATUS-128.
> Automatically restarting Asterisk.
>
>
> If I don't use the phone, Asterisk will stay running.
> I can dial the 1000 test extension along with the 500 inter-asterisk 
> test, these seem to work as expected as long as I dial the number and 
> hit 'dial' on the phone rather than selecting the line and trying to 
> dial each digit in turn. If I try that then at some random point (but 
> not always) Asterisk will fault.
>
> The firmware version on the phone is 7.2 to which I've had this phone 
> and several others running off a 1.2 setup for years (using 
> chan_skinny?) but thought it time to update Asterisk.
>
>
> Anyone have any pointers please on what to check next?
>
> Thanks,
> Wayne
>
>
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