From: Ioan I. <ioa...@mo...> - 2007-05-30 13:02:48
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Hello, I come back with the following report: 1. I add some swap but I experience the same crash (asterisk segmentation fault at 46 simultaneous SIP calls - with or without RTP traffic). 2. In order to avoid any discussions regarding Sipp, I create a simple test scenario: a. extensions.conf [test] exten => 000,1,Answer exten => 000,2,Echo exten => 000,3,Hangup exten => 001,1,Answer exten => 001,2,Playback(demo-congrats) exten => 001,3,Hangup b. create a call file (/tmp/test.call) Channel: Local/000@test MaxRetries: 0 Context: test Extension: 001 Priority: 1 c. Start the test: in one terminal run: # watch -n 1 asterisk -rx "show channels" in the seccond terminal run several times: # cp /tmp/test.call /var/spool/asterisk/outgoing/`date +%s`.call; free this will add 2 active calls. Result of the test: On our PBX (astlinux 0.4.5 on Via EN12000EG), asterisk crash at the transition from 46 to 48 channels: .... pbx asterisk # cp /tmp/test.call.orig /var/spool/asterisk/outgoing/`date +%s`.call; free total used free shared buffers Mem: 971136 37144 933992 0 1396 Swap: 2097136 0 2097136 Total: 3068272 37144 3031128 pbx asterisk # cp /tmp/test.call.orig /var/spool/asterisk/outgoing/`date +%s`.call; free total used free shared buffers Mem: 971136 32704 938432 0 1396 Swap: 2097136 0 2097136 Total: 3068272 32704 3035568 Could anybody confirmed that using astlinux 0.4.5 on their system, they could have more than 48 active calls using this simple test? Best regards, Ioan. www.modulo.ro |