[Asterisk-Dev] Protos SIP Test

[Nickolay Shestakov]

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