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
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.