[Asterisk-Dev] Protos SIP Test

[Mark Spencer]

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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