[asterisk-ss7] libss7 stability issues

Matthew Fredrickson creslin at digium.com
Mon Jun 9 12:14:43 CDT 2008

Matthew Fredrickson wrote:
> Andreas Kaufmann wrote:
>> Hi,
>> I have encountered stability problems when using libss7. I am using 
>> "sipp" to generate multiple calls over the link. After a few hundred 
>> test calls asterisk randomly stops working and the asterisk process does 
>> not exist any more in the unix process list.
>> A second problem I have encountered is that after generating a lot of 
>> concurrent calls,  some of the ss7 channels on the linkset are reported 
>> to be busy ("app_dial.c: Unable to create channel of type 'zap' (cause 
>> 34 - Circuit/channel congestion"), although this channels were not in 
>> use any more. I have to restart asterisk to get this channels free again.
>> I have tried the following configuration:
>> OS: OpenSuse 10.3
>> Zaptel: SVN-branch-1.4-r4315
>> libss7: SVN-trunk-r171
>> Asterisk: SVN-trunk-r118178M
>> DIGIUM TE220P 2xE1 Card
>> According to Matthews latest NEWS file (NEWS-05-30-2008) if have also tried:
>> libss7: SVN-trunk-r176
>> Zaptel: 1.4.11
>> Asterisk: 1.6.0
>> Can anyone recommend a stable version for libss7/zaptel/asterisk? Would 
>> it be better to use another distribution, e.g. Fedora?
> You might check to see if it is dumping core somewhere so we can get a 
> backtrace of why it failed.  If you start asterisk with the -g flag, it 
> will cause Asterisk to core dump when it crashes.  Then you can get a 
> backtrace of it's failure point with gdb.

One other thing I forgot to mention would be a debug trace up to the 
moment where it dies.  If you enable debugging (ss7 debug linkset x) and 
make sure your verbose output goes to some log file in logger.conf, that 
should leave a debug record I can use.

Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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