<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:12pt"><div>Hi,<br><br>I have already setup to rotate logs hourly & Debug level is 3.<br>Is there any other possibility of crash?<br><br>Thanks in advance!!<br><br><br>--<br>Regards,<br>Voipexpert<br><br></div><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><br><div style="font-family: arial,helvetica,sans-serif; font-size: 13px;"><font face="Tahoma" size="2"><hr size="1"><b><span style="font-weight: bold;">From:</span></b> Giorgio Incantalupo <gincantalupo@fgasoftware.com><br><b><span style="font-weight: bold;">To:</span></b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b><span style="font-weight: bold;">Sent:</span></b> Fri, November 20, 2009 10:06:06 PM<br><b><span style="font-weight:
bold;">Subject:</span></b> Re: [asterisk-users] Asterisk crashes : Failed to start PBX<br></font><br>Hi Neo,<br><br>have you checked your log files? It sometimes happened to me that <br>Asterisk crashed without a reason. I discovered my logrotate didn't make <br>its dirty work so I had huge log files. I lowered Asterisk log level and <br>forced logrotate to work and now I have no more crashes.<br><br>Hope it may help. :)<br><br>Giorgio.<br><br>Neo Anderson wrote:<br>> Hello,<br>><br>> I am using Asterisk 1.4.24.1 version in production.<br>> OS is Centos 5.3 64 bit & RAM is 8 GB.<br>> I am facing crash in asterisk approx each 12 hour.<br>> When it crashes I see below lines in asterisk logs.<br>> [Nov 18 06:47:23] WARNING[8730] pbx.c: Failed to create new channel thread<br>> [Nov 18 06:47:23] WARNING[8730] chan_sip.c: Failed to start PBX :(<br>> I debugged asterisk source code in details & I found that it
happens <br>> because it can not allocate memory to create thread.<br>><br>> Another thing is, when I check coredump using gdb, it's not showing <br>> any debug symbols.<br>><br>> Would you please let me know how to prevent or resolve this?<br>><br>> Thanks in advance!!<br>><br>> --<br>> Regards,<br>> voipexpert<br>><br>><br>> ------------------------------------------------------------------------<br>><br>> _______________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>><br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br><br>_______________________________________________<br>--
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