[asterisk-dev] [Code Review]: Add SIP INFO DTMF test to the Asterisk Test Suite
Matt Jordan
reviewboard at asterisk.org
Thu Feb 9 09:34:44 CST 2012
> On Feb. 9, 2012, 9:21 a.m., Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/info_dtmf/run-test, line 43
> > <https://reviewboard.asterisk.org/r/1723/diff/4/?file=24028#file24028line43>
> >
> > redundant
No, its not. TestCase sets passed to False in its constructor.
> On Feb. 9, 2012, 9:21 a.m., Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/info_dtmf/configs/ast1/extensions.conf, line 6
> > <https://reviewboard.asterisk.org/r/1723/diff/4/?file=24026#file24026line6>
> >
> > boo, wait(10)
This is needed in this case, although I agree its annoying. If the SIPp scenario hangs up first, not all of the (many) DTMF events will get sent over AMI, as the associated channel no longer exists. So we need to wait some period of time and have Asterisk hang up. The test itself cannot coordinate this, as it blocks waiting for SIPp to exit. So this is the best we've got in this somewhat contrived case.
> On Feb. 9, 2012, 9:21 a.m., Paul Belanger wrote:
> > /asterisk/trunk/tests/channels/SIP/info_dtmf/configs/ast1/sip.conf, line 1
> > <https://reviewboard.asterisk.org/r/1723/diff/4/?file=24027#file24027line1>
> >
> > udpbindaddr=127.0.0.1:5060
Done, but this shouldn't be necessary since by default, we'll bind to all interfaces on port 5060.
- Matt
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On Feb. 9, 2012, 9:15 a.m., Matt Jordan wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1723/
> -----------------------------------------------------------
>
> (Updated Feb. 9, 2012, 9:15 a.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Summary
> -------
>
> This adds a test to the Asterisk Test Suite to cover the regression that was introduced in ASTERISK-18924. This test sends a sequence of SIP INFO requests containing DTMF events covering 0-9, A-D, a-d, and 10-16. This checks only for SIP INFO requests that have a Content-Type of application/dtmf-relay; the test could be expanded to cover the less used Content-Type of application/dtmf if we feel its necessary.
>
>
> This addresses bug ASTERISK-19290.
> https://issues.asterisk.org/jira/browse/ASTERISK-19290
>
>
> Diffs
> -----
>
> /asterisk/trunk/tests/channels/SIP/info_dtmf/configs/ast1/extensions.conf PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/info_dtmf/configs/ast1/sip.conf PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/info_dtmf/run-test PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/info_dtmf/sipp/dtmf-relay.xml PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/info_dtmf/sipp/dtmf.xml PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/info_dtmf/test-config.yaml PRE-CREATION
> /asterisk/trunk/tests/channels/SIP/tests.yaml 3027
>
> Diff: https://reviewboard.asterisk.org/r/1723/diff
>
>
> Testing
> -------
>
>
> Thanks,
>
> Matt
>
>
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