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20 authors
  1. May 06, 2015
  3. May 03, 2015
  5. Apr 30, 2015
    • Corey Farrell's avatar
      Sample Configs: Fix syntax error in pjsip.conf · 6b208d8c
      Corey Farrell authored 
      The sample pjsip.conf has a few comment lines that are missing the
semicolons at the start of the comment, causing the config to fail
load.

Change-Id: I776a38c916a7df7ee3e072fd0b21dbf4cc457352
      6b208d8c
    • Richard Mudgett's avatar
      chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option. · 03c51cf5
      Richard Mudgett authored 
      Some telco switches occasionally ignore ISDN RESTART requests.  The fix
for ASTERISK-19608 added an escape clause for B channels in the restarting
state if the telco ignores a RESTART request.  If the telco fails to
acknowledge the RESTART then Asterisk will assume the telco acknowledged
the RESTART on the second call attempt requesting the B channel by the
telco.  The escape clause is good for dealing with RESTART requests in
general but it does cause the next call for the restarting B channel to be
rejected if the telco insists the call must go on that B channel.

chan_dahdi doesn't really need to issue a RESTART request in response to
receiving a cause 44 (Requested channel not available) code.  Sending the
RESTART in such a situation is not required (nor prohibited) by the
standards.  I think chan_dahdi does this for historical reasons to deal
with buggy peers to get channels unstuck in a similar fashion as the
chan_dahdi.conf resetinterval option.

* Add the chan_dahdi.conf force_restart_unavailable_chans compatability
option that when disabled will prevent chan_dahdi from trying to RESTART
the channel in response to a cause 44 code.

ASTERISK-25034 #close
Reported by: Richard Mudgett

Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65
      03c51cf5
  7. Apr 29, 2015
  9. Apr 27, 2015
    • Corey Farrell's avatar
      Astobj2: Allow reference debugging to be enabled/disabled by config. · 5c1d07ba
      Corey Farrell authored 
      * The REF_DEBUG compiler flag no longer has any effect on code that uses
  Astobj2.  It is used to determine if reference debugging is enabled by
  default.  Reference debugging can be enabled or disabled in asterisk.conf.
* Caller information is provided in logger errors for ao2 bad magic numbers.
* Optimizes AO2 by merging internal functions with the public counterpart.
  This was possible now that we no longer require a dual ABI.

ASTERISK-24974 #close
Reported by: Corey Farrell

Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1
      5c1d07ba
    • Rodrigo Ramírez Norambuena's avatar
      cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version · 358080e8
      Rodrigo Ramírez Norambuena authored 
      Add new column to INSERT new columns added in cdr 1.8 version. The columns are:
 * peeraccount
 * linkedid
 * sequence
This feature is configurable in cdr_odbc.conf using a new configuration
option, 'newcdrcolumns'.

ASTERISK-24976 #close

Change-Id: Ibe0c7540a88305c6012786f438a0813ad8b19127
      358080e8
  11. Apr 16, 2015
    • George Joseph's avatar
      res_pjsip: Add global option to limit the maximum time for initial qualifies · c6ed6816
      George Joseph authored 
      
Currently when Asterisk starts initial qualifies of contacts are spread out
randomly between 0 and qualify_timeout to prevent network and system overload.
If a contact's qualify_frequency is 5 minutes however, that contact may be
unavailable to accept calls for the entire 5 minutes after startup.  So while
staggering the initial qualifies is a good idea, basing the time on
qualify_timeout could leave contacts unavailable for too long.

This patch adds a new global parameter "max_initial_qualify_time" that sets the
maximum time for the initial qualifies.  This way you could make sure that all
your contacts are initialy, randomly qualified within say 30 seconds but still
have the contact's ongoing qualifies at a 5 minute interval.

If max_initial_qualify_time is > 0, the formula is initial_interval =
min(max_initial_interval, qualify_timeout * random().  If not set,
qualify_timeout is used.

The default is "0" (disabled).

ASTERISK-24863 #close

Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4
Tested-by: default avatarGeorge Joseph <george.joseph@fairview5.com>
      c6ed6816
    • George Joseph's avatar
      pjsip_options: Add qualify_timeout processing and eventing · 51886c68
      George Joseph authored 
      This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html



The basic issues are that changes in contact status don't cause events to be
emitted for the associated endpoint.  Only dynamic contact add/delete actions
update the endpoint.  Also, the qualify timeout is fixed by pjsip at 32 seconds
which is a long time.

This patch makes use of the new transaction timeout feature in r4585 and
provides the following capabilities...

1.  A new aor/contact variable 'qualify_timeout' has been added that allows the
user to specify the maximum time in milliseconds to wait for a response to an
OPTIONS message.  The default is 3000ms.  When the timer expires, the contact is
marked unavailable.

2.  Contact status changes are now propagated up to the endpoint as follows...
When any contact is 'Available', the endpoint is marked as 'Reachable'.  When
all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'.  The
existing endpoint events are generated appropriately.

ASTERISK-24863 #close

Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
Tested-by: Dmitriy Serov
Tested-by: default avatarGeorge Joseph <george.joseph@fairview5.com>
      51886c68
  13. Apr 10, 2015
  15. Apr 08, 2015
  17. Mar 27, 2015
  19. Mar 25, 2015
  21. Mar 24, 2015
    • Richard Mudgett's avatar
      chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages. · 4c2fc5b8
      Richard Mudgett authored 
      Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens.  If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.

Consequences of these unnecessary messages:

* The caller can start hearing ringback before the far end even gets the
call.

* Many phones tend to grab the first connected line information and refuse
to update the display if it changes.  The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.

When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled.  When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.

* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages.  The default is "no" to disable sending the
unnecessary messages.

ASTERISK-24781 #close
Reported by: Richard Mudgett

Review: https://reviewboard.asterisk.org/r/4473/
........

Merged revisions 433338 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433339 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4c2fc5b8
  23. Mar 17, 2015
  25. Mar 14, 2015
  27. Mar 13, 2015
  29. Mar 09, 2015
  31. Feb 25, 2015
  33. Feb 15, 2015
  35. Feb 10, 2015
    • George Joseph's avatar
      res_pjsip_config_wizard: Add ability to auto-create hints. · 49161d8d
      George Joseph authored 
      Looking at the Super Awesome Company sample reminded me that creating hints is 
just plain gruntwork.  So you can now have the pjsip conifg wizard auto-create 
them for you.

Specifying 'hint_exten' in the wizard will create 
'exten => <hint_exten>,hint/PJSIP/<wizard_id>'
in whatever is specified for 'hint_context'.

Specifying 'hint_application' in the wizard will create
'exten => <hint_exten>,1,<hint_application>'
in whatever is specified for 'hint_context'.

The default for 'hint_context' is the endpoint's context.
There's no default for 'hint_application'.  If not specified, no app is added.
There's no default for 'hint_exten'.  If not specified, neither the hint itself 
nor the application will be created.

Some may think this is the slippery slope to users.conf but hints are a basic 
necessity for phones unlike voicemail, manager, etc that users.conf creates.

Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4383/
........

Merged revisions 431643 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      49161d8d
  37. Jan 30, 2015
  39. Jan 20, 2015
  41. Jan 16, 2015
  43. Jan 12, 2015
  45. Jan 09, 2015
  47. Dec 24, 2014
  49. Dec 22, 2014
    • Matthew Jordan's avatar
      app_confbridge: Add the ability to pass options/command to MixMonitor · b79a4a46
      Matthew Jordan authored 
      This patch adds the ability to pass options and a command to MixMontor when
recording a conference using ConfBridge.

New options are -

* record_options: Options to MixMontor, eg: m(), W() etc.
* record_command: The command to execute when recording is over.
* record_file_timestamp: Append the start time to the file name.

These options can also be used with the CONFBRIDGE function, e.g.,
Set(CONFBRIDGE(bridge,record_command)=/path/to/command ^{MIXMONITOR_FILENAME}))

Review: https://reviewboard.asterisk.org/r/4023

ASTERISK-24351 #close
Reported by: Gareth Palmer
patches:
  record_command-428838.patch uploaded by Gareth Palmer (License 5169)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429934 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      b79a4a46
  51. Dec 15, 2014
  53. Dec 01, 2014
    • Matthew Jordan's avatar
      main/stasis: Allow subscriptions to use a threadpool for message delivery · 1106e8fd
      Matthew Jordan authored 
      Prior to this patch, all Stasis subscriptions would receive a dedicated
thread for servicing published messages. In contrast, prior to r400178
(see review https://reviewboard.asterisk.org/r/2881/), the subscriptions
shared a thread pool. It was discovered during some initial work on Stasis
that, for a low subscription count with high message throughput, the
threadpool was not as performant as simply having a dedicated thread per
subscriber.

For situations where a subscriber receives a substantial number of messages
and is always present, the model of having a dedicated thread per subscriber
makes sense. While we still have plenty of subscriptions that would follow
this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into
the following two categories:
* Large number of subscriptions, specifically those tied to endpoints/peers.
* Low number of messages. Some subscriptions exist specifically to coordinate
  a single message - the subscription is created, a message is published, the
  delivery is synchronized, and the subscription is destroyed.
In both of the latter two cases, creating a dedicated thread is wasteful (and
in the case of a large number of peers/endpoints, harmful). In those cases,
having shared delivery threads is far more performant.

This patch adds the ability of a subscriber to Stasis to choose whether or not
their messages are dispatched on a dedicated thread or on a threadpool. The
threadpool is configurable through stasis.conf.

Review: https://reviewboard.asterisk.org/r/4193

ASTERISK-24533 #close
Reported by: xrobau
Tested by: xrobau
........

Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12
........

Merged revisions 428687 from http://svn.asterisk.org/svn/asterisk/branches/13


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@428688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      1106e8fd
  55. Nov 21, 2014
  57. Nov 19, 2014
  59. Nov 17, 2014
  61. Nov 15, 2014
  63. Nov 14, 2014
  65. Nov 05, 2014
  67. Nov 04, 2014
  69. Nov 03, 2014