On Mon, Mar 14, 2016 at 7:41 PM, Tickling Contest <tickling.contest@xxxxxxxxx> wrote:
(1) https://github.com/asterisk/testsuite
(2) https://github.com/asterisk/testsuite/tree/master/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_user
(3) https://github.com/asterisk/testsuite/blob/master/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_user/sipp/playback_with_initial_sdp.xml
Also, it would be amazing if someone could tell me how to stop a SIPp test after one such call described in my OP.
A SIPp scenario will stop either when it encounters an error or runs to completion. You have to form your scenario in a way that it will execute all commands and receive all expected events in the proper order. For instance maybe in your scenario you'll have a call enter your application. The application does what it needs to do with the channel and then hangs it up. The SIPp scenario can be waiting to receive the hang up. Once it does it will end.
See more below about an example.
On Mon, Mar 14, 2016 at 8:37 PM, Tickling Contest <tickling.contest@xxxxxxxxx> wrote:Hello,I am testing an ARI application written over Asterisk 13.6.0. In my ARI application, I wrote a bunch of features that manipulate call state, and I would now like to stress test it so that the application is production ready.So, I am using SIPp, but I am having a LOT of trouble with docs.Here's what I want to do, assuming each SIPp "peer" in the following description means a single SIPp process on, say, a linux host:(a) REGISTER SIPp peer A and SIPp peer B to my Asterisk 13.6.0 using TCP transport using username and password.(b) Use SIPp peer A to call SIPp peer B, both of whom are registered to my Asterisk 13.6.0/TCP PBX using INVITE.(c) Wait for the call to be answered at peer B, and then pause 2 seconds and then kill the call by sending a BYE from SIPp peer B.Here is what I am able to do so far: I am able to REGISTER the peers, but for the life of me, not able to get the peers to call each other.I have tried sippy_cup, but I have had a similar issue where the documentation is spotty and I don't think it actually works for TCP. So that's not an option for me.I would like to know how to create XML scenario files so that I can generate my own scenario files for my own set up from actual pjsip set logger on output.
If you would like to simply originate a call from SIPp into Asterisk you can find many such examples that do just that in the Asterisk Testsuite (1). One such example can be found beneath the pjsip channel driver tests: ident_by_user (2). Check out the playback_with_initial_sdp.xml (3) file specifically.
Note, this example calls the playback extension, which then plays a file back and then hangs up. Instead of calling the playback app your's would call your app instead.
(1) https://github.com/asterisk/testsuite
(2) https://github.com/asterisk/testsuite/tree/master/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_user
(3) https://github.com/asterisk/testsuite/blob/master/tests/channels/pjsip/basic_calls/incoming/nominal/unauthed/ident_by_user/sipp/playback_with_initial_sdp.xml
Any help is deeply appreciated!Thanks!
--
Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org
_______________________________________________ asterisk-app-dev mailing list asterisk-app-dev@xxxxxxxxxxxxxxxx http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
