asterisk/asterisk.git
5 years agoThe assertion that peer was not found on final event
Scott Griepentrog [Fri, 29 Aug 2014 18:46:19 +0000 (18:46 +0000)]
The assertion that peer was not found on final event
message was being triggered on configuration reload.
This patch changes that case to just return instead.

Review: https://reviewboard.asterisk.org/r/3953/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422358 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoLICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP
Matthew Jordan [Thu, 28 Aug 2014 21:54:44 +0000 (21:54 +0000)]
LICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP

The UniMRCP project distributes Asterisk modules that integrate Asterisk with
UniMRCP, and other Asterisk users use the UniMRCP library as well.
Unfortunately, the UniMRCP license is Apache 2.0, which per the Free Software
Foundation, is not a compatible license with the GPLv2.

"Please note that this license is not compatible with GPL version 2, because it
has some requirements that are not in that GPL version. These include certain
patent termination and indemnification provisions. The patent termination
provision is a good thing, which is why we recommend the Apache 2.0 license for
substantial programs over other lax permissive licenses."

On the other hand, UniMRCP is a great project and we'd like to let people use
it with Asterisk.

This patch updates the LICENSE text to allow users to link Asterisk with
UniMRCP and distribute the resulting binaries.
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5 years agochan_iax2: Fix Dynamic IAX2 Registrations After Temporary DNS Failure
Michael L. Young [Thu, 28 Aug 2014 20:31:48 +0000 (20:31 +0000)]
chan_iax2: Fix Dynamic IAX2 Registrations After Temporary DNS Failure

The reporter on the issue found some issues when upgrading from version 10 to 11
on 55 hosts.

Two situations that can occur with dynamic registrations.

1.  With dnsmgr disabled, if the host is not resolvable we are not trying to
    resolve the host again when it is time to attempt to register again.  This
    results in never registering to the host.
2.  With dnsmgr enabled, when the host is temporarily not resolvable the
    address is set to 0.0.0.0:0 and then when the host is resolvable the port
    is not being restored and stays set to 0.

This patch resolves these two issues by:

* Storing the hostname so that it can be used for resolving with DNS.
* Resolve the hostname on the next scheduled attempt to register.
* Storing the port used to reach the host so that when the hostname is
  resolvable again, we can set the port again if the port is still unset after
  looking up the host.

ASTERISK-23767 #close
Reported by: David Herselman
Tested by: David Herselman, Michael L. Young
Patches:
    asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff
                                     uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/3856/
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5 years agoAdded ConfBridge AMI event note to UPGRADE.txt.
Richard Mudgett [Thu, 28 Aug 2014 17:29:16 +0000 (17:29 +0000)]
Added ConfBridge AMI event note to UPGRADE.txt.
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5 years agochan_sip.c: Add 'rtpbindaddr' setting
Paul Belanger [Thu, 28 Aug 2014 16:06:55 +0000 (16:06 +0000)]
chan_sip.c: Add 'rtpbindaddr' setting

Users now have the ability to bind the rtpengine instance to a specific IP
address.  For example, you want chan_sip (call control) on eth0 but rtp (media)
on eth1.

ASTERISK-24280 #close
Reported by: Paul Belanger
Tested by: Paul Belanger
Review: https://reviewboard.asterisk.org/r/3952/
Patches:
    rtpengine.diff uploaded by Paul Belanger

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5 years agoFix bug that did not allow for multiple batched RLS notifications to be sent.
Mark Michelson [Thu, 28 Aug 2014 15:50:41 +0000 (15:50 +0000)]
Fix bug that did not allow for multiple batched RLS notifications to be sent.

A misunderstanding of how the scheduler worked caused further batched notifications
beyond the first not to get scheduled. Now we reset our scheduler ID to -1 after
the batched notification is sent. This way, further notifications can be scheduled
when they arise.
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5 years agores/res_pjsip/pjsip_options.c: Eliminate excessive RAII_VAR usage.
Richard Mudgett [Thu, 28 Aug 2014 00:44:59 +0000 (00:44 +0000)]
res/res_pjsip/pjsip_options.c: Eliminate excessive RAII_VAR usage.

* Fix off nominal ref leak in find_or_create_contact_status().

* Add missing NULL check of status in update_contact_status() and
init_start_time().
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5 years agosched: Fix typo and whitespace change.
Richard Mudgett [Thu, 28 Aug 2014 00:16:01 +0000 (00:16 +0000)]
sched: Fix typo and whitespace change.
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5 years agoconfbridge: Add 'Admin' param to join, leave, mute, unmute and talking events
George Joseph [Wed, 27 Aug 2014 17:30:51 +0000 (17:30 +0000)]
confbridge: Add 'Admin' param to join, leave, mute, unmute and talking events

Currently there's no way to tell if a user is an admin or not when receiving
the join, leave, mute, unmute and talking events.  This patch adds that
capability.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3950/
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5 years agoCallerID: Fix parsing of malformed callerid
Kinsey Moore [Wed, 27 Aug 2014 15:39:35 +0000 (15:39 +0000)]
CallerID: Fix parsing of malformed callerid

This allows the callerid parsing function to handle malformed input
strings and strings containing escaped and unescaped double quotes.
This also adds a unittest to cover many of the cases where the parsing
algorithm previously failed.

Review: https://reviewboard.asterisk.org/r/3923/
Review: https://reviewboard.asterisk.org/r/3933/
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5 years agoconfbridge: Make kick, mute and unmute handle channel targets consistently.
George Joseph [Tue, 26 Aug 2014 23:30:00 +0000 (23:30 +0000)]
confbridge: Make kick, mute and unmute handle channel targets consistently.

Kick, mute and unmute were a little inconsistent in their handling of channel
targets.  This patch cleans that up by insuring they all handle the 'all'
target consistently and adds the 'participants' target which acts on
non-admins.  Documentation for kick was also cleaned up as it never
supported partial channel names.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3944/
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5 years agoFix race condition in the scheduler when deleting a running entry.
Mark Michelson [Tue, 26 Aug 2014 22:14:46 +0000 (22:14 +0000)]
Fix race condition in the scheduler when deleting a running entry.

When scheduled tasks run, they are removed from the heap (or hashtab).
When a scheduled task is deleted, if the task can't be found in the
heap (or hashtab), an assertion is triggered. If DO_CRASH is enabled,
this assertion causes a crash.

The problem is, sometimes it just so happens that someone attempts
to delete a scheduled task at the time that it is running, leading
to a crash. This change corrects the issue by tracking which task
is currently running. If that task is attempted to be deleted,
then we mark the task, and then wait for the task to complete.
This way, we can be sure to coordinate task deletion and memory
freeing.

ASTERISK-24212
Reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/3927
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5 years agores_musiconhold.c: Release any format refs before memset().
Richard Mudgett [Mon, 25 Aug 2014 16:45:40 +0000 (16:45 +0000)]
res_musiconhold.c: Release any format refs before memset().

* Clear the channel music_state pointer before destroying the music_state
object for safety.
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5 years agores_musiconhold: Fix MOH restarting where it left off from the last hold.
Richard Mudgett [Mon, 25 Aug 2014 16:16:52 +0000 (16:16 +0000)]
res_musiconhold: Fix MOH restarting where it left off from the last hold.

Restore code removed by https://reviewboard.asterisk.org/r/3536/ that
introduced a regression that prevents MOH from restarting were it left off
the last time.

ASTERISK-24019 #close
Reported by: Jason Richards
Patches:
      jira_asterisk_24019_v1.8.patch (license #5621) patch uploaded by rmudgett

Review: https://reviewboard.asterisk.org/r/3928/
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5 years agores_pjsip_transport_websocket: Attach the Websocket module on outgoing INVITEs.
Joshua Colp [Sun, 24 Aug 2014 19:37:00 +0000 (19:37 +0000)]
res_pjsip_transport_websocket: Attach the Websocket module on outgoing INVITEs.

In order to alter the Contact header on in-dialog requests and responses the
Websocket module must be attached on outgoing INVITEs. The Contact header is
modified so that the PJSIP transport layer can find and use the existing
Websocket connection based on the source IP address, port, and transport.

ASTERISK-24143 #close
Reported by: Aleksei Kulakov
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5 years agores_pjsip_transport_websocket: Fix a progressive memory growth.
Joshua Colp [Sun, 24 Aug 2014 19:21:33 +0000 (19:21 +0000)]
res_pjsip_transport_websocket: Fix a progressive memory growth.

The packet structure used to receive messages was using the transport
pool. This meant that for each parsing the pool would grow accordingly.
Since memory can not be reclaimed without resetting it this would
cause the memory pool to grow and grow.

This change uses a specific memory pool for the packet structure and
resets it to a fresh state after the message has been received and
handled.
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5 years agores_pjsip_transport_websocket: Ensure secure Websocket clients can be called.
Joshua Colp [Sun, 24 Aug 2014 18:54:00 +0000 (18:54 +0000)]
res_pjsip_transport_websocket: Ensure secure Websocket clients can be called.

This change enforces the transport in the Contact header for Websocket clients.
Previously a client may provide a transport of 'ws' when it is actually using
a transport of 'wss'. This would cause outgoing calls to fail as the existing
connection could not be found.
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5 years agochan_sip: Use the server reflexive ICE candidate RTCP port as provided.
Joshua Colp [Sun, 24 Aug 2014 17:22:48 +0000 (17:22 +0000)]
chan_sip: Use the server reflexive ICE candidate RTCP port as provided.

This code originally worked around an issue within res_rtp_asterisk itself.
The wrong socket was being used for the STUN check for RTCP, causing the
port to be the same as RTP. This was subsequently fixed and the RTCP port
provided for the ICE candidate is correct and does not need to be incremented.

ASTERISK-23997 #close
Reported by: Badalian Vyacheslav
Patches:
 plus1.diff submitted by Badalian Vyacheslav (license 5249)
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5 years agoFix a locking inversion in MixMonitor.
Mark Michelson [Fri, 22 Aug 2014 16:56:57 +0000 (16:56 +0000)]
Fix a locking inversion in MixMonitor.

We need to unlock the audiohook before trying to lock
the channel, since the correct locking order is channel
then audiohook.
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5 years agoARI: Fix a crash caused by hanging during playback to a channel in a bridge
Jonathan Rose [Fri, 22 Aug 2014 16:52:51 +0000 (16:52 +0000)]
ARI: Fix a crash caused by hanging during playback to a channel in a bridge

ASTERISK-24147 #close
Reported by: Edvin Vidmar
Review: https://reviewboard.asterisk.org/r/3908/
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5 years agomain/message: Add a new-line to a DEBUG message
Matthew Jordan [Fri, 22 Aug 2014 14:09:07 +0000 (14:09 +0000)]
main/message: Add a new-line to a DEBUG message
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5 years agores_musiconhold.c: Remove obsolete REF_DEBUG code.
Richard Mudgett [Thu, 21 Aug 2014 22:09:24 +0000 (22:09 +0000)]
res_musiconhold.c: Remove obsolete REF_DEBUG code.

Remove unneeded code that writes to the wrong file location in an obsolete
format.
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5 years agoSwitch from hostname to an IP address in the SDP origin line.
Mark Michelson [Thu, 21 Aug 2014 21:43:45 +0000 (21:43 +0000)]
Switch from hostname to an IP address in the SDP origin line.

Using the hostname in the SDP origin line may not satisfy the requirement
of RFC 4566 that we use a FQDN or IP address. This change has us use the
same information from the SDP connection line if possible. If not possible,
we'll use the configured media address. And if that's not possible, we use
the result of a PJLIB call to get the IP address of ourself.

ASTERISK-23994 #close
Reported by Private Name

Review: https://reviewboard.asterisk.org/r/3925
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5 years agoEnsure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge.
Mark Michelson [Thu, 21 Aug 2014 21:37:03 +0000 (21:37 +0000)]
Ensure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge.

Because of the departable state of channels that enter Stasis bridges, Stasis has to
take responsibility for directing the channel to its intended after-bridge destination
if the channel moves from a Stasis bridge to a non-Stasis bridge. This change ensures
that when such a move occurs, when the channel leaves the bridging system, any after
bridge gotos are honored.

Review: https://reviewboard.asterisk.org/r/3920
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5 years agores_musiconhold: Fix reference leaks caused when reloading with REF_DEBUG set
Jonathan Rose [Thu, 21 Aug 2014 21:35:58 +0000 (21:35 +0000)]
res_musiconhold: Fix reference leaks caused when reloading with REF_DEBUG set

Due to a faulty function for debugging reference decrementing, it was possible
to reduce the refcount on the wrong object if two moh classes of the same name
were in the moh class container.

(closes issue ASTERISK-22252)
Reported by: Walter Doekes
Patches:
    18_moh_debug_ref_patch.diff Uploaded by Jonathan Rose (license 6182)
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5 years agoLet's try checking the name and number, instead of the name twice.
Mark Michelson [Thu, 21 Aug 2014 21:28:16 +0000 (21:28 +0000)]
Let's try checking the name and number, instead of the name twice.
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5 years agoImprove consistency of party ID privacy usage.
Mark Michelson [Thu, 21 Aug 2014 21:19:06 +0000 (21:19 +0000)]
Improve consistency of party ID privacy usage.

Prior to this change, the Remote-Party-ID header took the position of
"If caller name and number are not explicitly allowed, then they are private"
and P-Asserted-Identity took the position of
"Caller name and number are only private if marked explicitly so"

Now both mechanisms of conveying party identification use the former approach.
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5 years agochan_sip: Don't use port derived from fromdomain if it isn't set
Matthew Jordan [Thu, 21 Aug 2014 17:35:15 +0000 (17:35 +0000)]
chan_sip: Don't use port derived from fromdomain if it isn't set

If a user does not provide a port in the fromdomain setting, chan_sip will set
the fromdomainport to STANDARD_SIP_PORT (5060). The fromdomainport value will
then get used unilaterally in certain places. This causes issues with TLS,
where the default port is expected to be 5061.

This patch modifies chan_sip such that fromdomainport is only used if it is
not the standard SIP port; otherwise, the port from the SIP pvt's recorded
self IP address is used.

Review: https://reviewboard.asterisk.org/r/3893/

ASTERISK-24178 #close
Reported by: Elazar Broad
patches:
  fromdomainport_fix.diff uploaded by Elazar Broad (License 5835)
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5 years agoARI: Fix implicit answer when playback is initiated on unanswered channel
Matthew Jordan [Thu, 21 Aug 2014 15:25:25 +0000 (15:25 +0000)]
ARI: Fix implicit answer when playback is initiated on unanswered channel

When issuing a POST /channels/{channel_id}/play on a channel that is not
yet answered, ARI is supposed to:
* Queue up an AST_CONTROL_PROGRESS on the channel
* Start up the playback of the media

Instead, we sneak an answer on the channel right before starting playing media.

This is due to ARI's usage of control_streamfile. This function implicitly
answers the channel (and doesn't give ARI the option to stop it). The answering
of the channel here is probably unnecessary:
* app_voicemail, by far the biggest consumer of this function, always answers
  the channels anyway
* control stream file (in res_agi) and ControlPlayback probably shouldn't be
  implicitly answering the channel. Answering should not be tied directly to
  playing back media.

As it turns out, the answering of the channel here is pretty old:
356042    twilson       if (ast_channel_state(chan) != AST_STATE_UP) {
  3087      anthm               res = ast_answer(chan);
180259   tilghman       }

(As in, ancient?)

Note that others ran into this problem and commented about it on various
mailing lists.

Review: https://reviewboard.asterisk.org/r/3907/

ASTERISK-24229 #close
Reported by: Matt Jordan
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5 years agoClean up files that do not end with newlines
Matthew Jordan [Thu, 21 Aug 2014 14:52:28 +0000 (14:52 +0000)]
Clean up files that do not end with newlines

Trivial patch to add new lines to several files missing them. This fixes
warnings when compiling with gcc 4.1.2 on CentOS 5.

ASTERISK-24245 #close
Reported by: Shaun Ruffell
patches:
  0002-Trivial-addition-of-newlines-at-end-of-three-files.patch uploaded by Shaun Ruffell (License 5417)
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5 years agouri: Quiet warning about type qualifiers ignored on function return type
Matthew Jordan [Thu, 21 Aug 2014 14:42:12 +0000 (14:42 +0000)]
uri: Quiet warning about type qualifiers ignored on function return type

This patch fixes gcc warnings that occur due to the type qualifier 'const'
being ignored on a return type of int.

ASTERISK-24246 #close
Reported by: Shaun Ruffell
patches:
  0001-main-uri-Quiet-warning-about-ignored-attribute-on-re.patch uploaded by Shaun Ruffell (License 5417)
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5 years agochan_pjsip: Update media translation paths when new SDP negotiated.
Richard Mudgett [Wed, 20 Aug 2014 22:52:44 +0000 (22:52 +0000)]
chan_pjsip: Update media translation paths when new SDP negotiated.

On a SIP reinvite that changes media strams, the PJSIP channel driver was
flooding the log with "Asked to transmit frame type %s, while native
formats is %s" warnings.

* Fixes PJSIP not setting up translation paths when the formats change on
a reinvite.  AFS-63 was effectively reintroduced because of the media
formats work.  res_pjsip_sdp_rtp.c:set_caps()

* Improved the unexpected frame format WARNING message to include more
information.

* Added protective locking while altering formats on a channel.  Reworked
set_format() to simplify and protect the formats under manipulation.

* Restored some code that got lost in the media_formats work.
(channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps())

AFS-137 #close
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/3906/
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5 years agocli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled.
Richard Mudgett [Wed, 20 Aug 2014 22:23:23 +0000 (22:23 +0000)]
cli.c: Fix tab completion of "module load" when MALLOC_DEBUG is enabled.

filename_completion_function() returns memory that was not allocated by
the MALLOC_DEBUG allocation tracker so the memory must be freed by
ast_std_free().
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5 years agoSet the role for inbound subscriptions correctly.
Mark Michelson [Wed, 20 Aug 2014 20:41:04 +0000 (20:41 +0000)]
Set the role for inbound subscriptions correctly.

This was causing the AMI show_subscriptions test in
the testsuite to fail since all subscriptions were being
seen as subscribers instead of notifiers.
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5 years agoMove evaluation of set_var options in pjsip to the end of channel initialization.
Mark Michelson [Wed, 20 Aug 2014 20:04:43 +0000 (20:04 +0000)]
Move evaluation of set_var options in pjsip to the end of channel initialization.

This allows for set_var to override certain defaults such as caller ID and codec
values. This also fixes a test suite regression. The "set_var" test suite test attempted
to use set_var to override caller ID, but a recent change caused that to no longer work.
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5 years agoStasis: Add information to blind transfer event
Kinsey Moore [Wed, 20 Aug 2014 13:06:33 +0000 (13:06 +0000)]
Stasis: Add information to blind transfer event

When a blind transfer occurs that is forced to create a local channel
pair to satisfy the transfer request, information about the local
channel pair is not published. This adds a field to describe that
channel to the blind transfer message struct so that this information
is conveyed properly to consumers of the blind transfer message.

This also fixes a bug in which Stasis() was unable to properly identify
the channel that was replacing an existing Stasis-controlled channel
due to a blind transfer.

Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3921/
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5 years agoAMI: Add AllVariables parameter to Status
Kinsey Moore [Wed, 20 Aug 2014 12:39:39 +0000 (12:39 +0000)]
AMI: Add AllVariables parameter to Status

This adds the AllVariables parameter to the Status AMI action such that
if defined and set to "true", all channel variables will be reported in
the subsequent Status event(s). This parameter does not negate the
functionality of the "Variables" parameter so that global variables and
dialplan functions can be requested.

Review: https://reviewboard.asterisk.org/r/3915/

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5 years agoAlter documentation for callerid_privacy to use correct values.
Mark Michelson [Tue, 19 Aug 2014 20:28:56 +0000 (20:28 +0000)]
Alter documentation for callerid_privacy to use correct values.
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5 years agoFix compilation error on certain versions of GCC.
Mark Michelson [Tue, 19 Aug 2014 19:55:56 +0000 (19:55 +0000)]
Fix compilation error on certain versions of GCC.
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5 years agoAMI Docs: Fix Status channel parameter optionality
Kinsey Moore [Tue, 19 Aug 2014 19:43:14 +0000 (19:43 +0000)]
AMI Docs: Fix Status channel parameter optionality
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5 years agoARI: Fix a bug where /channels/{channelID}/continue doesn't execute PBX
Jonathan Rose [Tue, 19 Aug 2014 16:36:30 +0000 (16:36 +0000)]
ARI: Fix a bug where /channels/{channelID}/continue doesn't execute PBX

If /channels/{channelID}/continue is called on a channel that was originated
without a PBX (such as the ARI command POST channel with a stasis application
argument), the channel will not start dialplan execution. This patch will now
run the PBX out of the stasis execution if the channel doesn't currently have
an active PBX upon continuing.

ASTERISK-24043 #close
Reported by: Krandon Bruse
Review: https://reviewboard.asterisk.org/r/3917/
Patches:
    stasis-continue.diff submitted by Krandon Bruse (license 6631)
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5 years agochan_pjsip: Fix attended transfer connected line name update.
Richard Mudgett [Tue, 19 Aug 2014 16:16:03 +0000 (16:16 +0000)]
chan_pjsip: Fix attended transfer connected line name update.

A calls B
B answers
B SIP attended transfers to C
C answers, B and C can see each other's connected line information
B completes the transfer
A has number but no name connected line information about C
  while C has the full information about A

I examined the incoming and outgoing party id information handling of
chan_pjsip and found several issues:

* Fixed ast_sip_session_create_outgoing() not setting up the configured
endpoint id as the new channel's caller id.  This is why party A got
default connected line information.

* Made update_initial_connected_line() use the channel's CALLERID(id)
information.  The core, app_dial, or predial routine may have filled in or
changed the endpoint caller id information.

* Fixed chan_pjsip_new() not setting the full party id information
available on the caller id and ANI party id.  This includes the configured
callerid_tag string and other party id fields.

* Fixed accessing channel party id information without the channel lock
held.

* Fixed using the effective connected line id without doing a deep copy
outside of holding the channel lock.  Shallow copy string pointers can
become stale if the channel lock is not held.

* Made queue_connected_line_update() also update the channel's
CALLERID(id) information.  Moving the channel to another bridge would need
the information there for the new bridge peer.

* Fixed off nominal memory leak in update_incoming_connected_line().

* Added pjsip.conf callerid_tag string to party id information from
enabled trust_inbound endpoint in caller_id_incoming_request().

AFS-98 #close
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/3913/
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5 years agoSkinny: Fixup compile warning for non dev-mode.
Damien Wedhorn [Mon, 18 Aug 2014 21:18:17 +0000 (21:18 +0000)]
Skinny: Fixup compile warning for non dev-mode.
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5 years agofunc_config: Change 'Not Found' message from ERROR to DEBUG
George Joseph [Mon, 18 Aug 2014 20:20:59 +0000 (20:20 +0000)]
func_config: Change 'Not Found' message from ERROR to DEBUG

When you call the CONFIG dialplan function with the name of a variable that
doesn't exist in the target context you get an ERROR.  This does nothing but
clutter up the logs with messages that may be perfectly acceptable.  Just
because a variable wasn't in the context doesn't mean it's an error.  Maybei
t's optional or just needs to be defaulted or ignored.

This patch changes the log level from ERROR to DEBUG.  If a dialplan developer
wants to debug their dialplan they still canby setting the console debug level
as needed.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3919/
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5 years agoMultiple revisions 421311-421312
Matthew Jordan [Mon, 18 Aug 2014 01:14:51 +0000 (01:14 +0000)]
Multiple revisions 421311-421312

........
  r421311 | mjordan | 2014-08-17 20:11:28 -0500 (Sun, 17 Aug 2014) | 9 lines

  res/ari/resource_channels: Don't return allocation failure on failed function

  If a function fails to execute, it is most likely due to one of two reasons:
  (1) The function doesn't exist or can't be read from
  (2) The function is dangerous and is restricted based on the user's permissions

  Currently we return allocation failure, which is incorrect. This updates the
  reason code to more accurately reflect why the request failed.

  ASTERISK-24215
........
  r421312 | mjordan | 2014-08-17 20:13:41 -0500 (Sun, 17 Aug 2014) | 4 lines

  res/ari/resource_channels: Fix compilation issue

  Forgot a parameter. Whoops.
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5 years agoImprove call forwarding reporting, especially with regards to ARI.
Matthew Jordan [Mon, 18 Aug 2014 00:57:01 +0000 (00:57 +0000)]
Improve call forwarding reporting, especially with regards to ARI.

This patch addresses a few issues:

1) The order of Dial events have been changed when performing a call forward.
   The order has now been altered to
    1) Dial begins dialing channel A.
    2) When A forwards the call to B, we issue the dial end event to channel
       A, indicating the dial is being canceled due to a forward to B.
    3) When the call to channel B occurs, we then issue a new dial begin to
       channel B.

2) Call forwards are now reported on the calling channel, not the peer channel.

3) AMI DialEnd events have been altered to display the extension the call is
   being forwarded to when relevant.

4) You can now get the values of channel variables for channels that are not
   currently in the Stasis application. This brings the retrieval of channel
   variables more in line with the rest of channel read operations since they
   may be performed on channels not in Stasis.

ASTERISK-24134 #close
Reported by Matt Jordan

ASTERISK-24138 #close
Reported by Matt Jordan

Patches:
forward-shenanigans.diff uploaded by Matt Jordan (License #6283)

Review: https://reviewboard.asterisk.org/r/3899
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5 years agoapps/app_meetme: Fix crash when publishing MeetMe messages with no channel
Matthew Jordan [Sun, 17 Aug 2014 23:29:34 +0000 (23:29 +0000)]
apps/app_meetme: Fix crash when publishing MeetMe messages with no channel

The same function, meetme_stasis_generate_msg, handles creating and publishing
Stasis message both when there are channels in the MeetMe conference and when
there are no channels in the conference. When the performance improvement was
made to use cached snapshots, this created a situation where Asterisk would
crash: obtaining a cached snapshot is not NULL tolerant.

This patch restores the previous implementation, which used a NULL safe set
of routines to produce a blob containing the channel snapshot (if available)
and information about the MeetMe conference.

ASTERISK-24234 #close
Reported by: Shaun Ruffell
Tested by: Shaun Ruffell
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5 years agoapps/app_dial: Fix Dial 'z' option
Matthew Jordan [Sun, 17 Aug 2014 23:10:21 +0000 (23:10 +0000)]
apps/app_dial: Fix Dial 'z' option

The 'z' option is supposed to disable the dial timeout in the case of a call
forward. Unfortunately, the wrong timeout timer was passed to the do_forward
function, resulting in the option not working.

ASTERISK-24225 #close
Reported by: dimitripietro
Tested by: dimitripietro
patches:
  jira_asterisk_24225_v1.8.patch uploaded by rmudgett (License 5621)
  jira_asterisk_24225_v11.patch uploaded by rmudgett (License 5621)
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Merged revisions 421234 from http://svn.asterisk.org/svn/asterisk/branches/12
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Merged revisions 421235 from http://svn.asterisk.org/svn/asterisk/branches/13

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5 years agoconfigure: Undefine FORTIFY_SOURCE prior to defining it for patched gcc
Matthew Jordan [Sun, 17 Aug 2014 22:35:27 +0000 (22:35 +0000)]
configure: Undefine FORTIFY_SOURCE prior to defining it for patched gcc

Some distributions of Linux patch gcc to define FORTIFY_SOURCE when gcc is
executed with optimization. This "help" unfortunately results in re-definition
warnings when FORTIFY_SOURCE is later defined in Asterisk's build system. This
patch undefines FORTIFY_SOURCE prior to defining it to prevent this warning.

Review: https://reviewboard.asterisk.org/r/3912/

ASTERISK-24032 #close
Reported by: Kilburn
Tested by: Kilburn, wdoekes
patches:
  1.8.diff uploaded by cloos (License 5956)
  10.diff uploaded by cloos (License 5956)
  11.diff uploaded by cloos (License 5956)
  12.diff uploaded by cloos (License 5956)
  13.diff uploaded by cloos (License 5956)
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5 years agores_http_websocket: Include query parameters in client connection requests.
Joshua Colp [Sun, 17 Aug 2014 16:11:27 +0000 (16:11 +0000)]
res_http_websocket: Include query parameters in client connection requests.

Review: https://reviewboard.asterisk.org/r/3914/
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5 years agoBridging: Fix a behavioral change when checking if a channel is leaving a bridge
Jonathan Rose [Fri, 15 Aug 2014 17:26:12 +0000 (17:26 +0000)]
Bridging: Fix a behavioral change when checking if a channel is leaving a bridge

r420934 introduced some failures in the test suite.  Upon investigating, it was
discovered that differences in the way we were evaluating whether a channel was in
the process of leaving a bridge were causing some reinvites not to occur (mostly
reinvites back to Asterisk when ending a call). This patch fixes that behavioral
change.

ASTERISK-24027 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3910/
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5 years agoapp_voicemail/app: Remove test events that were duplicated by r421059
Matthew Jordan [Fri, 15 Aug 2014 15:50:46 +0000 (15:50 +0000)]
app_voicemail/app: Remove test events that were duplicated by r421059

Moving the test event raised when a file is played back (which occurred in
r421059) broke the ever loving snot out of the voicemail tests. This caused
duplicate test events to get raised, as app_voicemail and main/app were raising
events prior to call ast_streamfile. The voicemail tests did not enjoy getting
multiple events.

Since raising the playback event in ast_streamfile is far more useful to the
vast majority of tests, this patch keeps the call there and simply removes the
extraneous calls that duplicated the event.
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5 years agores/res_hep_rtcp: Remove dependency on PJSIP
Matthew Jordan [Thu, 14 Aug 2014 21:16:32 +0000 (21:16 +0000)]
res/res_hep_rtcp: Remove dependency on PJSIP

The res_hep_rtcp module was incorrectly including <pjsip.h>. This didn't need
to be included, as the module does not using PJPROJECT any fashion.
Unfortunately, because res_hep_rtcp did not include pjsip in its MODULEINFO as
a dependency, this also meant that res_hep_rtcp will fail to compile on a
system without PJPROJECT.

This patch removes the include.

Thanks to Damien Wedhorn for pointing this out in #asterisk-dev.

ASTERISK-24236 #close
Reported by: Damien Wedhorn, Matt Jordan
Tested by: Damien Wedhorn
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5 years agomain/file: Move test event to emit PLAYBACK event more consistently
Matthew Jordan [Thu, 14 Aug 2014 20:59:15 +0000 (20:59 +0000)]
main/file: Move test event to emit PLAYBACK event more consistently

This is being done in advance of the test for ASTERISK-23953
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5 years agocel: Make sure channels in extra fields include their unique IDs as well
Matthew Jordan [Thu, 14 Aug 2014 19:21:51 +0000 (19:21 +0000)]
cel: Make sure channels in extra fields include their unique IDs as well

CEL typically tracks a lot of information using the unique ID of the channel.
This is typically needed due to tying events together using the linked ID of
the various channels involved in a "call", which is derived from the channel ID
of the oldest channel involved in a bridge (or in the case of a Dial, the
parent channel).

Previously, we had updated the extra fields to include the involved channel
names, but forgot to put in the unique ID. This patch corrects that error.
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5 years agoARI: Originate to app local channel subscription code optimization.
Richard Mudgett [Thu, 14 Aug 2014 16:33:27 +0000 (16:33 +0000)]
ARI: Originate to app local channel subscription code optimization.

Reduce the scope of local_peer and only get it if the ARI originate is
subscribing to the channels.

Review: https://reviewboard.asterisk.org/r/3905/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421012 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochannel_internal_api.c: Replace some code with ao2_replace().
Richard Mudgett [Thu, 14 Aug 2014 16:01:39 +0000 (16:01 +0000)]
channel_internal_api.c: Replace some code with ao2_replace().

Use ao2_replace() instead of ao2_cleanup(); ao2_bump().

ao2_replace() has the advantange of not altering the ref count if the
replaced pointer is the same.

Review: https://reviewboard.asterisk.org/r/3904/
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5 years agores_pjsip_send_to_voicemail.c: Fix svn file properties.
Richard Mudgett [Wed, 13 Aug 2014 17:05:01 +0000 (17:05 +0000)]
res_pjsip_send_to_voicemail.c: Fix svn file properties.
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5 years agoPJSIP: Prevent crash no-URI contacts
Kinsey Moore [Wed, 13 Aug 2014 16:56:14 +0000 (16:56 +0000)]
PJSIP: Prevent crash no-URI contacts

This prevents a crash from occurring when a contact with no URI is used
for the creation of an outbound out-of-dialog request with no
associated endpoint.
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5 years agoBridges: Fix feature interruption/unintended kick caused by external actions
Jonathan Rose [Wed, 13 Aug 2014 16:24:37 +0000 (16:24 +0000)]
Bridges: Fix feature interruption/unintended kick caused by external actions

If a manager or CLI user attached a mixmonitor to a call running a dynamic
bridge feature while in a bridge, the feature would be interrupted and the
channel would be forcibly kicked out of the bridge (usually ending the call
during a simple 1 to 1 call). This would also occur during any similar action
that could set the unbridge soft hangup flag, so the fix for this was to
remove unbridge from the soft hangup flags and make it a separate thing all
together.

ASTERISK-24027 #close
Reported by: mjordan
Review: https://reviewboard.asterisk.org/r/3900/
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5 years agoAMI: Improve documentation for Status action
Kinsey Moore [Wed, 13 Aug 2014 14:31:46 +0000 (14:31 +0000)]
AMI: Improve documentation for Status action
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5 years agologger: Don't store verbose-magic in the log files.
Walter Doekes [Wed, 13 Aug 2014 07:54:10 +0000 (07:54 +0000)]
logger: Don't store verbose-magic in the log files.

In r399267, the verbose2magic stuff was edited. This time it results
in magic characters in the log files for multiline messages.

In trunk (and 13) this was fixed by the "stripping" of those
characters from multiline messages (in r414798).

This fix is altered to actually strip the characters and not replace
them with blanks.

Review: https://reviewboard.asterisk.org/r/3901/
Review: https://reviewboard.asterisk.org/r/3902/
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5 years agochan_sip: Fix type mismatch when the format is changed.
Richard Mudgett [Tue, 12 Aug 2014 23:45:17 +0000 (23:45 +0000)]
chan_sip: Fix type mismatch when the format is changed.

Symptom is most likely an invalid ao2 object bad magic number message or a
less likely crash.
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5 years agores_stasis_snoop.c: Fix off nominial exit path leaving Snoop channel locked and not...
Richard Mudgett [Tue, 12 Aug 2014 23:36:37 +0000 (23:36 +0000)]
res_stasis_snoop.c: Fix off nominial exit path leaving Snoop channel locked and not hungup.

* Made use ast_copy_string() instead of strcpy() for snoop uniqueid for
safety.  There is no guarantee that the max channel uniqueid length will
remain the same as the snoop uniqueid space.
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5 years agoapp_voicemail: Fix the "test_voicemail_vm_info" unit test.
Joshua Colp [Tue, 12 Aug 2014 11:18:17 +0000 (11:18 +0000)]
app_voicemail: Fix the "test_voicemail_vm_info" unit test.
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5 years agores/stasis/command.c: Fix recent commit using spaces instead of tabs.
Richard Mudgett [Mon, 11 Aug 2014 21:04:21 +0000 (21:04 +0000)]
res/stasis/command.c: Fix recent commit using spaces instead of tabs.
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5 years agoAMI/ARI: Update version to 2.5.0/1.5.0 respectively
Matthew Jordan [Mon, 11 Aug 2014 18:51:43 +0000 (18:51 +0000)]
AMI/ARI: Update version to 2.5.0/1.5.0 respectively

This is to support the backwards compatible changes made in the next version
of Asterisk.
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5 years agoStasis: Use the correct return value
Kinsey Moore [Mon, 11 Aug 2014 18:46:59 +0000 (18:46 +0000)]
Stasis: Use the correct return value

Return the correct value instead of always returning 0 when setting
internal status on unreal channels.

Reported by: Richard Mudgett
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5 years agoStasis: Allow internal channels directly into bridges
Kinsey Moore [Mon, 11 Aug 2014 18:38:15 +0000 (18:38 +0000)]
Stasis: Allow internal channels directly into bridges

The patch to catch channels being shoehorned into Stasis() via external
mechanisms also happens to catch Announcer and Recorder channels
because they aren't known to be stasis-controlled channels in the usual
sense. This marks those channels as Stasis()-internal channels and
allows them directly into bridges.

Review: https://reviewboard.asterisk.org/r/3903/
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5 years agoFix crashing unit tests with regards to RLS.
Mark Michelson [Mon, 11 Aug 2014 17:40:07 +0000 (17:40 +0000)]
Fix crashing unit tests with regards to RLS.

The unit tests require a sorcery.conf file that has been
set up to store resource lists in memory rather than retrieving
from configuration.

With a setup that is not conducive to running the tests, a fault
in sorcery currently causes Asterisk to crash when attempting to
run any of the tests.

To get around the crash, this adds a function that verifies the
current environment and marks the tests as "not run" if the setup
is not correct.
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5 years agoFix crash encountered by the testsuite.
Mark Michelson [Mon, 11 Aug 2014 16:03:41 +0000 (16:03 +0000)]
Fix crash encountered by the testsuite.

Running testsuite tests locally produced no errors, but when
run using the continuous integration framework, crashes occurred.

The crashes occurred due to a refcounting error that had been fixed
for a similar situation.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420759 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_hep: Remove disabling of modules
Matthew Jordan [Mon, 11 Aug 2014 13:57:53 +0000 (13:57 +0000)]
res_hep: Remove disabling of modules

These modules were originally specified as being disabled, as they were
introduced midstream in Asterisk 12. That makes it nicer for folks who are
upgrading to a new release in the middle of Asterisk 12. That's not the case
for Asterisk 13: it's a brand new release. There's no reason to have the
modules disabled by default in that case.
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5 years agogeneral: Fix memory Corruption in __ast_string_field_ptr_build_va.
Walter Doekes [Mon, 11 Aug 2014 10:41:07 +0000 (10:41 +0000)]
general: Fix memory Corruption in __ast_string_field_ptr_build_va.

If the space left in a stringfield is between 0 and
(alignof(ast_string_field_allocation)-1) adding new data would cause
memory corruption, because we would assume enough space (unsigned
underrun).

Thanks Arnd Schmitter for reporting and finding out the cause!

ASTERISK-23508 #close
Reported by: Arnd Schmitter
Tested by: Arnd Schmitter, JoshE

Review: https://reviewboard.asterisk.org/r/3898/
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5 years agotcptls: Avoid compiler warning on non-dev-mode.
Walter Doekes [Mon, 11 Aug 2014 09:55:13 +0000 (09:55 +0000)]
tcptls: Avoid compiler warning on non-dev-mode.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420658 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agofuncs/func_jitterbuffer: Tweak documentation
Matthew Jordan [Mon, 11 Aug 2014 01:31:56 +0000 (01:31 +0000)]
funcs/func_jitterbuffer: Tweak documentation

This patch merely reformats and cleans up a bit of the jitterbuffer
documentation for the wiki.
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Merged revisions 420639 from http://svn.asterisk.org/svn/asterisk/branches/13

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5 years agoapp_queue: Add RealTime support for queue rules
Matthew Jordan [Mon, 11 Aug 2014 00:14:53 +0000 (00:14 +0000)]
app_queue: Add RealTime support for queue rules

This patch gives the optional ability to keep queue rules in RealTime. It is
important to note that with this patch:
 (a) Queue rules in RealTime are only examined on module load/reload
 (b) Queue rules are loaded both from the queuerules.conf file as well as the
     RealTime backend
To inform app_queue to examine RealTime for queue rules, a new setting has been
added to queuerules.conf's general section "realtime_rules". RealTime queue
rules will only be used when this setting is set to "yes".

The schema for the database table supports a rule_name, time, min_penalty, and
max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or
'+' literal is provided. Otherwise, the penalties are treated as constants.

For example:
rule_name, time, min_penalty, max_penalty
'default', '10', '20', '30'
'test2', '20', '30', '55'
'test2', '25', '-11', '+1111'
'test2', '400', '112', '333'
'test3', '0', '4564', '46546'
'test_rule', '40', '15', '50'

which would result in :

Rule: default
 - After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust
   QUEUE_MIN_PENALTY to 20
Rule: test2
 - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust
   QUEUE_MIN_PENALTY to 30
 - After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust
   QUEUE_MIN_PENALTY by -11
 - After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust
   QUEUE_MIN_PENALTY to 112
Rule: test3
 - After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust
   QUEUE_MIN_PENALTY to 4564
Rule: test_rule
 - After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust
   QUEUE_MIN_PENALTY to 15

If you use RealTime, the queue rules will be always reloaded on a module
reload, even if the underlying file did not change. With the option disabled,
the rules will only be reloaded if the file was modified.

Review: https://reviewboard.asterisk.org/r/3607/

ASTERISK-23823 #close
Reported by: Michael K
patches:
  app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420625 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoUpdate CHANGES file
Matthew Jordan [Sun, 10 Aug 2014 22:02:03 +0000 (22:02 +0000)]
Update CHANGES file
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420610 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoUpdate UPGRADE-13.txt file
Matthew Jordan [Sun, 10 Aug 2014 21:35:54 +0000 (21:35 +0000)]
Update UPGRADE-13.txt file

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420608 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoFix build in devmode.
Jason Parker [Fri, 8 Aug 2014 20:08:53 +0000 (20:08 +0000)]
Fix build in devmode.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420593 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoapp_voicemail: Add the ability to specify multiple email addresses.
Jason Parker [Fri, 8 Aug 2014 19:16:29 +0000 (19:16 +0000)]
app_voicemail: Add the ability to specify multiple email addresses.

ASTERISK-24045
Reported by: Jacob Barber
Review: https://reviewboard.asterisk.org/r/3833/
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Merged revisions 420577 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420578 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_sip: Mark chan_sip and its files as extended support
Matthew Jordan [Fri, 8 Aug 2014 17:53:39 +0000 (17:53 +0000)]
chan_sip: Mark chan_sip and its files as extended support
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420563 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agomake_ari_stubs: Update wiki prefix to '13'
Matthew Jordan [Fri, 8 Aug 2014 12:40:16 +0000 (12:40 +0000)]
make_ari_stubs: Update wiki prefix to '13'
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420539 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_ari_resource.c.mustache: Update template to emit module support level
Matthew Jordan [Fri, 8 Aug 2014 12:38:18 +0000 (12:38 +0000)]
res_ari_resource.c.mustache: Update template to emit module support level
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420537 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agomain/message: remove debug message
Matthew Jordan [Fri, 8 Aug 2014 12:33:06 +0000 (12:33 +0000)]
main/message: remove debug message
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420535 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoCEL: Update unit tests for additional information
Kinsey Moore [Fri, 8 Aug 2014 03:07:41 +0000 (03:07 +0000)]
CEL: Update unit tests for additional information

This updates the CEL unit tests for the new information contained in
the attended transfer CEL extra field.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420515 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoInitialize svnmerge from branches/13
Matthew Jordan [Fri, 8 Aug 2014 01:37:05 +0000 (01:37 +0000)]
Initialize svnmerge from branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420499 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoRemove 12 merge properties
Matthew Jordan [Fri, 8 Aug 2014 01:36:16 +0000 (01:36 +0000)]
Remove 12 merge properties

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420498 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoUpdate UPGRADE.txt for 13 branch
Matthew Jordan [Fri, 8 Aug 2014 01:33:18 +0000 (01:33 +0000)]
Update UPGRADE.txt for 13 branch

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420497 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_sip: Replace sip_tls_read() and resolve the large SDP poll issue.
Richard Mudgett [Thu, 7 Aug 2014 21:58:38 +0000 (21:58 +0000)]
chan_sip: Replace sip_tls_read() and resolve the large SDP poll issue.

Replace sip_tls_read() and sip_tcp_read() with a single function and
resolve the poll/wait issue with large SDP payloads.

ASTERISK-18345 #close
Reported by: Stephane Chazelas
Patches:
      tcptls_pollv4.diff (license #5835) patch uploaded by Elazar Broad

Review: https://reviewboard.asterisk.org/r/3882/
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5 years agoStasis: Correct blind transfer message generation
Kinsey Moore [Thu, 7 Aug 2014 21:17:05 +0000 (21:17 +0000)]
Stasis: Correct blind transfer message generation

This fixes the json object creation format string and key name for the
BridgeBlindTransfer Stasis event allowing it to be published properly.
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5 years agoStasis: Ensure transfer messages follow validation rules
Kinsey Moore [Thu, 7 Aug 2014 20:24:15 +0000 (20:24 +0000)]
Stasis: Ensure transfer messages follow validation rules

This makes Stasis() event generation for transfer messages follow
validation rules. Currently, ast_json_null() is being used in place of
omitting a key entirely which falls afoul of these validation rules.

https://reviewboard.asterisk.org/r/3892/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420410 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoFix build in dev mode
Kinsey Moore [Thu, 7 Aug 2014 20:11:15 +0000 (20:11 +0000)]
Fix build in dev mode

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420389 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoEnsure bridges exist when trying to determine bridged parties when publishing transfe...
Mark Michelson [Thu, 7 Aug 2014 19:44:32 +0000 (19:44 +0000)]
Ensure bridges exist when trying to determine bridged parties when publishing transfer information.
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5 years agoAdd support for RFC 4662 resource list subscriptions.
Mark Michelson [Thu, 7 Aug 2014 19:26:32 +0000 (19:26 +0000)]
Add support for RFC 4662 resource list subscriptions.

This commit adds the ability for a user to configure
a resource list in pjsip.conf. Subscribing to this
list simultaneously subscribes the subscriber to all
resources listed. This has the potential to reduce
the amount of SIP traffic when loads of subscribers
on a system attempt to subscribe to each others' states.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420384 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agochan_iax2: Several media format fixes.
Richard Mudgett [Thu, 7 Aug 2014 18:51:16 +0000 (18:51 +0000)]
chan_iax2: Several media format fixes.

* Fixed the iax.conf bandwidth option.  This is the root cause of
ASTERISK-24150.

* Added checks in iax2_request() to ensure that there are actual formats
requested for the new channel to prevent any more fracks from issues like
ASTERISK-24150.  This is a consequence of the iax.conf bandwidth option
not working.

* Fixed struct iax2_codec_pref.order member size mismatch issue when
converting to and from the codec preference order list passed over the
wire.  In addition the values sent over the wire are now compatible with
previous Asterisk versions.

* Fixed several issues dealing with the struct iax2_codec_pref members.
Off-by-one, array limit errors, and the order/framing members always need
to be updated together.

* Made iax2_request() setup the channel's native format preference order
according to the user's wishes.  The new media format strategy needs the
order specified earler.

* Fixed usage of ast_format_compatibility_bitfield2format().  The function
can return NULL if the bitfield was not associated with a function.

* Deleted dead code iax2_codec_pref_getsize() and
iax2_codec_pref_setsize().

* Made iax2_parse_allow_disallow() and iax2_codec_pref_string() call
iax2_codec_pref_to_cap() instead of inlining it.

* Made IAX_CAPABILITY_MEDBANDWIDTH, IAX_CAPABILITY_LOWBANDWIDTH, and
IAX_CAPABILITY_LOWFREE constants again as they were in Asterisk v1.8.

* Renamed prefs to prefs_global so it won't get confused with the local
pref versions.

* Fixed too small buffer in handle_cli_iax2_show_peer().

* Fixed ast_cli() calls in handle_cli_iax2_show_peer() to output complete
lines.

* Changed struct create_addr_info.prefs to be struct iax2_codec_pref as an
optimization so iax2_request() and iax2_call() do less work.

* Fixed a potential deadlock in ast_iax2_new() on an off-nominal path when
the pbx could not get started.

* Made set_config() setup a local prefs list along side the local
capability format bitfield.  Once the config is loaded, then the local
copies are put into the global versions.

* Fix unininialized codec_buf in function_iaxpeer().

ASTERISK-24150 #close
Reported by: Scott Griepentrog

Review: https://reviewboard.asterisk.org/r/3890/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420364 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agoStasis: Convey transfer information to applications
Kinsey Moore [Thu, 7 Aug 2014 15:30:19 +0000 (15:30 +0000)]
Stasis: Convey transfer information to applications

This fixes a class of issues where Stasis applications were not made
aware that their channels were being manipulated or replaced by
external entitiessuch as transfers, AMI commands, or dialplan
applications such as Bridge(). Inconsistent information such as
StasisEnd events with unknown channels as a result of masquerades has
also been corrected. To accomplish these fixes, several new fields
were added to blind and attended transfer messages as well as
StasisStart and BridgeAttendedTransfer Stasis events.

ASTERISK-23941 #close
Review: https://reviewboard.asterisk.org/r/3865/
Review: https://reviewboard.asterisk.org/r/3857/
Review: https://reviewboard.asterisk.org/r/3852/
Review: https://reviewboard.asterisk.org/r/3816/
Review: https://reviewboard.asterisk.org/r/3731/
Review: https://reviewboard.asterisk.org/r/3729/
Review: https://reviewboard.asterisk.org/r/3728/
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Merged revisions 420325 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420338 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_pjsip_publish_asterisk: Add support for exchanging device and mailbox state using...
Joshua Colp [Thu, 7 Aug 2014 14:37:26 +0000 (14:37 +0000)]
res_pjsip_publish_asterisk: Add support for exchanging device and mailbox state using SIP.

This module uses the inbound and outbound PUBLISH support to exchange device and mailbox
state between Asterisk instances. Each instance is configured to publish to the other and
requires no intermediary server. The functionality provided is similar to the XMPP and
Corosync support.

Review: https://reviewboard.asterisk.org/r/3780/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420315 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agores_pjsip_outbound_publish: Add module which provides outbound PUBLISH support.
Joshua Colp [Thu, 7 Aug 2014 14:35:09 +0000 (14:35 +0000)]
res_pjsip_outbound_publish: Add module which provides outbound PUBLISH support.

This module implements the core parts required for doing outbound PUBLISH.
It takes care of configuration, lifetime management, and authentication.
Additional modules implement the specific events that are published.

Review: https://reviewboard.asterisk.org/r/3780/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420314 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agopbx: Filter out pattern matching hints in responses sent to ExtensionStateList
Matthew Jordan [Thu, 7 Aug 2014 14:17:54 +0000 (14:17 +0000)]
pbx: Filter out pattern matching hints in responses sent to ExtensionStateList

Hints that are a pattern match are technically stored in the hint container in
the same fashion as concrete implementations of hints. The pattern matching
hints, however, are not "real" in the sense that things can subscribe to them:
rather, they are stored in the hints container so that when a subscription is
made a "real" hint can be generated for the subscription if one does not yet
exist. The extension state core takes care of this correctly by matching
against non-pattern matching extensions prior to pattern matching extensions.

Because of this, however, the ExtensionStateList AMI action was returning
pattern matching hints when executed. These hints are meaningless from the
perspective of AMI clients: their state will never change, they cannot be
subscribed to, and events would never normally be generated from them. As such,
we now filter these out of the response.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420309 65c4cc65-6c06-0410-ace0-fbb531ad65f3

5 years agobuild_tools: Skip managerEvent combining for AMI action responses
Matthew Jordan [Thu, 7 Aug 2014 03:04:49 +0000 (03:04 +0000)]
build_tools: Skip managerEvent combining for AMI action responses

AMI action responses can (and will) reference AMI events that they return.
These event references and definitions should not be combined with AMI events
raised elsewhere in the code, as they are specifically tied to the AMI action
that raised them.

ASTERISK-24156 #close
Reported by: Rusty Newton

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420289 65c4cc65-6c06-0410-ace0-fbb531ad65f3