Re: [asterisk-users] Asterisk Test Suite error
- Original Message - From: upendra uppi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 18, 2012 12:30:04 AM Subject: Re: [asterisk-users] Asterisk Test Suite error Hi Matthew , i have enabled the framework and tested the script, after running i am getting some FAILS - tests/channels/SIP/refer_replaces_to_self --- FAILED -- tests/channels/SIP/sip_tls_call --- FAILED -- tests/channels/SIP/sip_cause --- FAILED -- tests/masquerade --- FAILED let me know still what i am missing in the testsuite. So, as I explained earlier, the Asterisk Test Suite will tell you what dependencies you are missing. If a dependency is missing, it will skip the test and not execute it. Since the tests appear to have executed and failed, it isn't a dependency problem. So you probably aren't missing anything in the Test Suite. When a test fails, it will provide you with the messages from the Test Suite log file (located in ./logs) of verbosity WARNING and higher. Those will often (but not always) contain the reasons for the test failure. Even then, more often than not, you have to inspect the Test Suite logs and sometimes the archived Asterisk logs to determine why the test failed. As much as I'd love to say I'll debug your errors for you, the fact is that some of those tests are rather complex and debugging failures in them can take some significant effort. For example, the masquerade test creates 300 Local channels and collapses them all down through optimization. Finding the problem when that test fails is a non-trivial effort. (And these tests do pass on the current Bamboo build agent, as well as on my development machine. So off the top of my head, I don't know why they would be failing on your machine.) The Asterisk Test Suite is a tool to aid in Asterisk development and test. If you don't feel comfortable debugging problems in Asterisk, then it might not be the tool for you. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Test Suite error
Hi Matthew , i have enabled the framework and tested the script, after running i am getting some FAILS - tests/channels/SIP/refer_replaces_to_self --- FAILED -- tests/channels/SIP/sip_tls_call --- FAILED -- tests/channels/SIP/sip_cause --- FAILED -- tests/masquerade --- FAILED let me know still what i am missing in the testsuite. Regards Upendra On Thu, Sep 6, 2012 at 6:41 PM, Matthew Jordan mjor...@digium.com wrote: - Original Message - From: upendra uppi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, September 6, 2012 2:22:24 AM Subject: [asterisk-users] Asterisk Test Suite error Hi, i am trying to install the Asterisk test suite on my ubuntu system , i have followed all the installlation steps as mentioned in the link ( http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/ ) , but when i am trying to run the script some of the test cases are PASSED and most of them are FAILED and SKIPPPED. So please help me out to do the testing correctly. The following is the test report i got after running the script. snip There is newer documentation on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation There are pages that describe how to install and run the Test Suite under that page. Skipped tests happen when you are missing a dependency that the test needs to run. If you look at a verbose report for each test (which should also be output when the test run finishes), it should tell you what dependency was missing. Most of your failed tests appear to be those that require the TEST_FRAMEWORK compile time option in Asterisk. You can enable that in menuselect when you have configured Asterisk with --enable-dev-mode. Those tests *should* have picked up the fact that the TEST_FRAMEWORK wasn't enabled and should have been skipped (so long as the Asterisk directory that the Test Suite is sitting in has that option enabled in its build options), but there is a known bug with those test's YAML configuration that is preventing them from picking up the TEST_FRAMEWORK flag. Until we can get that cleaned up, enabling the TEST_FRAMEWORK flag should resolve that problem. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Test Suite error
Hi, i am trying to install the Asterisk test suite on my ubuntu system , i have followed all the installlation steps as mentioned in the link ( http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/) , but when i am trying to run the script some of the test cases are PASSED and most of them are FAILED and SKIPPPED. So please help me out to do the testing correctly. The following is the test report i got after running the script. === TEST RESULTS === PATH: /usr/local/sbin:/usr/local/bin:/usr/sbin:/usr/bin:/sbin:/bin -- tests/example --- PASSED -- tests/dynamic-modules --- SKIPPED -- Dependency: twisted -- Met: True -- Dependency: starpy -- Met: True -- tests/manager/login --- PASSED -- tests/manager/action-events-response --- PASSED -- tests/manager/authlimit --- PASSED -- tests/manager/authtimeout --- PASSED -- tests/manager/acl-login --- SKIPPED -- Dependency: twisted -- Met: True -- Dependency: starpy -- Met: True -- tests/cdr/console_dial_sip_answer --- PASSED -- tests/cdr/console_dial_sip_busy --- PASSED -- tests/cdr/console_dial_sip_congestion --- PASSED -- tests/cdr/cdr_originate_sip_congestion_log --- SKIPPED -- Dependency: twisted -- Met: True -- Dependency: starpy -- Met: True -- tests/cdr/console_dial_sip_transfer --- PASSED -- tests/cdr/console_fork_after_busy_forward --- PASSED -- tests/cdr/console_fork_before_dial --- PASSED -- tests/cdr/cdr_fork_end_time --- PASSED -- tests/cdr/cdr_unanswered_yes --- PASSED -- tests/cdr/cdr_accountcode --- PASSED -- tests/cdr/cdr_userfield --- PASSED -- tests/cdr/nocdr --- PASSED -- tests/cdr/blind-transfer-accountcode --- PASSED -- tests/cdr/originate-cdr-disposition --- PASSED -- tests/cdr/app_dial_G_flag --- SKIPPED -- Dependency: bash -- Met: True -- Dependency: asttest -- Met: True -- Dependency: SIPp -- Met: True -- tests/cdr/batch_cdrs --- PASSED -- tests/dialplan --- PASSED -- tests/channels/SIP/acl_call --- SKIPPED -- Dependency: twisted -- Met: True -- Dependency: starpy -- Met: True -- tests/channels/SIP/options --- PASSED -- tests/channels/SIP/refer_replaces_to_self --- FAILED -- tests/channels/SIP/info_dtmf --- PASSED -- tests/channels/SIP/tcpauthlimit --- PASSED -- tests/channels/SIP/tcpauthtimeout --- PASSED -- tests/channels/SIP/sip_outbound_address --- SKIPPED -- Dependency: bash -- Met: True -- Dependency: asttest -- Met: True -- Dependency: SIPp -- Met: False -- tests/channels/SIP/sip_attended_transfer --- PASSED -- tests/channels/SIP/sip_attended_transfer_tcp --- SKIPPED -- Dependency: twisted -- Met: True -- Dependency: starpy -- Met: True -- Dependency: pjsua -- Met: True -- tests/channels/SIP/sip_attended_transfer_v6 --- SKIPPED -- Dependency: twisted -- Met: True -- Dependency: starpy -- Met: True -- Dependency: pjsua -- Met: True -- Dependency: ipv6 -- Met: True -- Dependency: pjsuav6 -- Met: True -- tests/channels/SIP/sip_blind_transfer/callee_refer_only --- PASSED -- tests/channels/SIP/sip_blind_transfer/callee_with_reinvite --- PASSED -- tests/channels/SIP/sip_blind_transfer/caller_refer_only --- PASSED -- tests/channels/SIP/sip_blind_transfer/caller_with_reinvite --- PASSED -- tests/channels/SIP/sip_one_legged_transfer --- PASSED -- tests/channels/SIP/sip_one_legged_transfer_v6 --- SKIPPED -- Dependency: twisted -- Met: True -- Dependency: starpy -- Met: True -- Dependency: pjsua -- Met: True -- Dependency: ipv6 -- Met: True -- Dependency: pjsuav6 -- Met: True -- tests/channels/SIP/sip_register --- PASSED -- tests/channels/SIP/sip_register_domain_acl --- PASSED -- tests/channels/SIP/sip_channel_params --- PASSED -- tests/channels/SIP/sip_tls_call --- FAILED -- tests/channels/SIP/sip_tls_register --- PASSED -- tests/channels/SIP/sip_srtp --- SKIPPED -- Dependency: twisted -- Met: True -- Dependency: starpy -- Met: True -- Dependency: res_srtp -- Met: False -- tests/channels/SIP/noload_res_srtp --- PASSED -- tests/channels/SIP/noload_res_srtp_attempt_srtp --- SKIPPED -- Dependency: twisted -- Met: True -- Dependency: starpy -- Met: True -- Dependency: res_srtp -- Met: False -- tests/channels/SIP/secure_bridge_media --- SKIPPED -- Dependency: twisted -- Met: True -- Dependency: starpy -- Met: True -- Dependency: res_srtp -- Met: False -- tests/channels/SIP/message_disabled --- PASSED -- tests/channels/SIP/message_unauth --- PASSED -- tests/channels/SIP/message_unauth_from --- PASSED -- tests/channels/SIP/message_auth --- PASSED -- tests/channels/SIP/message_auth_cust_hdr --- PASSED -- tests/channels/SIP/message_send_ami --- SKIPPED -- Dependency: SIPp -- Met: True -- tests/channels/SIP/message_from_call --- PASSED -- tests/channels/SIP/message_mark_all_outbound --- SKIPPED -- Dependency: SIPp -- Met: True -- tests/channels/SIP/handle_response_address_incomplete --- PASSED --
Re: [asterisk-users] Asterisk Test Suite error
- Original Message - From: upendra uppi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, September 6, 2012 2:22:24 AM Subject: [asterisk-users] Asterisk Test Suite error Hi, i am trying to install the Asterisk test suite on my ubuntu system , i have followed all the installlation steps as mentioned in the link ( http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/ ) , but when i am trying to run the script some of the test cases are PASSED and most of them are FAILED and SKIPPPED. So please help me out to do the testing correctly. The following is the test report i got after running the script. snip There is newer documentation on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation There are pages that describe how to install and run the Test Suite under that page. Skipped tests happen when you are missing a dependency that the test needs to run. If you look at a verbose report for each test (which should also be output when the test run finishes), it should tell you what dependency was missing. Most of your failed tests appear to be those that require the TEST_FRAMEWORK compile time option in Asterisk. You can enable that in menuselect when you have configured Asterisk with --enable-dev-mode. Those tests *should* have picked up the fact that the TEST_FRAMEWORK wasn't enabled and should have been skipped (so long as the Asterisk directory that the Test Suite is sitting in has that option enabled in its build options), but there is a known bug with those test's YAML configuration that is preventing them from picking up the TEST_FRAMEWORK flag. Until we can get that cleaned up, enabling the TEST_FRAMEWORK flag should resolve that problem. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
