I have an Asterisk 13 that only processes app Transfer with PJSIP, to the tune of 60 per second. No voice calls. After like 2 hours, I can no longer get into Asterisk. This command, asterisk -r, fails, and also "asterisk -rx core show channels", etc. I am returned to the bash prompt. I checked the handles and
lsof | grep asterisk |wc -l 7098126 I think there is a kind of handle leak here. Nothing else runs in the box If there is a way to find out what happens, let me know. The dialplan is confidential, for it belongs to my customer,but I can give you access to the box. In short , the app receives a call, checks the number against a database and calls app_transfer. That is it. This is what I see when the command fails: asterisk -r Asterisk SVN-branch-13-r431555M, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer <[email protected]> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= [root@centos7 /]# this command shows the issue, thousands of lines lsof | grep asterisk asterisk 4077 root *450w FIFO 0,8 0t0 110430221 pipe asterisk 4077 root *451r FIFO 0,8 0t0 110429239 pipe asterisk 4077 root *452w FIFO 0,8 0t0 110429239 pipe asterisk 4077 root *453r FIFO 0,8 0t0 110417598 pipe asterisk 4077 root *454w FIFO 0,8 0t0 110417598 pipe asterisk 4077 root *455r FIFO 0,8 0t0 110426507 pipe asterisk 4077 root *456w FIFO 0,8 0t0 110426507 pipe^ It looks like https://issues.asterisk.org/jira/browse/ASTERISK-823 but in fact I am using PJSIP. It is definitely PJSIP, for I replaced the dialplan with plain SIP, and there is no issue, ceteris paribus. Note: I am not a developer and have no idea how to troubleshoot this.
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