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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1686/">https://reviewboard.asterisk.org/r/1686/</a>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">I made several comments on the first test that apply to all tests.
I think the coverage of the test could be improved by essentially re-running each test with the following variables:
1. Type of bridge in use. This can be controlled by the canreinvite setting and by adding various parameters to the Dial application calls in the dialplan.
Case A: Native remote RTP bridge. Set canreinvite=yes, make sure codecs are same on channels and no extra dial parameters are added.
Case B: Native local RTP bridge. Set canreinvite=no, make sure codecs are the same on channels and no extra dial parameters are added.
Case C: Asterisk generic bridge. Set canreinvite=no, and add parameters like 't' or 'T' to the dial application.
2. Which phone gets hung up.
Case A: Hang up the transferee.
Case B: Hang up the transfer target.
3. Whether Asterisk or PJSUA hangs up the call. This is like variable 2 except that instead of sending 'h' commands to PJSUA, you use a Manager hangup request on Asterisk.
Case A: Request hangup on the transferee's channel.
Case B: Request hangup on the transfer target's channel.</pre>
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<a href="https://reviewboard.asterisk.org/r/1686/diff/1/?file=23515#file23515line44" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_refer_only/run-test</a>
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(Diff revision 1)
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> self.ast[0].cli_exec("core show channels")# if channels are still up for some reason, we want to know that as well</pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Channels are going to be up. Specifically, the A and C channels are going to be up because they're still bridged.</pre>
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<a href="https://reviewboard.asterisk.org/r/1686/diff/1/?file=23515#file23515line58" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/sip_blind_transfer/callee_refer_only/run-test</a>
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(Diff revision 1)
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> reactor.callLater(5, self.a_call_b)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> reactor.callLater(10, self.b_transfer_a_to_c)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> reactor.callLater(15, self.ami_check_bridge)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> reactor.callLater(20, self.read_result)</pre></td>
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<pre style="margin-left: 2em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">I'm not a big fan of this. Waiting a set amount of seconds is the type of thing that can cause a test to fail for seemingly no good reason.
I like the idea of calling the various functions here based on events that occur rather than elapsed time.
So for instance, self.a_call_b should be called once the PJSUA processes are up and running. I don't know if subprocess.Popen waits to find out if the subprocess actually starts successfully. There's a reactor type for spawning processes that may be useful for this sort of thing. See http://twistedmatrix.com/documents/current/api/twisted.internet.interfaces.IReactorProcess.html
self.b_transfer_a_to_c should be called when we receive an AMI event stating that channels A and B are bridged. In 1.4, you can wait on the "Link" manager event, and in 1.6.2+ you can wait on the "Bridge" manager event with the BridgeState set to "Link".