Commit Graph

2983 Commits

Author SHA1 Message Date
Kevin Harwell
f6593b4156 ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404
Was returning a 404 on a valid technology with an empty list of endpoints.
Now checking against the channel tech to make sure the tech itself is valid
and not just an empty list of endpoints.

(issue ASTERISK-22803)
Reported by: David M. Lee
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Merged revisions 402793 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-12 23:17:45 +00:00
Kevin Harwell
4be01b4598 ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404
Implementation listing endpoints by technology returned an empty array if no
matching endpoints were found.  Fixed so a "404 Not Found" will be returned
instead.

(closes issue ASTERISK-22803)
Reported by: David M. Lee
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Merged revisions 402787 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-12 22:17:28 +00:00
Kevin Harwell
12a0edac69 pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer dereferences
Both res_pjsip_messaging and res_pjsip_header_funcs were causing asterisk to
crash because they were trying to dereference a NULL pointer.

In the case of res_pjsip_messaging it was attempting to "print" a contact
header that did not exist.  In fact contact headers should not be part of
a SIP MESSAGE, so the offending code was simply removed.

In the case of res_pjsip_header_funcs a null private channel tech was being
passed to the function and then later dereferenced.  Added null checks (and
error logging) to the read/write function handlers to guard against crashing.

(closes issue ASTERISK-22821)
Reported by: Anthony Messina
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Merged revisions 402757 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-12 16:49:17 +00:00
David M. Lee
5701c51bb6 Fixed a typ.
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2013-11-12 15:27:00 +00:00
Richard Mudgett
7323809607 res_stasis.c: Fix locking issues with the app_bridge_moh container.
* Fix unlinking from the app_bridges_moh container in remove_bridge_moh()
without a lock under normal circumstances.

* Made check ast_bridge_set_after_callback() return value in
bridge_moh_create() to handle failure.

* Fixed SCOPED_AO2LOCK() locking over too much scope in
stasis_app_bridge_moh_channel() and stasis_app_bridge_moh_stop().

* Fixed unusual usage of ao2_unlink_flag() in control_unlink().

* Fixed orphaned bridge from off nominal path in
stasis_app_bridge_create().

* Fixed strange construct in stasis_app_unsubscribe().  From a bad merge?

* Made load_module() cleanup on failure.

Review: https://reviewboard.asterisk.org/r/2962/
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Merged revisions 402593 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-08 20:37:08 +00:00
Mark Michelson
518f091a1a Clarify an ambiguous error message.
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Merged revisions 402582 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-08 19:22:53 +00:00
David M. Lee
4c128198c8 res_pjsip: Print a helpful error message if sorcery registration fails
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Merged revisions 402570 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-08 18:53:14 +00:00
David M. Lee
b83a3965b8 Changes from make ari-stubs after r402560
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Merged revisions 402561 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-08 18:52:19 +00:00
Kevin Harwell
4f1bdeed1c ARI playback: Rename ARI Playback to Playbacks
Before playback was the only non plural resource.  It has been renamed to
playbacks for consistency.

(closes issue ASTERISK-22737)
Reported by: Paul Belanger
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Merged revisions 402560 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-08 17:59:16 +00:00
David M. Lee
97a8debd90 ari: Add application/x-www-form-urlencoded parameter support
ARI POST calls only accept parameters via the URL's query string.
While this works, it's atypical for HTTP API's in general, and
specifically frowned upon with RESTful API's.

This patch adds parsing for application/x-www-form-urlencoded request
bodies if they are sent in with the request. Any variables parsed this
way are prepended to the variable list supplied by the query string.

(closes issue ASTERISK-22743)
Review: https://reviewboard.asterisk.org/r/2986/
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Merged revisions 402555 from http://svn.asterisk.org/svn/asterisk/branches/12


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2013-11-08 17:29:53 +00:00
Jonathan Rose
3c645e8520 PJSIP: Improve error handling in digest authenticator
Previously, regardless of whether failure to authenticate was due to
lacking any authentication or actually failing authentication, the
Digest Authenticator would simply return that a challenge was still
needed. It will continue to do that when no authentication information
is in the received SIP digest, but when authentication information
is present and does not pass authentication, that will be treated as
an authentication error. This is to ensure that PJSIP will issue
security events indicated failed auths.
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2013-11-07 23:42:31 +00:00
David M. Lee
7d0d1a1efb ari: User better nicknames for ARI operations
While working on building client libraries from the Swagger API, I
noticed a problem with the nicknames.

    channel.deleteChannel()
    channel.answerChannel()
    channel.muteChannel()

Etc. We put the object name in the nickname (since we were generating C
code), but it makes OO generators redundant.

This patch makes the nicknames more OO friendly. This resulted in a lot
of name changing within the res_ari_*.so modules, but not much else.

There were a couple of other fixed I made in the process.

 * When reversible operations (POST /hold, POST /unhold) were made more
   RESTful (POST /hold, DELETE /unhold), the path for the second operation
   was left in the API declaration. This worked, but really the two
   operations should have been on the same API.
 * The POST /unmute operation had still not been REST-ified.

Review: https://reviewboard.asterisk.org/r/2940/
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2013-11-07 21:10:31 +00:00
Joshua Colp
7678fd040e res_ari_channels: Add ring operation, dtmf operation, hangup reasons, and tweak early media.
The ring operation sends ringing to the specified channel it is invoked on.
The dtmf operation can be used to send DTMF digits to the specified channel
of a specific length with a wait time in between. Finally hangup reasons
allow you to specify why a channel is being hung up (busy, congestion).

Early media behavior has also been tweaked slightly. When playing media to a channel
it will no longer automatically answer. If it has not been answered a progress indication
is sent instead.

(closes issue ASTERISK-22701)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2916/
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2013-11-01 14:38:21 +00:00
Kinsey Moore
98dea21bc1 chan_sip: Fix RTCP port for SRFLX ICE candidates
This corrects one-way audio between Asterisk and Chrome/jssip as a
result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX
ICE candidates. This also exposes an ICE component enumeration to
extract further details from candidates.

(closes issue ASTERISK-21383)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/2967/
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Merged revisions 402345 from http://svn.asterisk.org/svn/asterisk/branches/11
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2013-11-01 12:40:40 +00:00
Joshua Colp
4053f36a71 res_ari_channels: Fix a deadlock when originating multiple channels close to eachother.
If a Stasis application is specified an implicit subscription is done on the originated
channel. This was previously done with the channel lock held which is dangerous as the
underlying code locks the container and iterates items. This change releases the lock
on the originated channel before subscribing occurs.

(closes issue ASTERISK-22768)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/2979/
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2013-11-01 12:33:09 +00:00
Joshua Colp
d17a780333 res_stasis: Ensure the channel is always departed from the bridge when it leaves.
This change adds a command to the command queue to explicitly depart the channel
from the bridge when it is told it has left. If the channel has already been departed
or has entered a different bridge this command will become a no-op.

(closes issue ASTERISK-22703)
Reported by: John Bigelow

(closes issue ASTERISK-22634)
Reported by: Kevin Harwell

Review: https://reviewboard.asterisk.org/r/2965/
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2013-11-01 12:13:09 +00:00
David M. Lee
069da1e75a stasis: add functions embarrassingly missing from r400522
I neglected to implement two of the endpoint subscription functions when
I did the work. Normally, you'll only hit that when you unsubscribe from
a specific endpoint.
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2013-10-31 14:45:03 +00:00
Kevin Harwell
4746c068dc pjsip_messaging: Added debug for in dialog messaging
(issue ASTERISK-22777)
Reported by: Matt Jordan
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2013-10-30 17:54:26 +00:00
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


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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


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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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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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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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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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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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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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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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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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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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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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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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2013-10-19 14:45:14 +00:00
Richard Mudgett
a80a6a7631 res_parking: Remove setting useless flag.
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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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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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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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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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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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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


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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


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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


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2013-10-12 16:53:06 +00:00
David M. Lee
9234804a3b Multiple revisions 400508,400842-400843,400848
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  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


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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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Merged revisions 400833 from http://svn.asterisk.org/svn/asterisk/branches/12


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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


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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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Merged revisions 400812 from http://svn.asterisk.org/svn/asterisk/branches/12


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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


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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


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2013-10-08 20:52:04 +00:00