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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 <markster@digium.com (markster@digium.com)>
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.