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20 authors
  1. Jul 22, 2014
    • Michael L. Young's avatar
      core/db: Improve I/O When Updating Rows · b4a68168
      Michael L. Young authored 
      When updating a row, we are currently doing an INSERT OR REPLACE INTO.  The
downside to this is that the row is deleted if it exists and then a new row is
inserted.  So, we are hitting the disk twice.  One for the deletion and one for
the insertion.

This patch changes this statement to an INSERT INTO and if the insert fails
because a row with that key exists, we will IGNORE the failure.  Then we will
attempt to perform an UPDATE on the existing row if that row wasn't just
INSERTed.

ASTERISK-24050 #close
Reported by: Michael L. Young
patches:
  astdb-insert-update-io-help_trunk_v2.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/3815/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      b4a68168
    • 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
  3. Jul 21, 2014
  5. Jul 20, 2014
  7. Jul 18, 2014
  9. Jul 17, 2014
  11. Jul 16, 2014
  13. Jul 14, 2014
  15. Jul 13, 2014
  17. Jul 11, 2014
  19. Jul 10, 2014
  21. Jul 09, 2014
  23. Jul 08, 2014
  25. 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
  27. 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
  29. Jul 03, 2014
  31. Jun 30, 2014
  33. Jun 27, 2014
  35. Jun 26, 2014
    • Kinsey Moore's avatar
      Bridging: Allow channels to define bridging hooks · e977b793
      Kinsey Moore authored 
      This patch allows the current owner of a channel to define various
feature hooks to be made available once the channel has entered a
bridge. This includes any hooks that are setup on the
ast_bridge_features struct such as DTMF hooks, bridge event hooks
(join, leave, etc.), and interval hooks.

Review: https://reviewboard.asterisk.org/r/3649/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e977b793
    • Matthew Jordan's avatar
      res_http_websocket: Close websocket correctly and use careful fwrite · 365ae752
      Matthew Jordan authored 
      When a client takes a long time to process information received from Asterisk,
a write operation using fwrite may fail to write all information. This causes
the underlying file stream to be in an unknown state, such that the socket
must be disconnected. Unfortunately, there are two problems with this in
Asterisk's existing websocket code:
1. Periodically, during the read loop, Asterisk must write to the connected
   websocket to respond to pings. As such, Asterisk maintains a reference to
   the session during the loop. When ast_http_websocket_write fails, it may
   cause the session to decrement its ref count, but this in and of itself
   does not break the read loop. The read loop's write, on the other hand,
   does not break the loop if it fails. This causes the socket to get in a
   'stuck' state, preventing the client from reconnecting to the server.
2. More importantly, however, is that the fwrite in ast_http_websocket_write
   fails with a large volume of data when the client takes awhile to process
   the information. When it does fail, it fails writing only a portion of
   the bytes. With some debugging, it was shown that this was failing in a
   similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite
   with a long enough timeout solved the problem.

Note that this version of the patch, unlike r417310 in Asterisk 11, exposes
configuration options beyond just chan_sip's sip.conf. Configuration options
to configure the write timeout have also been added to pjsip.conf and ari.conf.

#ASTERISK-23917 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3624/
........

Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      365ae752
  37. Jun 25, 2014
  39. Jun 24, 2014
  41. Jun 22, 2014
  43. Jun 20, 2014
  45. Jun 19, 2014
  47. Jun 16, 2014
    • Kevin Harwell's avatar
      res_http_websocket: read/write string fixup · bd0aa4fb
      Kevin Harwell authored 
      There was a problem when reading a string from the websocket. It assumed the
received data had a null terminator and tried to write the data to an ast_str.
This of course could/would read past the end of the given buffer while
writing the data to the internal buffer of ast_str. Modified the the code to
correctly place a null terminator on the result string.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      bd0aa4fb
  49. Jun 13, 2014
    • Matthew Jordan's avatar
      stasis: Reduce creation of channel snapshots to improve performance · 9cc1a8e8
      Matthew Jordan authored 
      During some performance testing of Asterisk with AGI, ARI, and lots of Local
channels, we noticed that there's quite a hit in performance during channel
creation and releasing to the dialplan (ARI continue). After investigating
the performance spike that occurs during channel creation, we discovered
that we create a lot of channel snapshots that are technically unnecessary.
This includes creating snapshots during:
 * AGI execution
 * Returning objects for ARI commands
 * During some Local channel operations
 * During some dialling operations
 * During variable setting
 * During some bridging operations
And more.

This patch does the following:
 - It removes a number of fields from channel snapshots. These fields were
   rarely used, were expensive to have on the snapshot, and hurt performance.
   This included formats, translation paths, Log Call ID, callgroup, pickup
   group, and all channel variables. As a result, AMI Status,
   "core show channel", "core show channelvar", and "pjsip show channel" were
   modified to either hit the live channel or not show certain pieces of data.
   While this is unfortunate, the performance gain from this patch is worth
   the loss in behaviour.
 - It adds a mechanism to publish a cached snapshot + blob. A large number of
   publications were changed to use this, including:
   - During Dial begin
   - During Variable assignment (if no AMI variables are emitted - if AMI
     variables are set, we have to make snapshots when a variable is changed)
   - During channel pickup
   - When a channel is put on hold/unhold
   - When a DTMF digit is begun/ended
   - When creating a bridge snapshot
   - When an AOC event is raised
   - During Local channel optimization/Local bridging
   - When endpoint snapshots are generated
   - All AGI events
   - All ARI responses that return a channel
   - Events in the AgentPool, MeetMe, and some in Queue
 - Additionally, some extraneous channel snapshots were being made that were
   unnecessary. These were removed.
 - The result of ast_hashtab_hash_string is now cached in stasis_cache. This
   reduces a large number of calls to ast_hashtab_hash_string, which reduced
   the amount of time spent in this function in gprof by around 50%.

#ASTERISK-23811 #close
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/3568/
........

Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      9cc1a8e8
    • Richard Mudgett's avatar
      AST-2014-007: Fix of fix to allow AMI and SIP TCP to send messages. · 13e697f8
      Richard Mudgett authored 
      ASTERISK-23673 #close
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/3617/
........

Merged revisions 416066 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 416067 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 416070 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      13e697f8
  51. Jun 12, 2014
    • Richard Mudgett's avatar
      AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. · 4ca5745d
      Richard Mudgett authored 
      Simply establishing a TCP connection and never sending anything to the
configured HTTP port in http.conf will tie up a HTTP connection.  Since
there is a maximum number of open HTTP sessions allowed at a time you can
block legitimate connections.

A similar problem exists if a HTTP request is started but never finished.

* Added http.conf session_inactivity timer option to close HTTP
connections that aren't doing anything.  Defaults to 30000 ms.

* Removed the undocumented manager.conf block-sockets option.  It
interferes with TCP/TLS inactivity timeouts.

* AMI and SIP TLS connections now have better authentication timeout
protection.  Though I didn't remove the bizzare TLS timeout polling code
from chan_sip.

* chan_sip can now handle SSL certificate renegotiations in the middle of
a session.  It couldn't do that before because the socket was non-blocking
and the SSL calls were not restarted as documented by the OpenSSL
documentation.

* Fixed an off nominal leak of the ssl struct in
handle_tcptls_connection() if the FILE stream failed to open and the SSL
certificate negotiations failed.

The patch creates a custom FILE stream handler to give the created FILE
streams inactivity timeout and timeout after a specific moment in time
capability.  This approach eliminates the need for code using the FILE
stream to be redesigned to deal with the timeouts.

This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of
the SSL_read/SSL_write operations.

ASTERISK-23673 #close
Reported by: Richard Mudgett
........

Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4ca5745d
    • Joshua Colp's avatar
      res_pjsip_pubsub: Persist subscriptions in sorcery so they are recreated on startup. · 58f4c18a
      Joshua Colp authored 
      This change makes res_pjsip_pubsub persist inbound subscriptions in sorcery. By default
this uses the local astdb but it can also be configured to store within an outside
database. When Asterisk is started these subscriptions are recreated if they have not
expired. Notifications are sent to the devices which have subscribed and they are none
the wiser that the system has restarted.

Review: https://reviewboard.asterisk.org/r/3598/
........

Merged revisions 415766 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415767 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      58f4c18a