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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/2513/">https://reviewboard.asterisk.org/r/2513/</a>
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<p style="margin-top: 0;">On May 8th, 2013, 11:31 p.m. UTC, <b>Paul Belanger</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;">Mostly because I am curious, why is this change needed for chan_sip? Is this for the updates to the bridging API?</pre>
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<p>On May 8th, 2013, 11:37 p.m. UTC, <b>rmudgett</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;">Yes, that is why this is being done.</pre>
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<p>On May 8th, 2013, 11:46 p.m. UTC, <b>Paul Belanger</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;">Does this mean anything using the bridge api would need to be updated? Or is this a nice to have?</pre>
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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;">All of the channel drivers in Asterisk 12 will need to call ast_bridge_transfer_blind() and/or ast_bridge_transfer_attended() to support transfers initiated by the channel driver's protocol (SIP, IAX2, ISDN, etc.). The nice thing about this change for existing channel drivers is that it mostly involves deleting code.</pre>
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<p>- rmudgett</p>
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<p>On May 8th, 2013, 7:02 p.m. UTC, Mark Michelson wrote:</p>
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<div>Review request for Asterisk Developers and Joshua Colp.</div>
<div>By Mark Michelson.</div>
<p style="color: grey;"><i>Updated May 8, 2013, 7:02 p.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-21520">ASTERISK-21520</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 converts attended transfers in chan_sip.c to use ast_bridge_transfer_attended(). The majority of the change is in the local_attended_transfer() function and handle_invite_replaces().</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;">Tested this by performing attended transfers with the transferer being in different states of bridging.
* Two bridged transferer channels
* Initially unbridged transferer to bridged transferer
* Initially bridged transferer to unbridged transferer.
Used SIPp to send INVITE with Replaces in order to take the place of a channel. Tested this both with taking the place of bridged and unbridged channels.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/team/mmichelson/sip_transfer/channels/chan_sip.c <span style="color: grey">(388013)</span></li>
<li>/team/mmichelson/sip_transfer/channels/sip/include/sip.h <span style="color: grey">(388013)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/2513/diff/" style="margin-left: 3em;">View Diff</a></p>
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