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Several Troubles with CPU load and Asterisk.

Added by Leopoldo Martinez 3 months ago

Hello everyone, I am going to tell you all about my VPS to find some operating problems.

I made a complete installation from scratch on CentOS7 with the installation guide offered in this forum.
My goautodial system is updated with the latest version of github, the outgoing calls work perfectly.
But the consumption of the cpu is high (1800% on dashboard view and 300% on my VPS provider GUI), it is worth mentioning that my VPS has 16Gb of Ram, 6 Cores and 320Gb of storage.
This problem happens having approximately 36 agents connected and calling at the same time from the same VoIP account and 3 different campaigns. My marketing campaigns are to Spain, my voip provider transmits in G711alaw / G729 / codecs and everything works perfect up to this point.
This is where the real problems begin, eventually asterisk crashes for no apparent reason, as much as I look for the coredumps files, I can't get them, I can't see exactly what the reason for the crash is. On the other hand, they have provided me in this same forum a command that apparently starts asterisk, in the process of that script it opens a screen to leave asterisk running in the background, the problem with that command is that it creates a new asterisk window and asterisk does not start, first you have to delete it and then run the command. Even so, asterisk does not start correctly, in some service / library / dependency it is missing to start something and all the sounds of ast_openfull fail and there is no way to recover it if it is not through a reboot.

this is the command provided

/usr/share/astguiclient/start_asterisk_boot.pl

How can I make asterisk start automatically after a crash? But 100% optimal, we found an unusual way and it works 80% of the time but it starts asterisk with the error of the welcome audio for the agent and the call notifications.

apart from that, asterisk also causes a problem when you have incoming calls, if there are outgoing calls and a call comes in at the same time, asterisk crashes, I can only hear incoming calls if they are made from a VoIP, SIP or trunk, if it is from a cell phone, or some other from the PSTN network, the call hangs up the second and the one who makes the call only hears 3 beeps and then hangs up.

Right now I see that someone said that changing app_konference to meetme, could solve it, but I can't find a way to change from konference to meetme either.

apparently there are problems with the codecs for europe or something like that?

how can i change konference for meetme?


Replies (2)

RE: Several Troubles with CPU load and Asterisk. - Added by Jackie Alfonso 3 months ago

Hi,

We do suggest to upgrade your server to a multi server. please follow the these hardware specs for your 36 agents.

Up to 2 servers
Eight core processor
8 GB RAM
2 X 512 GB SSD in RAID1

To switch to app_meetme please try this one, in /etc/asterisk/extensions.conf just uncomment the entries that has "meetme" and in /etc/asterisk/extensions-goautodial.conf comment the entries that has "konference".

RE: Several Troubles with CPU load and Asterisk. - Added by Leopoldo Martinez 3 months ago

Jackie Alfonso wrote in RE: Several Troubles with CPU load and Asterisk.:

Hi,

We do suggest to upgrade your server to a multi server. please follow the these hardware specs for your 36 agents.

Up to 2 servers
Eight core processor
8 GB RAM
2 X 512 GB SSD in RAID1

To switch to app_meetme please try this one, in /etc/asterisk/extensions.conf just uncomment the entries that has "meetme" and in /etc/asterisk/extensions-goautodial.conf comment the entries that has "konference".

-- Called 99998248837597@default
-- Executing [99998248837597@default:1] Dial("Local/99998248837597@default-00000000;2", "SIP/8248837597@kamailio,,tTo") in new stack
Using SIP RTP CoS mark 5
-- Called SIP/8248837597@kamailio
-- SIP/kamailio-00000000 is ringing
-- Local/99998248837597@default-00000000;1 is ringing
-- SIP/kamailio-00000000 answered Local/99998248837597@default-00000000;2
-- Channel SIP/kamailio-00000000 joined 'simple_bridge' basic-bridge <e0d1ffa9-1765-4b7e-aadc-346403bf0b0f>
-- Local/99998248837597@default-00000000;1 answered
[Jun 25 17:29:36] WARNING[1906][C-00000001]: pbx.c:2864 pbx_extension_helper: No application 'Meetme' for extension (default, 8600051, 1)
Spawn extension (default, 8600051, 1) exited non-zero on 'Local/99998248837597@default-00000000;1'
[Jun 25 17:29:36] WARNING1906[C-00000001]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
-- Executing [h@default:1] AGI") in new stack
Manager 'sendcron' logged off from 127.0.0.1
-- Channel Local/99998248837597@default-00000000;2 joined 'simple_bridge' basic-bridge <e0d1ffa9-1765-4b7e-aadc-346403bf0b0f>
-- <Local/99998248837597@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) completed, returning 0
-- Channel Local/99998248837597@default-00000000;2 left 'simple_bridge' basic-bridge <e0d1ffa9-1765-4b7e-aadc-346403bf0b0f>
Spawn extension (default, 99998248837597, 1) exited non-zero on 'Local/99998248837597@default-00000000;2'
-- Executing [h@default:1] AGI") in new stack
-- Channel SIP/kamailio-00000000 left 'simple_bridge' basic-bridge <e0d1ffa9-1765-4b7e-aadc-346403bf0b0f>
-- <Local/99998248837597@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----0-----0-----SIP 200 OK) completed, returning 0

I miss some step?

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