On Thu, May 1, 2014 at 11:08 AM, Administrator TOOTAI <ad...@tootai.net> wrote: <snip> > As explained in one on my previous message, it's a bug, easily reproducible: > take a queues.conf (or sip.conf or iax.conf or voicemail.conf or ...) like > this (what is important is the #include): <snip> > NOTICE[3346]: app_queue.c:6811 reload_queue_rules: queuerules.conf has not > changed since it was last loaded. Not taking any action. > > despite the fact that modification was done in a .conf file. I took this > example as with module reload app_queue the above message appears. For sip, > iax, voicemail, aso there is no message, just "SIP reload" or ... > > To make asterisk take the modification in account, you have to open > /etc/asterisk/[sip|iax|voicemail|queue|..].conf and save it without making > any change. After this the command will be execute. It you run it a second > time in a raw, you will see that the false behavior appears again till you > again open/save the original file.
Hi! I tried to reproduce using your description here and could not reproduce the issue. I tried with both sip.conf and queues.conf. Making a change in an included .conf file, but NOT the parent .conf file and then reloading that module from the CLI results in: centosclean*CLI> module reload app_queue.so -- Reloading module 'app_queue.so' (True Call Queueing) [May 6 17:51:39] NOTICE[16211]: app_queue.c:7765 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action. == Parsing '/etc/asterisk/queues.conf': Found == Parsing '/etc/asterisk/queue_include_1.conf': Found == Parsing '/tmp/queue_include_2.conf': Found I get the same behavior with sip.conf, it appears to work fine, whether I'm making only changes in the parent .conf or the included children. I even tried with two different included files in each sip.conf and queues.conf, one in /tmp and one in /etc/asterisk. Same working behavior. I used SVN-branch-11-r413305, so you might want to test there. However I'm still confused as to how you are seeing the behavior you are seeing. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users