Commit Graph

4165 Commits

Author SHA1 Message Date
Kinsey Moore
aa7f9e55f2 ARI: Remove channels/{channelId}/dial
This removes the /ari/channels/{channelId}/dial URI since it is
redundant, overly complex, is likely to become more externally complex
over time, and is too high-level compared with other ARI operations.
See the following for further information:
http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000002.html

(closes issue ASTERISK-22784)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2968/
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Merged revisions 402152 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402153 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 12:51:57 +00:00
Joshua Colp
fd98037fe2 res_ari_playback: Add missing 404 error response for GET and DELETE.
(closes issue ASTERISK-22722)
Reported by: Richard Mudgett
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Merged revisions 402139 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-29 11:15:59 +00:00
Richard Mudgett
8e4084b586 res_stasis.c: Made use the ao2_container callback templates.
* Made res_stasis.c use the OBJ_SEARCH_XXX defines.
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Merged revisions 402055 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-26 00:36:31 +00:00
Richard Mudgett
7eea4ab872 You'd think that new files would be free of whitespace issues. But you would be wrong.
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Merged revisions 402003 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 22:03:04 +00:00
Jonathan Rose
5c696bde67 ARI: channel/bridge recording errors when invalid format specified
Asterisk will now issue 422 if recording is requested against channels
or bridges with an unknown format

(closes issue ASTERISK-22626)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2939/
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Merged revisions 402001 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402002 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 22:01:43 +00:00
Jonathan Rose
d8a760307e ARI recordings: Issue HTTP failures for recording requests with file conflicts
If a file already exists in the recordings directory with the same name as what
we would record, issue a 422 instead of relying on the internal failure and
issuing success.

(closes issue ASTERISK-22623)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/2922/
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Merged revisions 401973 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401999 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 21:28:32 +00:00
Jonathan Rose
bb1568caa1 PJSIP: Add log messages when requests are received for non-existent endpoints
(closes issue ASTERISK-22552)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2934/
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Merged revisions 401938 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-25 17:41:38 +00:00
Jonathan Rose
d7bac6cf4b res_rtp_asterisk: Address jittery DTMF events in RTP streams
(closes issue ASTERISK-21170)
Reported by: NITESH BANSAL
Patches:
    dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
Review: https://reviewboard.asterisk.org/r/2938/
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Merged revisions 401619 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401620 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401621 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-23 17:56:44 +00:00
Matthew Jordan
f04a4328d8 res_rtp_asterisk: Fix crash when RTCP is not available during SSRC change
In r400089, a patch was put in to correct erroneous RTCP statistic resets.
Unfortunately, ast_rtp_read can be called on an RTP instance that does not
have RTCP information. This patch prevents that crash by only resetting
the statistics if we do actually have an RTCP instance.

(issue AST-1174)

(closes issue ASTERISK-22667)
Reported by: John Bigelow
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Merged revisions 401445 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401446 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401447 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22 23:10:22 +00:00
Richard Mudgett
8f03a463e1 res_parking: Give parking timeout comebacktoorigin channel DTMF features.
Parking timeouts did not set any DTMF features for the channel calling the
parker back.

* Added code to set the parkedcalltransfers, parkedcallreparking,
parkedcallhangup, and parkedcallrecording options appropriately for the
channels when a parking timeout occurs.  The recall channel DTMF options
are set using the BRIDGE_FEATURES channel variable to allow the other
timeout options to have the DTMF features available.

(closes issue ASTERISK-22630)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2942/
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Merged revisions 401422 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22 16:33:16 +00:00
Richard Mudgett
c17731620d res_parking: Update XML documention for DTMF features after parking timeout.
* Updated the XML documentation to indicate that the parkedcalltransfers,
parkedcallreparking, parkedcallhangup, and parkedcallrecording
configuration options also apply to parking timeouts.

(issue ASTERISK-22630)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2942/
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Merged revisions 401420 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-22 16:28:05 +00:00
David M. Lee
c87aae26f8 Fixed malformed Access-Control-Allow-Methods header. Was causing Safari to barf on POST and DELETE.
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Merged revisions 401106 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-21 18:59:22 +00:00
Joshua Colp
d183c6e134 Return a channel snapshot when originating using ARI, and subscribe the Stasis application to it.
This change allows a user of ARI to know what channel it has originated and also follow any
progress. If a Stasis application is provided it will be automatically subscribed to the
originated channel immediately.

(closes issue ASTERISK-22485)
Reported by: David Lee

Review: https://reviewboard.asterisk.org/r/2910/
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Merged revisions 401281 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401282 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-19 14:45:14 +00:00
Richard Mudgett
a80a6a7631 res_parking: Remove setting useless flag.
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Merged revisions 401271 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18 22:52:35 +00:00
Richard Mudgett
057d105c5a Add channel lock protection around translation path setup.
Most callers of ast_channel_make_compatible() happen before the channels
enter a two party bridge.  With the new bridging framework, two party
bridging technologies may also call ast_channel_make_compatible() when
there is more than one thread involved with the two channels.

* Added channel lock protection in set_format() and
ast_channel_make_compatible_helper() when dealing with the channel's
native formats while setting up a translation path.

* Fixed best_src_fmt and best_dst_fmt usage consistency in
ast_channel_make_compatible_helper().  The call to
ast_translator_best_choice() got them backwards.

* Updated some callers of ast_channel_make_compatible() and the function
documentation.  There is actually a difference between the two channels
passed in.

* Fixed the deadlock potential in res_fax.c dealing with
ast_channel_make_compatible().  The deadlock potential was already there
anyway because res_fax called ast_channel_make_compatible() with chan
locked.

(closes issue ASTERISK-22542)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2915/
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Merged revisions 401239 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401240 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-18 16:59:09 +00:00
Jonathan Rose
9a45173715 res_parking: Fix bug where reloading immediately wipes new parkpos extensions
(closes issue ASTERISK-22631)
Reported by: Kevin Harwell
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Merged revisions 401158 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401159 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-17 18:25:35 +00:00
Kinsey Moore
4cdd02ce26 Reduce log level of a non-pubsub error message
Drop an error log message to debug level 1 since distributed device
state functions correctly when receiving this message and it spams the
logs.

(closes issue ASTERISK-22410)
Reported by: abelbeck
Patches:
    asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
    asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
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Merged revisions 401119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 401120 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 401121 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-17 15:41:22 +00:00
Richard Mudgett
868a76f897 ARI: Fix crash when POST /playback/{id}/control does not have an operation parameter.
(closes issue ASTERISK-22680)
Reported by: John Bigelow
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Merged revisions 401107 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-16 21:22:25 +00:00
Kinsey Moore
2fc0a8873c Clarify documentation for channel and bridge list
This makes it clear that the ARI API calls for listing channels and
bridges will list all channels or bridges in the system and not just
those that are in or are controlled by a Stasis application.

(closes issue ASTERISK-22635)
Reported by: Kevin Harwell
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Merged revisions 401087 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401088 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-16 14:02:06 +00:00
Paul Belanger
7955e89d06 Use POST / DELETE to toggle ARI bridge moh
Review: https://reviewboard.asterisk.org/r/2911/
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Merged revisions 401040 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401041 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-16 00:12:36 +00:00
Kinsey Moore
ec83706003 Ensure bridge record error responses validate
This adds the list of expected errors to the /bridges/{bridgeId}/record
ARI documentation so that outbound 4xx errors validate properly.
Previously, this would result in a response validation failure.

(closes issue ASTERISK-22627)
Reported by: Joshua Colp
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Merged revisions 401018 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-15 20:03:19 +00:00
Paul Belanger
6072e043cf Use POST / DELETE to toggle hold / moh for ARI channels
This change updates how we handle toggle events, rather then create two
different function names, we'll just use POST / DELETE from HTTP to handle it.

Review: https://reviewboard.asterisk.org/r/2906/
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Merged revisions 400999 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-15 15:30:39 +00:00
Kevin Harwell
61b60fab2d pjsip outbound registration: Log message says received a 408 when we didn't
If the server didn't exist that we are trying to register to the log message
would say that a 408 was received from that server when in reality one wasn't.
Added log messages stating no response was received if the response does not
exist.

(closes issue ASTERISK-22554)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/2893/
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Merged revisions 400890 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400891 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-14 15:54:06 +00:00
Matthew Jordan
fd4919e466 Remove duplicate module info block
The module info block was repeated twice. Once is sufficient.
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Merged revisions 400881 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-14 15:01:59 +00:00
Joshua Colp
b47851264e Fix a race condition in res_pjsip_session with rapidly terminating the session.
The INVITE session state callback wrongly assumes that a session will always exist, but
when rapidly terminating the session this assumption goes out the window. As all handler
code for the INVITE session state callback requires the session it will now just exit
immediately if no session exists.

(closes issue ASTERISK-22668)
Reported by: John Bigelow
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Merged revisions 400872 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-13 15:42:20 +00:00
Kinsey Moore
1a0a2b3e4c Fix realm comparison for outbound auth
When generating the list of authentication credentials to pass to
PJSIP, Asterisk was using the raw pointer of a pj_str_t which is not
always NULL-terminated. This sometimes resulted in incorrect text for
the realm and a failure to match the realm for authentication purposes
which was causing the outbound nominal auth pjsip basic call test to
bounce. This now uses the pj_str_t that contains the realm instead of
generating a new one. Thanks to John Bigelow for helping to narrow this
down.
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Merged revisions 400863 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-12 16:53:06 +00:00
David M. Lee
9234804a3b Multiple revisions 400508,400842-400843,400848
........
  r400508 | dlee | 2013-10-03 23:54:51 -0500 (Thu, 03 Oct 2013) | 1 line
  
  Corrected response class for stopPlayback
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  r400842 | dlee | 2013-10-10 14:23:24 -0500 (Thu, 10 Oct 2013) | 1 line
  
  Correct some ARI wiki rendering errors
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  r400843 | dlee | 2013-10-10 14:26:19 -0500 (Thu, 10 Oct 2013) | 1 line
  
  Updated /play resource docs. The playback of http: resources isn't implemented... yet
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  r400848 | dlee | 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) | 5 lines
  
  Fix a stupid copy/paste error in ARI docs.
  
  Patches:
      ari-doc-patch.txt uploaded by jbigelow (license 5091)
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Merged revisions 400508,400842-400843,400848 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-11 16:36:00 +00:00
Joshua Colp
113c281782 Perform validation of permanent contacts on AORs in res_pjsip.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-10 18:21:55 +00:00
Joshua Colp
cbbcf1808c Fix an assertion in res_pjsip when specifying an invalid outbound proxy.
This change fixes two issues when setting an outbound proxy:

1. The outbound proxy URI was not parsed and validated during configuration.
2. If an outgoing dialog was created and the outbound proxy could not be set an assertion would
occur because the usage count on the dialog was not decremented.

The documentation has also been updated to specify that a full URI must be specified for
the outbound proxy.

(closes issue ASTERISK-22672)
Reported by: Antti Yrjola
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Merged revisions 400824 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-10 12:26:20 +00:00
Matthew Jordan
fcf6c84666 Use 'z' as the format specifier for size_t
Using 'lu' will produce a compiler warning for some versions of gcc and on some
architectures. 'z' should be portable as a format specifier for size_t.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400813 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-09 11:02:04 +00:00
Matthew Jordan
b36ef0b412 Add PJSIP_HEADER function for manipulation of SIP headers in the PJSIP stack
This patch adds support to the PJSIP stack in Asterisk for SIP header
manipulation. Note that this is analagous to SIPAddHeader/SIPRemoveHeader.

For PJSIP_HEADER, an incoming supplemental session callback is registered that
takes the pjsip_hdrs from the incoming session and stores them in a linked
list in the session datastore.  Calls to PJSIP_HEADER traverse over the list
and return the nth matching header where 'n' is the 'number' argument to the
function.

When adding a header, the first call creates a datastore and linked list and
adds the datastore to the session.  The header is then created as a pjsip_hdr
and added to the list.  An outgoing supplemental session callback then
traverses the list and adds the headers to the outgoing pjsip_msg.

When removing a header, the list created with PJSIP_HEADER(add,...) is
traversed and all matching entries are removed.

(closes issue ASTERISK-22498)
Reported by: George Joseph
patch:
  res_pjsip_header_funcs_v1.patch uploaded by george.joseph (License 6322)
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Merged revisions 400771 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-08 22:59:32 +00:00
Mark Michelson
2904a198d5 Switch from using pjsip_strerror to pj_strerror.
pjsip_strerror is only aware of PJSIP-specific error
codes. pj_strerror() is aware of all PJProject error
codes and OS-specific error codes.

This specifically fixes an oft-seen error in transport
configuration code where EADDRINUSE would result in
"Unknown PJSIP error 120098" instead of a useful
message.
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Merged revisions 400749 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-08 20:52:04 +00:00
Kinsey Moore
4873c11f64 Fix STUN crash when using IPv6 any address
Ensure that when chan_sip binds to the IPv6 any address ([::]), IPv4
candidates are also added.

(closes issue ASTERISK-21917)
Reported by: Torrey Searle
Patches:
    0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License 5334)
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Merged revisions 400681 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400682 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-08 15:46:16 +00:00
Mark Michelson
a4d6f43962 Push CLI qualify into the threadpool.
If you run Asterisk in the background and then connect to
it through a separate console, the thread that runs CLI commands
is not registered with PJLIB. Thus PJLIB does not like it when
you attempt to send OPTIONS requests from that thread. So now
we push the task into the threadpool, which we know to be registered
with PJLIB.

Thanks to Antti Yrjola for reporting this.
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Merged revisions 400680 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-08 15:44:47 +00:00
Richard Mudgett
8eec8fbf83 Make app_queue and res_agi independent of AMI being enabled.
The https://reviewboard.asterisk.org/r/2888/ review changes manager to not
subscribe to stasis when it is disabled for performance reasons.  When
manager is disabled app_queue and res_agi decline to load and fail to
clean up what they have already allocated.

* Made app_queue and res_agi clean up allocated resources when they
decline to load.

* Made app_queue and res_agi use their own subscriptions to the stasis
topics instead of borrowing manager's message router structure
inappropriately.

(closes issue ASTERISK-22604)
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/2902/
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Merged revisions 400671 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-08 15:12:46 +00:00
Jonathan Rose
f4ebdca52a chan_pjsip: Make logger togglable without loading/unloading
This patch makes the res_pjsip_logger do a few things... First, it
will be built and installed by default now, so end users won't need
to enable it in menuselect. Second, while it is loaded, it no longer
will immediately issue log messages. Upon loading, it is in the
disabled state and must be turned on with the new CLI command. The
CLI command 'pjsip set logger <on/off/host> has been added and can be
used to do the following:
pjsip set logger on:
    Enables logger for all PJSIP traffic
pjsip set logger off:
    Disables logger for all PJSIP traffic
pjsip set logger host <host>:
    Enables logger for the specific host

Review: https://reviewboard.asterisk.org/r/2900/
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2013-10-04 19:11:38 +00:00
Matthew Jordan
8d7873b836 ARI: Add subscription support
This patch adds an /applications API to ARI, allowing explicit management of
Stasis applications.

 * GET /applications - list current applications
 * GET /applications/{applicationName} - get details of a specific application
 * POST /applications/{applicationName}/subscription - explicitly subscribe to
   a channel, bridge or endpoint
 * DELETE /applications/{applicationName}/subscription - explicitly unsubscribe
   from a channel, bridge or endpoint

Subscriptions work by a reference counting mechanism: if you subscript to an
event source X number of times, you must unsubscribe X number of times to stop
receiveing events for that event source.

Review: https://reviewboard.asterisk.org/r/2862

(issue ASTERISK-22451)
Reported by: Matt Jordan
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2013-10-04 16:01:48 +00:00
Joshua Colp
b52c972b17 Enclose the To URI and update its user portion if a request user has been specified.
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2013-10-04 15:49:39 +00:00
Joshua Colp
47da03e737 Replace the connection address at the SDP level if altering the SDP with the external media address.
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2013-10-04 14:55:22 +00:00
Joshua Colp
bd8cf10d60 Fix a crash caused by muting and unmuting a channel in ARI without specifying a direction.
(closes issue ASTERISK-22637)
Reported by: Scott Griepentrog

Patch by Matt Jordan, whose office I have taken over in the name of Canada.
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2013-10-03 19:06:48 +00:00
Kinsey Moore
c3b1d8f8c1 res_rtp_multicast: Ensure SSRC is set properly
This fixes a bug where the SSRC field on multicast RTP can be stuck at
0 which can cause problems for endpoints trying to make sense of
incoming streams.

(closes issue ASTERISK-22567)
Reported by: Simone Camporeale
Patches:
    22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale (License 6536)
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2013-10-03 18:32:59 +00:00
Mark Michelson
ee21eee7e0 Cache string values of formats on ast_format_cap() to save processing.
Channel snapshots have string representations of the channel's native formats.
Prior to this change, the format strings were re-created on ever channel snapshot
creation. Since channel native formats rarely change, this was very wasteful.
Now, string representations of formats may optionally be stored on the ast_format_cap
for cases where string representations may be requested frequently. When formats
are altered, the string cache is marked as invalid. When strings are requested, the
cache validity is checked. If the cache is valid, then the cached strings are copied.
If the cache is invalid, then the string cache is rebuilt and copied, and the cache
is marked as being valid again.

Review: https://reviewboard.asterisk.org/r/2879
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2013-10-03 14:58:16 +00:00
Joshua Colp
c977f73e13 Fix crashes in res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected and external_media_address is set.
The callback function for changing the media address in streams wrongly assumes that a connection line
will always be present. This is false as no line is present if a stream has been rejected.

(closes issue ASTERISK-22645)
Reported by: Rusty Newton
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2013-10-03 14:52:24 +00:00
Mark Michelson
addbf276f5 Multiple revisions 400318-400319
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  r400318 | mmichelson | 2013-10-02 17:08:49 -0500 (Wed, 02 Oct 2013) | 12 lines
  
  Remove unnecessary waits from stasis.
  
  Since caches are updated on publisher threads, there is no need
  to wait for the cache updates to occur after a stasis message
  is published.
  
  In the case of chan_pjsip device state changes, this set of
  changes caused an improvement to performance.
  
  Review: https://reviewboard.asterisk.org/r/2890
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  r400319 | mmichelson | 2013-10-02 17:10:54 -0500 (Wed, 02 Oct 2013) | 3 lines
  
  Remove svn:mergeinfo property.
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2013-10-02 22:22:17 +00:00
Joshua Colp
51a9b93a14 Fix a crash in res_pjsip_t38 caused by the wrong assumption that a session will always have a channel.
When starting up or shutting down this assumption is false.
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2013-10-02 18:44:48 +00:00
Richard Mudgett
97fcd6366d MALLOC_DEBUG: Fix some misuses of free() when MALLOC_DEBUG is enabled.
* There were several places in ARI where an external library was mallocing
memory that must always be released with free().  When MALLOC_DEBUG is
enabled, free() is redirected to the MALLOC_DEBUG version.  Since the
external library call still uses the normal malloc(), MALLOC_DEBUG
complains that the freed memory block is not registered and will not free
it.  These cases must use ast_std_free().

* Changed calls to asprintf() and vasprintf() to the equivalent
ast_asprintf() and ast_vasprintf() versions respectively.
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2013-10-02 17:12:49 +00:00
Joshua Colp
424c0f2eb7 Fix a random one way audio issue in PJSIP.
Due to the asynchronous design of the PJMEDIA SDP negotiator it was possible for
the SDP to be negotiated *after* a channel was created and after it was being wait
on by an application. It is only after negotiation occurs that the file descriptors
for RTP are placed on the channel. Since the channel was already being waited on
these file descriptors were not monitored, causing incoming media to never be read.

This change wakes up any application waiting on the channel so that added file
descriptors end up being monitored.

(closes issue AST-1227)
Reported by: John Bigelow
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2013-10-02 15:33:56 +00:00
Joshua Colp
83d55f85ec Allow specifying a channel to dial an extension and context in an ARI dial operation.
(issue ASTERISK-22625)
Reported by: Scott Griepentrog
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2013-10-02 15:24:16 +00:00
Joshua Colp
c33aac75e4 Retrieve and store the hostname only once so multiple threads do not potentially initialize it at the same time.
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2013-10-02 14:13:55 +00:00
David M. Lee
2de42c2a25 Multiple revisions 399887,400138,400178,400180-400181
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  r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line
  
  Minor performance bump by not allocate manager variable struct if we don't need it
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  r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines
  
  Stasis performance improvements
  
  This patch addresses several performance problems that were found in
  the initial performance testing of Asterisk 12.
  
  The Stasis dispatch object was allocated as an AO2 object, even though
  it has a very confined lifecycle. This was replaced with a straight
  ast_malloc().
  
  The Stasis message router was spending an inordinate amount of time
  searching hash tables. In this case, most of our routers had 6 or
  fewer routes in them to begin with. This was replaced with an array
  that's searched linearly for the route.
  
  We more heavily rely on AO2 objects in Asterisk 12, and the memset()
  in ao2_ref() actually became noticeable on the profile. This was
  #ifdef'ed to only run when AO2_DEBUG was enabled.
  
  After being misled by an erroneous comment in taskprocessor.c during
  profiling, the wrong comment was removed.
  
  Review: https://reviewboard.asterisk.org/r/2873/
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  r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines
  
  Taskprocessor optimization; switch Stasis to use taskprocessors
  
  This patch optimizes taskprocessor to use a semaphore for signaling,
  which the OS can do a better job at managing contention and waiting
  that we can with a mutex and condition.
  
  The taskprocessor execution was also slightly optimized to reduce the
  number of locks taken.
  
  The only observable difference in the taskprocessor implementation is
  that when the final reference to the taskprocessor goes away, it will
  execute all tasks to completion instead of discarding the unexecuted
  tasks.
  
  For systems where unnamed semaphores are not supported, a really
  simple semaphore implementation is provided. (Which gives identical
  performance as the original taskprocessor implementation).
  
  The way we ended up implementing Stasis caused the threadpool to be a
  burden instead of a boost to performance. This was switched to just
  use taskprocessors directly for subscriptions.
  
  Review: https://reviewboard.asterisk.org/r/2881/
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  r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines
  
  Optimize how Stasis forwards are dispatched
  
  This patch optimizes how forwards are dispatched in Stasis.
  
  Originally, forwards were dispatched as subscriptions that are invoked
  on the publishing thread. This did not account for the vast number of
  forwards we would end up having in the system, and the amount of work it
  would take to walk though the forward subscriptions.
  
  This patch modifies Stasis so that rather than walking the tree of
  forwards on every dispatch, when forwards and subscriptions are changed,
  the subscriber list for every topic in the tree is changed.
  
  This has a couple of benefits. First, this reduces the workload of
  dispatching messages. It also reduces contention when dispatching to
  different topics that happen to forward to the same aggregation topic
  (as happens with all of the channel, bridge and endpoint topics).
  
  Since forwards are no longer subscriptions, the bulk of this patch is
  simply changing stasis_subscription objects to stasis_forward objects
  (which, admittedly, I should have done in the first place.)
  
  Since this required me to yet again put in a growing array, I finally
  abstracted that out into a set of ast_vector macros in
  asterisk/vector.h.
  
  Review: https://reviewboard.asterisk.org/r/2883/
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  r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines
  
  Remove dispatch object allocation from Stasis publishing
  
  While looking for areas for performance improvement, I realized that an
  unused feature in Stasis was negatively impacting performance.
  
  When a message is sent to a subscriber, a dispatch object is allocated
  for the dispatch, containing the topic the message was published to, the
  subscriber the message is being sent to, and the message itself.
  
  The topic is actually unused by any subscriber in Asterisk today. And
  the subscriber is associated with the taskprocessor the message is being
  dispatched to.
  
  First, this patch removes the unused topic parameter from Stasis
  subscription callbacks.
  
  Second, this patch introduces the concept of taskprocessor local data,
  data that may be set on a taskprocessor and provided along with the data
  pointer when a task is pushed using the ast_taskprocessor_push_local()
  call. This allows the task to have both data specific to that
  taskprocessor, in addition to data specific to that invocation.
  
  With those two changes, the dispatch object can be removed completely,
  and the message is simply refcounted and sent directly to the
  taskprocessor.
  
  Review: https://reviewboard.asterisk.org/r/2884/
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2013-09-30 18:55:27 +00:00