Asterisk says to process the call correctly:== Using SIP RTP TOS bits 184sipp says "Aborting call on an unexpected BYE for call: 96-1956@192.168.200.185" "asterisk -rx 'core show channels'|tail -n3" shows: 80 active channels -> constant 80 active calls -> constant 160 calls processed -> increase every second the sipp command I use is "./sipp 192.168.200.64 -sn uac -i 192.168.200.185 -s 17000 -d 90000 -l 10000 -r 100 -rp 30000 -t un" that generate 100 calls every 30 seconds. every call last 90 seconds. I'm not trying to break the limit of 10000 calls, I want just to have 200 or 300 calls. sip does not have setted any limit, and call-limit is deprecated in asterisk 1.8. On 30/03/2012 14:04, Danny Nicholas wrote: Check the sip.conf.sample file. I think it is the call-limit parameter that is getting you. The sample file should tell you what the default is. Another possibility is that your rtp range is set too low; the "normal" range is 10000-20000, which allows for 2500 calls(or 5000 if you set other things "correctly"). -----Original Message----- From: asterisk-users-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-users-bounces@xxxxxxxxxxxxxxxx] On Behalf Of Steven Howes Sent: Friday, March 30, 2012 7:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: concurrent channels limit |
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