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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3601/">https://reviewboard.asterisk.org/r/3601/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By rmudgett.</div>
<p style="color: grey;"><i>Updated June 23, 2014, 3:56 p.m.</i></p>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Changes</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;">Updated to make local channels swap accountcode and peeraccount across the special bridge between local;1 and local;2.
You can now set CHANNEL(peeraccount) before Dial, FollowMe, and Queue to have the outgoing channel use that accountcode.</pre>
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<div style="margin-top: 1.5em;">
<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/AFS-65">AFS-65</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 (updated)</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;">Accountcode propagation:
The current behavior is to simply set the accountcode of an outgoing
channel to the accountcode of the channel initiating the call. It was
done this way a long time ago to allow the accountcode set on the SIP/100
channel to be propagated to a local channel so the dialplan execution on
the Local;2 channel would have the SIP/100 accountcode available.
SIP/100 -> Local;1/Local;2 -> SIP/200
Propagating the SIP/100 accountcode to the local channels is very useful.
However, a side effect of this is it will overwrite any accountcode set by
the SIP channel driver configuration for the SIP/200 channel with the
accountcode from the Local;2 channel. Without any dialplan manipulation,
all channels in this call would have the same accountcode.
Using dialplan, you can set a different accountcode on the SIP/200 channel
either by setting the accountcode on the Local;2 channel or by the Dial
application's b(pre-dial), M(macro) or U(gosub) options, or by the
FollowMe application's b(pre-dial) option, or by the Queue application's
macro or gosub options. The altered accountcode on SIP/200 will remain
until the local channels optimized out when the accountcode would change
to the SIP/100 accountcode.
Asterisk v1.8 attempted to add peeraccount support but ultimately had to
punt on the support. The peeraccount support was rendered useless because
of how the CDR code needed to unconditionally force the caller's
accountcode onto the peer channel's accountcode. The CEL events were thus
intentionally made to always use the channel's accountcode as the
peeraccount value.
With the arrival of Asterisk v12, the situation has improved so
peeraccount support can be made to work. Using the indicated example,
the goal is to have the accountcode values become as follows:
SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200
acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200
peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100
If a channel already has an accountcode it can only change by explicit
user command by the following methods:
1) A channel originate method that can specify an accountcode to use.
2) Dialplan using CHANNEL(accountcode).
3) Dialplan using CHANNEL(peeraccount) on the other end of a local
channel pair.
If a channel does not have an accountcode it can get one from the
following places:
1) The channel driver's configuration at channel creation.
2) Before dialing from the calling channel's peeraccount value; or as a
legacy fallback the calling channel's accountcode.
3) Explicit user command as already indicated.
4) Entering a bridge from a peer channel's peeraccount value.
You can specify the accountcode for an outgoing channel that does not have
one by setting the CHANNEL(peeraccount) before using the Dial, FollowMe,
and Queue applications.
Accountcode and peeraccount values propagate to an outgoing channel before
dialing. Accountcodes also propagate when channels enter or leave a
bridge. The peeraccount value only makes sense for channels in two party
bridges.
* Made peeraccount functional by changing accountcode propagation as
described above.
* Fixed the basic bridge sub-class to update peeraccount codes when the
number of channels in the bridge drops back down to two parties.
* Refactored ast_bridge_channel_update_accountcodes() to handle channels
joining/leaving the bridge.
* Fixed the basic bridge to not call the base pull method twice.
* Fixed CEL extracting the wrong ie value for the peeraccount. This was
done intentionally in Asterisk v1.8 when that version had to punt on
peeraccount.
* Fixed some incorrect CEL event ie types. These seem to only be used
by the unit tests now.
* Fixed a few places dealing with accountcodes that were reading from
channels without the lock held.
NOTE: This patch is against v12 because that was the original target for
the patch. However, because the peeraccount changes were more extensive
than anticipated and resulted in a larger behavior change it will only be
checked into trunk.
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing (updated)</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;">Set the accountcode on the initial channel and let the channel driver supply or not the accountcode for the other channel.
Without the channel driver supplying the accountcode, the outgoing channel got the same accountcode as the initial channel.
With the channel driver supplying the accountcode, the outgoing channel kept its accountcode.
Let the channel driver supply the accountcode for all endpoint channels.
Performed a DTMF attended transfer with consultation and creation of a three party bridge.
When the transferrer channel left the three party bridge, the remaining two channels got the peeraccount updated to the other party.
Throughout each of these tests, the CEL log had the expected peeraccount value. Note the peeraccount value is meaningless if the bridge contains more than two parties.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> (updated)</h1>
<ul style="margin-left: 3em; padding-left: 0;">
<li>/branches/12/res/parking/parking_bridge_features.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/main/pbx.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/main/event.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/main/dial.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/main/core_unreal.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/main/channel.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/main/cel.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/main/bridge_channel.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/main/bridge_basic.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/main/bridge.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/include/asterisk/channel.h <span style="color: grey">(417163)</span></li>
<li>/branches/12/include/asterisk/bridge_channel.h <span style="color: grey">(417163)</span></li>
<li>/branches/12/apps/app_queue.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/apps/app_followme.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/apps/app_dial.c <span style="color: grey">(417163)</span></li>
<li>/branches/12/UPGRADE.txt <span style="color: grey">(417163)</span></li>
</ul>
<p><a href="https://reviewboard.asterisk.org/r/3601/diff/" style="margin-left: 3em;">View Diff</a></p>
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