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Asterisk says to process the call correctly:<br>
<blockquote> == Using SIP RTP TOS bits 184<br>
== Using SIP RTP CoS mark 5<br>
-- Executing [17000@sipp:1] Answer("SIP/sipp-0000005a", "") in
new stack<br>
-- Executing [17000@sipp:2] Set("SIP/sipp-0000005a", "rn=100")
in new stack<br>
-- Executing [17000@sipp:3] Goto("SIP/sipp-0000005a",
"set100") in new stack<br>
-- Goto (sipp,17000,12)<br>
-- Executing [17000@sipp:12] Answer("SIP/sipp-0000005a", "")
in new stack<br>
-- Executing [17000@sipp:13] BackGround("SIP/sipp-0000005a",
"you-seem-impatient") in new stack<br>
-- <SIP/sipp-0000005a> Playing 'you-seem-impatient.ulaw'
(language 'en')<br>
-- Executing [17000@sipp:14] Wait("SIP/sipp-00000055", "20")
in new stack<br>
</blockquote>
sipp says "Aborting call on an unexpected BYE for call:
<a class="moz-txt-link-abbreviated" href="mailto:96-1956@192.168.200.185">96-1956@192.168.200.185</a>"<br>
<br>
"asterisk -rx 'core show channels'|tail -n3" shows:<br>
80 active channels -> constant<br>
80 active calls -> constant<br>
160 calls processed -> increase every second<br>
<br>
<br>
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"<br>
that generate 100 calls every 30 seconds. every call last 90
seconds.<br>
<br>
I'm not trying to break the limit of 10000 calls, I want just to
have 200 or 300 calls.<br>
sip does not have setted any limit, and call-limit is deprecated in
asterisk 1.8.<br>
<br>
<br>
On 30/03/2012 14:04, Danny Nicholas wrote:
<blockquote cite="mid:002c01cd0e75$af3d3900$0db7ab00$@debsinc.com"
type="cite">
<pre wrap="">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: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Steven Howes
Sent: Friday, March 30, 2012 7:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] concurrent channels limit
</pre>
</blockquote>
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