> On Sept. 19, 2014, 2:34 p.m., Mark Michelson wrote: > > There are a couple of problems with this test: > > > > 1) It's quite a bit more complicated than it needs to be. What's actually > > being tested here is that Asterisk does not send a 503 in addition to a 486 > > on an INVITE retransmission. This only requires a UA to send and retransmit > > the INVITE and Asterisk. This should be doable entirely within a SIPp > > scenario and does not need voxcallcontrol, a Perl AGI load balancer, or > > grepping any logs. > > 2) The included sip.conf file is about 95% comments. This makes reviewing > > the config file much more difficult than it needs to be. > > > > I think this test can be accomplished using the SIPpTestCase and defining > > the test details entirely within test-config.yaml, with no run-test file > > necessary. If you're unfamiliar with how this is done, there are several > > examples of this in the testsuite. A simple example can be found at > > tests/channels/SIP/directrtpsetup/test-config.yaml. In that test, there is > > a test-modules section that tells the testsuite to use sipp.SIPpTestCase as > > the main test object for the test (it also has an unnecessary > > add-test-to-search-path option set. You can ignore that). The corresponding > > test-object-config provides details about the SIPp scenarios to run. In > > that test, there are two scenarios run, but I suspect that for your test, > > you would only need a single scenario to run. If you're curious about what > > options are available for configuring the SIPpTestCase, you can look in > > sample-yaml/sipptestcase-config.yaml.sample for some more details. Your > > test can pass or fail based on whether the SIPp scen ario succeeds or fails.
The problem is that I have to ignore INVITES for awhile by using a pause block. The 503 happens during that ignoring time after loosing a day and a half I couldn't find a better way then to pause then grep the log to see what happened. If you can make a version that can work entirely in sipp let me know :-) agiload balancer and voxcallcontrol is there by mistake, (thought I had removed all references to them already) I'll check again and remove the references and post the new patch - Torrey ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4006/#review13368 ----------------------------------------------------------- On Sept. 19, 2014, 8:09 a.m., Torrey Searle wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4006/ > ----------------------------------------------------------- > > (Updated Sept. 19, 2014, 8:09 a.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-24335 > https://issues.asterisk.org/jira/browse/ASTERISK-24335 > > > Repository: testsuite > > > Description > ------- > > This is a test for the test suite to reproduce the issue described in > ASTERISK-24335 > > > Diffs > ----- > > /asterisk/trunk/tests/channels/SIP/tests.yaml 5608 > /asterisk/trunk/tests/channels/SIP/invite_retransmit/test-config.yaml > PRE-CREATION > /asterisk/trunk/tests/channels/SIP/invite_retransmit/sipp/A_PARTY.xml > PRE-CREATION > /asterisk/trunk/tests/channels/SIP/invite_retransmit/run-test PRE-CREATION > /asterisk/trunk/tests/channels/SIP/invite_retransmit/configs/ast1/sip.conf > PRE-CREATION > > /asterisk/trunk/tests/channels/SIP/invite_retransmit/configs/ast1/extensions.conf > PRE-CREATION > > Diff: https://reviewboard.asterisk.org/r/4006/diff/ > > > Testing > ------- > > test passes when 4003 patch applied, fails when patch not applied > > > Thanks, > > Torrey Searle > >
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