Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10) - Solved.
On Wed, Jul 15, 2009 at 5:23 PM, Waynewa...@planetwayne.com 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10) - Solved.
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10) - Solved.
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
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=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On Sat, Jul 11, 2009 at 3:43 AM, Michiel van Baak mich...@vanbaak.infowrote: 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=getsearch=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) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On Sat, Jul 11, 2009 at 8:53 AM, Steve Totaro stot...@asteriskhelpdesk.comwrote: On Sat, Jul 11, 2009 at 3:43 AM, Michiel van Baak mich...@vanbaak.infowrote: 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=getsearch=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) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On 08:58, Sat 11 Jul 09, Steve Totaro wrote: On Sat, Jul 11, 2009 at 8:53 AM, Steve Totaro stot...@asteriskhelpdesk.comwrote: On Sat, Jul 11, 2009 at 3:43 AM, Michiel van Baak mich...@vanbaak.infowrote: 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=getsearch=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=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On 08:53, Sat 11 Jul 09, Steve Totaro wrote: On Sat, Jul 11, 2009 at 3:43 AM, Michiel van Baak mich...@vanbaak.infowrote: 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=getsearch=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=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
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. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On Sat, Jul 11, 2009 at 12:09 PM, Wayne wa...@planetwayne.com 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
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.gzAsterisk
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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Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
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 wa...@planetwayne.com 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)
On Fri, Jul 10, 2009 at 6:44 PM, Wayne wa...@planetwayne.com 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 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. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On Fri, Jul 10, 2009 at 7:33 PM, Wayne wa...@planetwayne.com 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) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
On Fri, Jul 10, 2009 at 4:33 PM, Wayne wa...@planetwayne.com 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Segmentation Faults Using Skinny (v1.6.0.10)
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.gzAsterisk
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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