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20 authors
  1. Jul 24, 2014
    • Richard Mudgett's avatar
      accountcode: Slightly change accountcode propagation. · a2ce95d9
      Richard Mudgett authored 
      The previous behavior was 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.

AFS-65 #close

Review: https://reviewboard.asterisk.org/r/3601/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      a2ce95d9
  3. Jul 23, 2014
  5. Jul 22, 2014
    • Matthew Jordan's avatar
      ARI: Fix endpoint/channel subscription issues; allow for subscriptions to tech · bb87796f
      Matthew Jordan authored 
      This patch serves two purposes:
(1) It fixes some bugs with endpoint subscriptions not reporting all of the
    channel events
(2) It serves as the preliminary work needed for ASTERISK-23692, which allows
    for sending/receiving arbitrary out of call text messages through ARI in a
    technology agnostic fashion.

The messaging functionality described on ASTERISK-23692 requires two things:
(1) The ability to send/receive messages associated with an endpoint. This is
    relatively straight forwards with the endpoint core in Asterisk now.
(2) The ability to send/receive messages associated with a technology and an
    arbitrary technology defined URI. This is less straight forward, as
    endpoints are formed from a tech + resource pair. We don't have a
    mechanism to note that a technology that *may* have endpoints exists.

This patch provides such a mechanism, and fixes a few bugs along the way.

The first major bug this patch fixes is the forwarding of channel messages
to their respective endpoints. Prior to this patch, there were two problems:
(1) Channel caching messages weren't forwarded. Thus, the endpoints missed
    most of the interesting bits (such as channel creation, destruction, state
    changes, etc.)
(2) Channels weren't associated with their endpoint until after creation.
    This resulted in endpoints missing the channel creation message, which
    limited the usefulness of the subscription in the first place (a major use
    case being 'tell me when this endpoint has a channel'). Unfortunately,
    this meant another parameter to ast_channel_alloc. Since not all channel
    technologies support an ast_endpoint, this patch makes such a call
    optional and opts for a new function, ast_channel_alloc_with_endpoint.

When endpoints are created, they will implicitly create a technology endpoint
for their technology (if one does not already exist). A technology endpoint is
special in that it has no state, cannot have channels created for it, cannot
be created explicitly, and cannot be destroyed except on shutdown. It does,
however, have all messages from other endpoints in its technology forwarded to
it.

Combined with the bug fixes, we now have Stasis messages being properly
forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar),
where bar has a single channel associated with it and foo has two channels
associated with it. The messages would be forwarded as follows:

channel PJSIP/foo-1 --
                      \
                       --> endpoint PJSIP/foo --
                      /                         \
channel PJSIP/foo-2 --                           \
                                                  ---- > endpoint PJSIP
                                                /
channel PJSIP/bar-1 -----> endpoint PJSIP/bar --

ARI, through the applications resource, can:
 - subscribe to endpoint:PJSIP/foo and get notifications for channels
   PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo
 - subscribe to endpoint:PJSIP/bar and get notifications for channels
   PJSIP/bar-1 and endpoint PJSIP/bar
 - subscribe to endpoint:PJSIP and get notifications for channels
   PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar

Note that since endpoint PJSIP never changes, it never has events itself. It
merely provides an aggregation point for all other endpoints in its technology
(which in turn aggregate all channel messages associated with that endpoint).

This patch also adds endpoints to res_xmpp and chan_motif, because the actual
messaging work will need it (messaging without XMPP is just sad).

Review: https://reviewboard.asterisk.org/r/3760/

ASTERISK-23692
........

Merged revisions 419196 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      bb87796f
    • Kinsey Moore's avatar
      Fix more dev-mode build issues · 9056c23b
      Kinsey Moore authored 
      ........

Merged revisions 419129 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 419162 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 419163 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      9056c23b
  7. Jul 21, 2014
  9. Jul 20, 2014
  11. Jul 18, 2014
  13. Jul 16, 2014
  15. Jul 09, 2014
  17. Jul 08, 2014
  19. Jul 07, 2014
    • Joshua Colp's avatar
      res_pjsip_dialog_info_body_generator: Add dialog-info+xml support for presence. · 534ffd84
      Joshua Colp authored 
      This module implements dialog-info+xml for the purposes of presence. This means
that phones such as Grandstreams can now subscribe to receive presence information
for an extension.

ASTERISK-21443 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3705/
........

Merged revisions 418116 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418117 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      534ffd84
    • Matthew Jordan's avatar
      ARI/res_stasis: Subscribe to both Local channel halves when originating to app · d4b436d0
      Matthew Jordan authored 
      This patch fixes two bugs:

1. When originating a channel into a Stasis application, we already create a
   subscription for the channel that is going into our Stasis app.
   Unfortunately, when you create a Local channel and pass it off to a Stasis
   app, you really aren't creating just one channel: you're creating two. This
   patch snags the second half of the Local channel pair (assuming it is a
   Local channel pair, but luckily core_local is kind about such assumptions)
   and subscribes to it as well.

2. Subscriptions are a bit sticky right now. If a subscription is made, the
   'interest' count gets bumped on the Stasis subscription - but unless
   something explicitly unsubscribes the channel, said subscription sticks
   around. This is not much of a problem is a user is creating the subscription
   - if they made it, they must want it. However, when we are creating
   implicit subscriptions, we need to make sure something clears them out.
   This patch takes a pessimistic approach: it watches the cache updates
   coming from Stasis and, if we notice that the cache just cleared out an
   object, we delete our subscription object. This keeps our ao2 container of
   Stasis forwards in an application from growing out of hand; it also is a
   bit more forgiving for end users who may not realize they were supposed to
   unsubscribe from that channel that just hung up.

Review: https://reviewboard.asterisk.org/r/3710/
#ASTERISK-23939 #close
........

Merged revisions 418089 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d4b436d0
    • Kinsey Moore's avatar
      CEL: Fix incorrect/missing extra field information · edcaa540
      Kinsey Moore authored 
      This corrects two issues with the extra field information in Asterisk
12+ in channel event logs.

It is possible to inject custom values into the dialstatus provided by
ast_channel_dial_type() Stasis messages that fall outside the
enumeration allowed for the DIALSTATUS channel variable. CEL now
filters for the allowed values and ignores other values.

The "hangupsource" extra field key is always blank if the far end
channel is a chan_pjsip channel. This is because the hangupsource is
never set for the pjsip channel driver. This change sets the
hangupsource whenever a hangup is queued for chan_pjsip channels.

This corrects an issue with the pjsip channel driver where the
hangupcause information was not being set properly.

Review: https://reviewboard.asterisk.org/r/3690/
........

Merged revisions 418071 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418084 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      edcaa540
  21. Jul 04, 2014
    • Matthew Jordan's avatar
      Remove many deprecated modules · 97834718
      Matthew Jordan authored 
      Billing records are fair,
To get paid is quite bright,
You should really use ODBC;
Good-bye cdr_sqlite.

Microsoft did once push H.323,
Hell, we all remember NetMeeting.
But try to compile chan_h323 now
And you will take quite a beating.

The XMPP and SIP war was fierce,
And in the distant fray
Was birthed res_jabber/chan_jingle;
But neither to stay.

For everyone did care and chase what Google professed.
"Free Internet Calling" was what devotees cried,
But Google did change the specs so often
That the developers were happy the day chan_gtalk died.

And then there was that odd application
Dedicated to the Polish tongue.
app_saycountpl was subsumed by Say;
One could say its bell was rung.

To read and parse a file from the dialplan
You could (I guess) use an application.
app_readfile did fill that purpose, but I think
A function is perhaps better in its creation.

Barging is rude, I'm not sure why we do it.
Inwardly, the caller will probably sigh.
But if you really must do it,
Don't use app_dahdibarge, use ChanSpy.

We all despise the sound of tinny robots
It makes our queues so cold.
To control such an abomination
It's better to not use Wait/SetMusicOnHold.

It's often nice to know properties of a channel
It makes our calls right
We have a nice function called CHANNEL
And so SIPCHANINFO is sent off into the night.

And now things get odd;
Apparently one could delimit with a colon
Properties from the SIPPEER function!
Commas are in; all others are done.

Finally, a word on pipes and commas.
We're sorry. We can't say it enough.
But those compatibility options in asterisk.conf;
To maintain them forever was just too tough.

This patch removes:

* cdr_sqlite
* chan_gtalk
* chan_jingle
* chan_h323
* res_jabber
* app_saycountpl
* app_readfile
* app_dahdibarge

It removes the following applications/functions:

* WaitMusicOnHold
* SetMusicOnHold
* SIPCHANINFO

It removes the colon delimiter from the SIPPEER function.

Finally, it also removes all compatibility options that were configurable from
asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems.

Review: https://reviewboard.asterisk.org/r/3698/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      97834718
  23. Jul 03, 2014
  25. Jul 02, 2014
  27. Jul 01, 2014
  29. Jun 30, 2014
  31. Jun 27, 2014
  33. Jun 26, 2014
  35. Jun 25, 2014
  37. Jun 23, 2014
  39. Jun 20, 2014
  41. Jun 19, 2014
  43. Jun 17, 2014
  45. Jun 16, 2014
  47. Jun 15, 2014
  49. Jun 14, 2014