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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2337/">https://reviewboard.asterisk.org/r/2337/</a>
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<p style="margin-top: 0;">On February 21st, 2013, 4:59 p.m., <b>opticron</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/2337/diff/1/?file=33505#file33505line8" style="color: black; font-weight: bold; text-decoration: underline;">/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_media_restrict.xml</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; "> <span class="nt"><action></span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="nt"><ereg</span> <span class="na">regexp=</span><span class="s">".*"</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="na">header=</span><span class="s">"Call-ID:"</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="na">search_in=</span><span class="s">"hdr"</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="na">check_it=</span><span class="s">"true"</span></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> <span class="na">assign_to=</span><span class="s">"global_call_id"</span><span class="nt">/></span></pre></td>
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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;">Picking out the Call-ID is unnecessary in this case. [call_id] or [last_Call-ID] will work for this purpose.</pre>
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<p>On February 21st, 2013, 8:50 p.m., <b>Matt Jordan</b> wrote:</p>
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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'm not sure they will work in this scenario. If I recall correctly, [call_id] generates a different Call-ID then the received INVITE request, which makes the subsequent INVITE request that the scenario sends a different dialog. You can try [last_Call-ID], but I vaguely remember that SIPp doesn't handle re-INVITEs properly in this regard.
It's possible I'm wrong so it's worth trying, but when I first wrote the SIPp hold tests I spent quite awhile hacking around on the Call-ID trying to get the re-INVITE to work.</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">I double-checked http://svn.digium.com/svn/testsuite/asterisk/trunk/tests/channels/SIP/session_timers/basic_uac_refresh/sipp/uas-no-hangup.xml and it seems to be operating as expected. It uses [last_Call-ID] throughout, but only accepts reinvites and does not initiate them so it's a slightly different use case.</pre>
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<p>- opticron</p>
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<p>On February 18th, 2013, 2:30 p.m., Kevin Harwell wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Kevin Harwell.</div>
<p style="color: grey;"><i>Updated Feb. 18, 2013, 2:30 p.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Tests to make sure the music on hold event is not triggered if the "discard_remote_hold_retrieval" option is set to "yes". In the test multiple scenarios are ran where one SIP phone puts another SIP phone on hold by sending a re-INVITE with a modified SDP containing either a restricted audio direction, an IP address of 0.0.0.0, or a combination thereof. This is tested both for a local RTP bridge, and a non-bridged scenario.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Ran the test and made sure all scenarios passed. Also set the discard_remote_hold_retrieval to "no" and made sure the test failed. </pre>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ABE-2899">ABE-2899</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/configs/ast1/extensions.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/configs/ast1/sip.conf <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_A.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_IP_media_restrict.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_IP_restrict.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/sipp/phone_B_media_restrict.xml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/sip_hold_no_moh/test-config.yaml <span style="color: grey">(PRE-CREATION)</span></li>
<li>/asterisk/trunk/tests/channels/SIP/tests.yaml <span style="color: grey">(3642)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2337/diff/" style="margin-left: 3em;">View Diff</a></p>
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