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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4316/">https://reviewboard.asterisk.org/r/4316/</a>
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<p style="margin-top: 0;">On January 19th, 2015, 4:25 p.m. UTC, <b>Joshua Colp</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/4316/diff/1/?file=70700#file70700line155" style="color: black; font-weight: bold; text-decoration: underline;">/branches/13/res/res_pjsip_multihomed.c</a>
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(Diff revision 1)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "></pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span><span class="tb"> </span><span class="n">ast_debug</span><span class="p">(</span><span class="mi">4</span><span class="p">,</span> <span class="s">"Re-wrote Contact URI host/port to %s:%d</span><span class="se">\n</span><span class="s">"</span><span class="p">,</span> <span class="n">pj_strbuf</span><span class="p">(</span><span class="o">&</span><span class="n">uri</span><span class="o">-></span><span class="n">host</span><span class="p">),</span> <span class="n">uri</span><span class="o">-></span><span class="n">port</span><span class="p">);</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;">There's no guarantee that the host will be NULL terminated.</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;">If you're looking for how we typically go about printing pj_str_t, you can find examples in res_pjsip.c. Taking a quick look, there are two examples in register_service_noref().</pre>
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<p>- Mark</p>
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<p>On January 19th, 2015, 3:16 a.m. UTC, Matt Jordan wrote:</p>
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<div>Review request for Asterisk Developers and Joshua Colp.</div>
<div>By Matt Jordan.</div>
<p style="color: grey;"><i>Updated Jan. 19, 2015, 3:16 a.m.</i></p>
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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/ASTERISK-24015">ASTERISK-24015</a>,
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24703">ASTERISK-24703</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<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;">This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability.
*New Feature*
A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology.
- Preemptive question: why 'redirect', and not 'transfer'? Mostly because 'transfer' was always kind of a bad name. If the channel isn't answered, we aren't transferring, we're forwarding. If it is answered, the type of transfer being performed is somewhat vague - is it blind? Is it attended? 'redirect' - while also a slightly loaded term - is a bit more generic and yet descriptive of what is happening: we're redirecting the channel to somewhere else. Answered, not answered, it doesn't matter: your channel is no good here!
*Bug fixes*
In the process of writing this new feature, two bugs were fixed in the PJSIP stack:
(1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to.
(2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers.</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;">Tests were written both for the PJSIP stack as well as the new ARI operation. See https://reviewboard.asterisk.org/r/4352.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/13/rest-api/api-docs/channels.json <span style="color: grey">(430751)</span></li>
<li>/branches/13/res/stasis/control.c <span style="color: grey">(430751)</span></li>
<li>/branches/13/res/res_pjsip_transport_websocket.c <span style="color: grey">(430751)</span></li>
<li>/branches/13/res/res_pjsip_nat.c <span style="color: grey">(430751)</span></li>
<li>/branches/13/res/res_pjsip_multihomed.c <span style="color: grey">(430751)</span></li>
<li>/branches/13/res/res_ari_channels.c <span style="color: grey">(430751)</span></li>
<li>/branches/13/res/ari/resource_channels.c <span style="color: grey">(430751)</span></li>
<li>/branches/13/res/ari/resource_channels.h <span style="color: grey">(430751)</span></li>
<li>/branches/13/include/asterisk/stasis_app.h <span style="color: grey">(430751)</span></li>
<li>/branches/13/channels/chan_pjsip.c <span style="color: grey">(430751)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/4316/diff/" style="margin-left: 3em;">View Diff</a></p>
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