[Asterisk-Dev] Protos SIP Test
Nickolay Shestakov
npshe at mail.ru
Mon Mar 17 18:18:20 MST 2003
Yes, "sip show channels" show the channels are active.
And, now I've noticed, that in my simple config: 10.0.0.1- Asterisk server,
10.0.0.2 -Cisco 7960, 10.0.0.3 -eStara Softphone
after the first call "sip show channels" shows the 1 active SIP channel,
Peer - 10.0.0.1, Username (None)...
after the second - 2 active SIP channels, Peer - 10.0.0.1, 10.0.01
and so on...
What's is the reason may be?
Nickolay
-----Original Message-----
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com]On Behalf Of Mark Spencer
Sent: Tuesday, March 18, 2003 5:38 AM
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] Protos SIP Test
Does it not tear down the calls? Does "sip show channels" show the
channels are still there?
Mark
On Tue, 18 Mar 2003, Nickolay Shestakov wrote:
> Hi, all
> I have tested the newest CVS version * with PROTOS SIP Test-Suite (see
> http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/sip/ ,
> http://www.cert.org/advisories/CA-2003-06.html )
> It sends various INVITE test-cases (4526 testcases). The first results
are:
> Test 1: java -jar c07-sip-r1.jar -touri
> user at asterisk_server -teardown -validcase
> INVITE test-case
> CANCEL
> ACK for the teardown
> valid INVITE
> CANCEL for the valid INVITE
> ACK for the valid INVITE teardown
> Test is ok.
> Test 2: : java -jar c07-sip-r1.jar -touri user at asterisk_server -fromuri
> user at 1.2.3.4
> Only INVITE test-case
> Everything is ok, but the number of active sip channels grows during this
> test, and when it == 1003 asterisk stops answer :
> WARNING[7176] : File rtp.c, Line 533 (ast_rtp_new): Unable to allocate
> socket: Too many open files
> WARNING[7176] : File chan_sip.c, Line 1167 (sip_alloc): Unable to create
RTP
> session: Too many open files
> My sip context is very simple (only for test):
> [incoming]
> exten => 1001,1,Dial,SIP/BYEXTENSION,20
> exten => t,1,Hangup
> exten => i,1,Hangup
>
> It seems, that * creates sip channels, and don't destroy it . Any ideas ?
>
> Regards,
> Nickolay
>
>
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