Review Board 1.7.16


accountcode: Slightly change accountcode propagation.

Review Request #3601 - Created June 6, 2014 and submitted

rmudgett
trunk
AFS-65
Reviewers
asterisk-dev
Asterisk
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.
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.  Before Asterisk v12, the altered accountcode on
SIP/200 will remain until the local channels optimize out and 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 somewhat so
peeraccount support can be made to work.  Using the indicated example, the
the accountcode values become as follows when the peeraccount is set on
SIP/100 before calling SIP/200:

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 the
following explicit user actions:

1) A channel originate method that can specify an accountcode to use.

2) The calling channel propagating its non-empty peeraccount or its
non-empty accountcode if the peeraccount was empty to the outgoing
channel's accountcode before initiating the dial.  e.g., Dial and
FollowMe.  The exception to this propagation method is Queue.  Queue will
only propagate peeraccounts this way only if the outgoing channel does not
have an accountcode.

3) Dialplan using CHANNEL(accountcode).

4) 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) Explicit user action as already indicated.

3) Entering a basic or stasis-mixing bridge from a peer channel's
peeraccount value.

You can specify the accountcode for an outgoing channel by setting the
CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue
applications.  Queue adds the wrinkle that it will not overwrite an
existing accountcode on the outgoing channel with the calling channels
values.

Accountcode and peeraccount values propagate to an outgoing channel before
dialing.  Accountcodes also propagate when channels enter or leave a basic
or stasis-mixing bridge.  The peeraccount value only makes sense for
mixing bridges with two channels; it is meaningless otherwise.

* Made peeraccount functional by changing accountcode propagation as
described above.

* 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 a few places dealing with accountcodes that were reading from
channels without the lock held.
* Ran tests from https://reviewboard.asterisk.org/r/3832/ all tests tagged
with accountcode pass.

* Set the accountcode on the calling channel and let the channel driver
supply or not the accountcode for the outgoing channel.  The acccountcode
on the calling channel forces itself on the outgoing channel.  This is the
same as previous behavior.

* Set the accountcode and peeraccount code on the calling channel and let
the channel driver supply or not the acccountcode for the outgoing
channel.  The outgoing channel's accountcode and peeraccount got the
calling channel's peeraccount and accountcode values respectively.

* Let dialplan set accountcodes on channels and performed a DTMF attended
transfer to create 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.  Unfortunately, you only see the
updated peeraccount values in the CEL log when the channels leave the
bridge.

Throughout each of these tests, the CEL log had the expected peeraccount
value.  Some interpolation is needed in the CEL log for complicated
accountcode manipulation because there aren't enough events to indicate
when the account codes change.  Note the peeraccount value is meaningless
if the bridge does not contain two parties.
/branches/12/UPGRADE.txt
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===========================================================
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===========================================================
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From 12.3.1 to 12.4.0:
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From 12.3.1 to 12.4.0:
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 - The accountcode propagation is now consistently propagated to outgoing
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 - The accountcode propagation is now consistently propagated to outgoing
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   channels before dialing.  In addition, if a channel already has an
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   channels before dialing.  The channel accountcode can change from its
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   accountcode then it will not be changed unless explicitly set by
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   original non-empty value on channel creation for the following specific
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   CHANNEL(accountcode) or an originate method that can specify an
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   reasons.  One, dialplan sets it using CHANNEL(accountcode).  Two, an
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   accountcode value.  The change has three visible effects.  One, if the
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   originate method that can specify an accountcode value.  Three, the
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   channel driver is configured with an accountcode for new channels then
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   calling channel propagates its peeraccount or accountcode to the
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   those channels will always have that accountcode unless explicitly
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   outgoing channel's accountcode before dialing.  The change has two
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   changed.  Two, local channels now cross accountcode and peeraccount
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   visible effects.  One, local channels now cross accountcode and
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   across the special bridge between the ;1 and ;2 channels just like
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   peeraccount across the special bridge between the ;1 and ;2 channels
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   channels between normal bridges.  Three, the CHANNEL(peeraccount) value
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   just like channels between normal bridges.  Two, the CHANNEL(peeraccount)
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   now can be set before Dial, FollowMe, and Queue to set an empty
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   value now can be set before Dial, FollowMe, and Queue to set the
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   accountcode on the outgoing channel(s).  For Queue this now includes
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   accountcode on the outgoing channel(s).  For Queue this now includes
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   local channel members where the accountcodes are propagated early
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   local channel members where the accountcodes are propagated early
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   enough to be useful on the ;2 channel.
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   enough to be useful on the ;2 channel.
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 - The safe_asterisk script was previously not installed on top of an existing
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 - The safe_asterisk script was previously not installed on top of an existing
[+20] [20] 6 lines
[+20]
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   you use tools to parse either of them, update your parse functions
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   you use tools to parse either of them, update your parse functions
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   accordingly. The changed strings are:
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   accordingly. The changed strings are:
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   - "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
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   - "Exited on signal $EXITSIGNAL" => "Asterisk exited on signal $EXITSIGNAL."
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   - "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"
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   - "Asterisk Died" => "Asterisk on $MACHINE died (sig $EXITSIGNAL)"
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 - Added a compatibility option for ari, chan_sip, and chan_pjsip

    
   
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   'websocket_write_timeout'. When a websocket connection exists where Asterisk

    
   
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   writes a substantial amount of data to the connected client, and the connected

    
   
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   client is slow to process the received data, the socket may be disconnected.

    
   
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   In such cases, it may be necessary to adjust this value. Default is 100 ms.

    
   
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 - Added a 'force_avp' option to chan_pjsip which will force the usage of

    
   
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   'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' as the media transport type

    
   
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   in SDP offers depending on settings, even when DTLS is used for media

    
   
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   encryption.

    
   
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 - Added a 'media_use_received_transport' option to chan_pjsip which will

    
   
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   cause the SDP answer to use the media transport as received in the SDP

    
   
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   offer.

    
   
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 - Added a 'force_avp' option for chan_sip. When enabled this option will

    
   
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   cause the media transport in the offer or answer SDP to be 'RTP/AVP',

    
   
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   'RTP/AVPF', 'RTP/SAVP', or 'RTP/SAVPF' even if a DTLS stream has been

    
   
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   configured. This option can be set to improve interoperability with WebRTC

    
   
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   clients that don't use the RFC defined transport for DTLS.

    
   
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 - The 'dtlsverify' option in chan_sip now has additional values besides

    
   
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   'yes' and 'no'. If 'yes' is specified both the certificate and fingerprint

    
   
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   will be verified. If 'no' is specified then neither the certificate or

    
   
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   fingerprint is verified. If 'certificate' is specified then only the

    
   
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   certificate is verified. If 'fingerprint' is specified then only the

    
   
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   fingerprint is verified.

    
   
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 - A 'dtlsfingerprint' option has been added to chan_sip which allows the

    
   
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   hash to be specified for the DTLS fingerprint placed in SDP. Supported

    
   
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   values are 'sha-1' and 'sha-256' with 'sha-256' being the default.

    
   
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From 12.3.0 to 12.3.1:
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From 12.3.0 to 12.3.1:
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 - MixMonitor AMI actions now require users to have authorization classes.
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 - MixMonitor AMI actions now require users to have authorization classes.
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   * MixMonitor - system
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   * MixMonitor - system
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   * MixMonitorMute - call or system
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   * MixMonitorMute - call or system
[+20] [20] 676 lines
/branches/12/apps/app_dial.c
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/branches/12/apps/app_followme.c
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/branches/12/apps/app_queue.c
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/branches/12/include/asterisk/bridge_channel.h
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/branches/12/include/asterisk/channel.h
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/branches/12/main/bridge.c
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/branches/12/main/bridge_basic.c
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/branches/12/main/bridge_channel.c
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/branches/12/main/cel.c
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/branches/12/main/channel.c
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/branches/12/main/core_unreal.c
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/branches/12/main/dial.c
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/branches/12/main/event.c
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/branches/12/main/pbx.c
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/branches/12/res/parking/parking_bridge_features.c
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  1. /branches/12/UPGRADE.txt: Loading...
  2. /branches/12/apps/app_dial.c: Loading...
  3. /branches/12/apps/app_followme.c: Loading...
  4. /branches/12/apps/app_queue.c: Loading...
  5. /branches/12/include/asterisk/bridge_channel.h: Loading...
  6. /branches/12/include/asterisk/channel.h: Loading...
  7. /branches/12/main/bridge.c: Loading...
  8. /branches/12/main/bridge_basic.c: Loading...
  9. /branches/12/main/bridge_channel.c: Loading...
  10. /branches/12/main/cel.c: Loading...
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  15. /branches/12/main/pbx.c: Loading...
  16. /branches/12/res/parking/parking_bridge_features.c: Loading...

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