asterisk/asterisk.git
6 years agores/res_pjsip_outbound_registration: Initialize auth_reject_permanent parameter
Matthew Jordan [Tue, 7 Oct 2014 17:44:36 +0000 (17:44 +0000)]
res/res_pjsip_outbound_registration: Initialize auth_reject_permanent parameter

Prior to this patch, the auth_reject_permanent parameter was not initialized on
the registration client state, leading to the parameter being disabled
regardless of the value specified in pjsip.conf.

This patch initialized the setting on the registration client state to the
provided configuration value.

ASTERISK-24398 #close
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6 years agores/res_pjsip_pubsub: Fix typo in WARNING message
Matthew Jordan [Tue, 7 Oct 2014 14:09:47 +0000 (14:09 +0000)]
res/res_pjsip_pubsub: Fix typo in WARNING message
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6 years agomessage: Don't close an AMI connection on SendMessage action error
Matthew Jordan [Mon, 6 Oct 2014 18:39:54 +0000 (18:39 +0000)]
message: Don't close an AMI connection on SendMessage action error

If SendMessage encounters an error (such as incorrect input provided to the
action), it will currently return -1. Actions should only return -1 if the
connection to the AMI client should be closed. In this case, SendMessage
causing the client to disconnect is inappropriate.

This patch causes the action to return 0, which simply causes the action to
fail.

Review: https://reviewboard.asterisk.org/r/4024

ASTERISK-24354 #close
Reported by: Peter Katzmann
patches:
  sendMessage.patch uploaded by Peter Katzmann (License 5968)
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6 years agofeatures.c: Fix lingering channel ref while Bridge() application is active.
Richard Mudgett [Mon, 6 Oct 2014 15:41:32 +0000 (15:41 +0000)]
features.c: Fix lingering channel ref while Bridge() application is active.

Using the Bridge application to bridge a channel that is executing an
applicaiton such as Wait results in a lingering Surrogate channel in the
CLI "core show channels" output even though it has already hungup.

* Fix bridge_exec() to not hold onto the current_dest_chan ref once it has
been put into the bridge.

* Eliminated bridge_exec()'s use of RAII_VAR().

ASTERISK-24224 #close
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/4041/
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6 years agosdp_srtp: Add new lines to some WARNING messages
Matthew Jordan [Mon, 6 Oct 2014 12:39:03 +0000 (12:39 +0000)]
sdp_srtp: Add new lines to some WARNING messages
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6 years agores_pjsip/pjsip_options: Do not 404 an OPTIONS request not sent to an endpoint
Matthew Jordan [Mon, 6 Oct 2014 01:01:43 +0000 (01:01 +0000)]
res_pjsip/pjsip_options: Do not 404 an OPTIONS request not sent to an endpoint

An OPTIONS request that is sent to Asterisk but not to a specific endpoint is
currently sent a 404 in response. This is because, not surprisingly, an empty
extension is never going to be found in the dialplan.

This patch makes it so that we only attempt to look up the endpoint in the
dialplan if it is specified in the OPTIONS request URI.

#SIPit31

ASTERISK-24370 #close
Reported by: Matt Jordan
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6 years agopjsip/dialplan_functions: Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels
Matthew Jordan [Mon, 6 Oct 2014 00:53:37 +0000 (00:53 +0000)]
pjsip/dialplan_functions: Handle PJSIP_MEDIA_OFFER called on non-PJSIP channels

Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health.
It will treat the channels as a PJSIP channel, eventually hitting an ao2 error,
FRACKing on assertion error, and quite likely crashing.

This patch adds checks to the read/write callbacks that ensure that the channel
technology is of type 'PJSIP' before attempting to operate on the channel.

#SIPit31

ASTERISK-24382 #close
Reported by: Matt Jordan
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6 years agores_pjsip: Prevent crashes when PJPROJECT presents an rdata with no message
Matthew Jordan [Mon, 6 Oct 2014 00:31:48 +0000 (00:31 +0000)]
res_pjsip: Prevent crashes when PJPROJECT presents an rdata with no message

When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable
transport, it is possible (although it shouldn't occur) for pjproject to pass
up an rdata object with a NULL msg in the msg_info. Needless to say, things
that attempt to dereference this are in for a rough ride.

In particular, this caused crashes in three different locations, all of which
are 'low level' enough to intercept an rdata object early in processing:

(1) res_pjsip_logger
(2) res_hep_pjsip
(3) res_pjsip/distributor

Anything that can intercept an rdata object before res_pjsip/distributor should
be defensive when looking at the received packet.

#SIPit31

ASTERISK-24369 #close
Reported by: Matt Jordan
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6 years agores/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptions
Matthew Jordan [Mon, 6 Oct 2014 00:13:58 +0000 (00:13 +0000)]
res/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptions

A subscription that has been persisted can - for various reasons - fail to be
re-created on startup. This patch resolves a number of crashes that occurred
when a subscription cannot be re-created on several off-nominal paths.

#SIPit31

ASTERISK-24368 #close
Reported by: Matt Jordan
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6 years agoRelease AMI connections on shutdown.
Corey Farrell [Sun, 5 Oct 2014 00:49:45 +0000 (00:49 +0000)]
Release AMI connections on shutdown.

ASTERISK-24378 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4037/
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6 years agochan_motif: Correct last commit to use ao2_cleanup to free format cap
Corey Farrell [Sun, 5 Oct 2014 00:15:43 +0000 (00:15 +0000)]
chan_motif: Correct last commit to use ao2_cleanup to free format cap

This fix applies to 13 and trunk.

ASTERISK-24384 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4043/
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6 years agochan_motif: Release format capabilities and config on module load error
Corey Farrell [Sun, 5 Oct 2014 00:02:39 +0000 (00:02 +0000)]
chan_motif: Release format capabilities and config on module load error

ASTERISK-24384 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4043/
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6 years agores_pjsip: Fix XML typo and update CHANGES.
Richard Mudgett [Fri, 3 Oct 2014 21:58:03 +0000 (21:58 +0000)]
res_pjsip: Fix XML typo and update CHANGES.

ASTERISK-24199
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6 years agoaudiohooks: Reevaluate the bridge technology when an audiohook is added or removed.
Richard Mudgett [Fri, 3 Oct 2014 19:42:54 +0000 (19:42 +0000)]
audiohooks: Reevaluate the bridge technology when an audiohook is added or removed.

Adding a mixmonitor to a channel causes the bridge to change technologies
from native to simple_bridge so the call can be recorded.  However, when
the mixmonitor is stopped the bridge does not switch back to the native
technology.

* Added unbridge requests to reevaluate the bridge when a channel
audiohook is removed.

* Moved the unbridge request into ast_audiohook_attach() ensure that the
bridge reevaluates whenever an audiohook is attached.  This simplified the
mixmonitor and chan_spy start code as well.

* Added defensive code to stop_mixmonitor_full() in case additional
arguments are ever added to the StopMixMonitor application.

* Made ast_framehook_detach() not do an unbridge request if the framehook
does not exist.

* Made ast_framehook_list_fixup() do an unbridge request if there are any
framehooks.  Also simplified the loop.

ASTERISK-24195 #close
Reported by: Jonathan Rose

Review: https://reviewboard.asterisk.org/r/4046/
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6 years agoapp_queue: Add dialplan function to get the channel name at the specified position...
Richard Mudgett [Fri, 3 Oct 2014 18:54:53 +0000 (18:54 +0000)]
app_queue: Add dialplan function to get the channel name at the specified position in a queue.

The QUEUE_GET_CHANNEL function returns the caller's channel name at the
specified position in a queue.

QUEUE_GET_CHANNEL(<queuename>[,<position>])

The queue position parameter defaults to 1 if not specified.

Noop(${QUEUE_GET_CHANNEL(queuename, 2)})
"SIP/peer-00000002", if queue exist and have at least 2 callers

Noop(${QUEUE_GET_CHANNEL(queuename, 1)})
Noop(${QUEUE_GET_CHANNEL(queuename)})
"SIP/peer-00000000", if queue exist and have at least 1 caller

ASTERISK-24365 #close
Reported by: Kristian Hogh
Patches:
      queue_get_firstchannel.patch (license #6639) patch uploaded by Kristian Hogh
      rb4035.patch (license #6639) patch uploaded by Kristian Hogh
      Patch morphed from QUEUE_GET_FIRSTCHANEL to the more general QUEUE_GET_CHANNEL
      on reviewbord.

Review: https://reviewboard.asterisk.org/r/4035/

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6 years agochan_pjsip: Fix deadlock when masquerading PJSIP channels.
Richard Mudgett [Fri, 3 Oct 2014 17:47:42 +0000 (17:47 +0000)]
chan_pjsip: Fix deadlock when masquerading PJSIP channels.

Performing a directed call pickup resulted in a deadlock when PJSIP
channels were involved.

A masquerade needs to hold onto the channel locks while it swaps channel
information between the two channels involved in the masquerade.  With
PJSIP channels, the fixup routine needed to push a fixup task onto the
PJSIP channel's serializer.  Unfortunately, if the serializer was also
processing a task that needed to lock the channel, you get deadlock.

* Added a new control frame that is used to notify the channels that a
masquerade is about to start and when it has completed.

* Added the ability to query taskprocessors if the current thread is the
taskprocessor thread.

* Added the ability to suspend/unsuspend the PJSIP serializer thread so a
masquerade could fixup the PJSIP channel without using the serializer.

ASTERISK-24356 #close
Reported by: rmudgett

Review: https://reviewboard.asterisk.org/r/4034/
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6 years agosorcery: Prevent SEGV in sorcery_wizard_create when there's no create function
George Joseph [Fri, 3 Oct 2014 15:55:57 +0000 (15:55 +0000)]
sorcery: Prevent SEGV in sorcery_wizard_create when there's no create function

When you call ast_sorcery_create() you don't necessarily know which wizard is
going to be invoked.  If it happens to be a wizard like 'config' that doesn't
have a 'create' virtual function you get a segfault in the
sorcery_wizard_create callback.  This patch catches the null function pointer,
does an ast_assert, and logs an error.

Review: https://reviewboard.asterisk.org/r/4044/
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6 years agoPJSIP: Restore functional default for callerid_privacy
Kinsey Moore [Fri, 3 Oct 2014 13:59:09 +0000 (13:59 +0000)]
PJSIP: Restore functional default for callerid_privacy

The pjsip config option default fixups from r424263 altered the
functional default from "allowed_not_screened" to "allowed". This
change restores the functional default value when none is provided.
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6 years agoManager: Add missing fields and documentation for CoreShowChannels
Kinsey Moore [Fri, 3 Oct 2014 13:33:11 +0000 (13:33 +0000)]
Manager: Add missing fields and documentation for CoreShowChannels

This corrects some issues introduced in the responses to the
CoreShowChannels AMI command as well as adding documentation for the
responses. The command in Asterisk 12 was missing the following fields:
Duration, Application, ApplicationData, and BridgedChannel and
BridgedUniqueID (replaced with BridgeId).

ASTERISK-24262 #close
Reported by: Mitch Claborn
Review: https://reviewboard.asterisk.org/r/4040/
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6 years agores_pjsip: Make transport cipher option accept a comma separated list of cipher names.
Richard Mudgett [Thu, 2 Oct 2014 21:55:37 +0000 (21:55 +0000)]
res_pjsip: Make transport cipher option accept a comma separated list of cipher names.

Improvements to the res_pjsip transport cipher option.

* Made the cipher option accept a comma separated list of OpenSSL cipher
names.  Users of realtime will be glad if they have more than one name to
list.

* Added the CLI command 'pjsip list ciphers' so a user can know what
OpenSSL names are available for the cipher option.

* Updated the cipher option online XML documentation to specify what is
expected for the value.

* Updated pjsip.conf.sample to not indicate that ALL is acceptable since
ALL does not imply a preference order for the ciphers and PJSIP does not
simply pass the string to OpenSSL for interpretation.

ASTERISK-24199 #close
Reported by: Joshua Colp

Review: https://reviewboard.asterisk.org/r/4018/
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6 years agoAlembic: Add enumerator value to sippeers -> directmedia - 'outgoing'
Jonathan Rose [Thu, 2 Oct 2014 20:23:38 +0000 (20:23 +0000)]
Alembic: Add enumerator value to sippeers -> directmedia - 'outgoing'

The 'outgoing' value was left off of the enumerator when first creating the
column. This patch adds it, and should gracefully upgrade keeping the existing
data in tact.

ASTERISK-23781 #close
Reported by: Stephen More
Review: https://reviewboard.asterisk.org/r/4013/
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6 years agochan_pjsip: Fix an assertion for channels that lack formats on creation
Jonathan Rose [Thu, 2 Oct 2014 15:33:50 +0000 (15:33 +0000)]
chan_pjsip: Fix an assertion for channels that lack formats on creation

ASTERISK-24222 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4017/
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6 years agores_pjsip: document use of rewrite_contact in sample conf
Scott Griepentrog [Thu, 2 Oct 2014 13:36:01 +0000 (13:36 +0000)]
res_pjsip: document use of rewrite_contact in sample conf

Without setting rewrite_contact, an invite to an endpoint
behind NAT will not reach it - unless the endpoint itself
uses STUN or TURN to discover it's public URI.  Thus, the
use of this should be in the sample documentation.

Review: https://reviewboard.asterisk.org/r/4036/
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6 years agores_hep: Release allocation reference to configuration.
Corey Farrell [Wed, 1 Oct 2014 20:37:31 +0000 (20:37 +0000)]
res_hep: Release allocation reference to configuration.

ASTERISK-24362 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/4026/
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6 years agores_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash.
Joshua Colp [Wed, 1 Oct 2014 16:39:45 +0000 (16:39 +0000)]
res_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash.

During the latest update to DTLS-SRTP support the ability to configure
the hash used for fingerprints was added. This gave us two supported ones:
SHA-1 and SHA-256. The default was accordingly updated to SHA-256.
Unfortunately this configuration ability was not exposed within res_pjsip.
This change adds a dtls_fingerprint option that controls it.

#SIPit31
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6 years agores_pjsip_sdp_rtp: Accept DTLS attributes in top level, not just media session.
Joshua Colp [Wed, 1 Oct 2014 16:20:40 +0000 (16:20 +0000)]
res_pjsip_sdp_rtp: Accept DTLS attributes in top level, not just media session.

#SIPit31
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6 years agoPJSIP: Handle defaults properly
Kinsey Moore [Wed, 1 Oct 2014 12:28:05 +0000 (12:28 +0000)]
PJSIP: Handle defaults properly

This updates the code behind PJSIP configuration options with custom
handlers to deal with the assigned default values properly where it
makes sense and adjusting the default value where it doesn't. Before
applying this patch, there were several cases where the default value
for an option would prevent that config section from loading properly.

Reported by: Thomas Thompson
Review: https://reviewboard.asterisk.org/r/4019/
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6 years agoPJSIP: Force transport on contact rewrite
Kinsey Moore [Wed, 1 Oct 2014 12:15:56 +0000 (12:15 +0000)]
PJSIP: Force transport on contact rewrite

If contact rewriting is enabled but the contact differs in transport
from what is actually being used, messages after the initial INVITE
transaction can be sent to an incorrect transport/port combination. In
the case where this bug occurred the remote party never received a BYE
since it was sent to the remote party's TCP port over UDP.

Review: https://reviewboard.asterisk.org/r/4032/
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6 years agochan_sip: Simplify some unref code by removing unlink_peer_from_tables.
Walter Doekes [Wed, 1 Oct 2014 10:10:41 +0000 (10:10 +0000)]
chan_sip: Simplify some unref code by removing unlink_peer_from_tables.

ASTERISK-22945 #related
Reported by: ibercom
Patches:
  asterisk11-chan_sip-simplifies.patch uploaded by ibercom (License #6599)
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6 years agochan_sip: Remove excess ref of realtime peer before sip_poke_peer.
Walter Doekes [Wed, 1 Oct 2014 09:55:10 +0000 (09:55 +0000)]
chan_sip: Remove excess ref of realtime peer before sip_poke_peer.

The peer is referenced at the end of sip_poke_peer, it should not get
an extra ref before the call to sip_poke_peer. This fixes a memory
leak.

ASTERISK-22945 #close
Reported by: ibercom
Tested by: Yuriy Gorlichenko
Patches:
  asterisk11.patch uploaded by ibercom (License #6599)

Review: https://reviewboard.asterisk.org/r/4031/
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6 years agores_pjsip_sdp_rtp: Don't place an extra whitespace before 'rport' and don't put IPv6...
Joshua Colp [Tue, 30 Sep 2014 11:42:00 +0000 (11:42 +0000)]
res_pjsip_sdp_rtp: Don't place an extra whitespace before 'rport' and don't put IPv6 addresses in brackets.

#SIPit31
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6 years agores_rtp_asterisk: Ensure that the base and mapped address for candidates is present...
Joshua Colp [Tue, 30 Sep 2014 11:36:14 +0000 (11:36 +0000)]
res_rtp_asterisk: Ensure that the base and mapped address for candidates is present in SDP.

This change fixes an issue where ICE candidates put into the SDP did not contain
the 'raddr' and 'rport' information for server reflexive and relay candidates.

#SIPit31
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6 years agopjsip_cli: Suppress header print on error or no objects
George Joseph [Mon, 29 Sep 2014 22:00:38 +0000 (22:00 +0000)]
pjsip_cli: Suppress header print on error or no objects

If there's an error on the pjsip command line or there are no objects, don't
print the column headers.

ASTERISK-24350 #close
Reported-by: Brad Latus
Tested-by: George Joseph
Tested-by: Brad Latus

Review: https://reviewboard.asterisk.org/r/4025/
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6 years agoautosupport: Fix bashism.
Walter Doekes [Mon, 29 Sep 2014 21:32:10 +0000 (21:32 +0000)]
autosupport: Fix bashism.

'==' is bashism (bashspecific, fails when dash is /bin/sh). Anyway, a
'case' works better there.

Originally committed in r375059 and r375060 on 2012-10-16 21:13:08.

ASTERISK-20567 #close
Reported by: Tzafrir Cohen
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6 years agoSimplify UUID generation in several places.
Richard Mudgett [Mon, 29 Sep 2014 21:18:54 +0000 (21:18 +0000)]
Simplify UUID generation in several places.

Replace code using ast_uuid_generate() with simpler and faster code using
ast_uuid_generate_str().  The new code avoids a malloc(), free(), and
copy.
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6 years agothreadpool.c: Minor cleanup fixes.
Richard Mudgett [Mon, 29 Sep 2014 20:28:24 +0000 (20:28 +0000)]
threadpool.c: Minor cleanup fixes.

* Fix threadpool_alloc() prototype.

* Add missing off-nominal NULL check of pool in threadpool_alloc().

* searializer_create() does not need to create the object with a lock as
the lock is not used.
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6 years agores_pjsip_session: Reduce SDP size by removing duplicate connection lines.
Joshua Colp [Sat, 27 Sep 2014 17:29:05 +0000 (17:29 +0000)]
res_pjsip_session: Reduce SDP size by removing duplicate connection lines.

Due to the architecture of how media streams are handled each individual
handler adds connection details (IP address) for it. The first media stream
is then used as the top level SDP connection line. In practice each
line ends up being the same so to reduce the SDP size stream-level connection
information is also added to the SDP if it differs from the top level SDP
connection line.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424077 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agores_pjsip_session: Add additional checks for delaying session refreshes.
Joshua Colp [Sat, 27 Sep 2014 12:44:38 +0000 (12:44 +0000)]
res_pjsip_session: Add additional checks for delaying session refreshes.

There are certain situations which no checks existed for which need to prevent
session refreshes. This includes sending a session refresh with SDP before SDP
negotiation has completed and sending a session refresh before the dialog itself
has been established. Checks for these have been added.

Additionally COLP related UPDATEs were including SDP when it is not needed.

Review: https://reviewboard.asterisk.org/r/4008/
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6 years agoformat_mp3: Made the get script conditionally apply patch if not already there.
Richard Mudgett [Fri, 26 Sep 2014 15:51:22 +0000 (15:51 +0000)]
format_mp3: Made the get script conditionally apply patch if not already there.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424039 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agocore: Ouch, forgot to undo a test free() in r423978.
Walter Doekes [Fri, 26 Sep 2014 15:43:04 +0000 (15:43 +0000)]
core: Ouch, forgot to undo a test free() in r423978.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@424038 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agores_fax: Fix out of bounds error in update_modem_bits().
Richard Mudgett [Fri, 26 Sep 2014 15:28:39 +0000 (15:28 +0000)]
res_fax: Fix out of bounds error in update_modem_bits().

ASTERISK-24357 #close
Reported by: Jeremy Laine
Patches:
      res_fax_bounds.patch (license #6561) patch uploaded by Jeremy Laine
  Modified patch to not use magic numbers.
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6 years agocore: Don't allow free to mean ast_free (and malloc, etc..).
Walter Doekes [Fri, 26 Sep 2014 14:41:38 +0000 (14:41 +0000)]
core: Don't allow free to mean ast_free (and malloc, etc..).

This gets rid of most old libc free/malloc/realloc and replaces them
with ast_free and friends. When compiling with MALLOC_DEBUG you'll
notice it when you're mistakenly using one of the libc variants. For
the legacy cases you can define WRAP_LIBC_MALLOC before including
asterisk.h.

Even better would be if the errors were also enabled when compiling
without MALLOC_DEBUG, but that's a slightly more invasive header
file change.

Those compiling addons/format_mp3 will need to rerun
./contrib/scripts/get_mp3_source.sh.

ASTERISK-24348 #related
Review: https://reviewboard.asterisk.org/r/4015/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423978 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agodocs: Escape unescaped minus sign in asterisk.8 manpage.
Walter Doekes [Fri, 26 Sep 2014 08:26:24 +0000 (08:26 +0000)]
docs: Escape unescaped minus sign in asterisk.8 manpage.

ASTERISK-23768 #close
Reported by: Jeremy Lainé
Patches:
  escape_manpage_hyphen.patch uploaded by Jeremy Lainé (License #6561)
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6 years agores_pjsip.c: Add missing off nominal cleanup in ast_sip_push_task_synchronous().
Richard Mudgett [Thu, 25 Sep 2014 21:03:51 +0000 (21:03 +0000)]
res_pjsip.c: Add missing off nominal cleanup in ast_sip_push_task_synchronous().

* Made memset the std struct in ast_sip_push_task_synchronous() because if
DEBUG_THREADS is enabled then uninitialized lock tracking data is used.
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6 years agomusiconhold: Add preferchannelclass=no option to prefer app class.
Walter Doekes [Thu, 25 Sep 2014 20:49:04 +0000 (20:49 +0000)]
musiconhold: Add preferchannelclass=no option to prefer app class.

The new option 'preferchannelclass' is added to musiconhold.conf. If yes
(the default) the CHANNEL(musicclass) is preferred when choosing the
hold music. If it is no, the class suggested by the application that
calls the MoH (e.g. the Queue() app) gets preferred (new behaviour).

This way you set a different hold-music from the Queue-music by setting
both the CHANNEL(musicclass) and the queue-context musicclass.

ASTERISK-24276 #close
Reported by: Kristian Høgh
Patches:
  app_override_channel_moh.patch uploaded by Kristian Høgh (License #6639)

Review: https://reviewboard.asterisk.org/r/4010/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423893 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agopjsip_options.c: Fix race condition stopping periodic out of dialog OPTIONS request.
Richard Mudgett [Wed, 24 Sep 2014 18:35:47 +0000 (18:35 +0000)]
pjsip_options.c: Fix race condition stopping periodic out of dialog OPTIONS request.

The crash on the issues is a result of an invalid transport configuration
change when asterisk is restarted.  The attempt to send the qualify
request fails and we cleaned up.  However, the callback is also called
which results in a double unref of the objects involved.

* Put a wrapper around pjsip_endpt_send_request() to detect when the
passed in callback is called because of an error so callers can know to
not cleanup.

* Made send_request_cb() able to handle repeated challenges (Up to 10).

* Fix periodic endpoint qualify OPTIONS sched deletion race by avoiding
it.  The sched entry will no longer self stop and must be externally
stopped.

* Added REF_DEBUG description tags to struct sched_data in
pjsip_options.c.

* Fix some off-nominal ref leaks in schedule_qualify(),
qualify_and_schedule().

* Reordered pjsip_options.c module start/stop code to cleanup better on
error.

ASTERISK-24295 #close
Reported by: Rogger Padilla

Review: https://reviewboard.asterisk.org/r/3954/
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6 years agochan_sip: Unref outbound proxy structure on dialog/pvt destruction.
Walter Doekes [Wed, 24 Sep 2014 08:55:02 +0000 (08:55 +0000)]
chan_sip: Unref outbound proxy structure on dialog/pvt destruction.

Make sure outbound proxy refs are always unreffed on dialog destruction.

Review: https://reviewboard.asterisk.org/r/4016/
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6 years agoMake CDR and CEL unit tests less FRACKy.
Mark Michelson [Tue, 23 Sep 2014 14:36:00 +0000 (14:36 +0000)]
Make CDR and CEL unit tests less FRACKy.

Prior to this commit, CDR and CEL tests were expected to trigger
FRACKs (i.e. assertions) due to the fact that the channels they
create have no formats on them. Some code was independently added
recently that attempts to prevent FRACKs from occurring by failing
early when attempting to set up translation paths if one or both
channels support no formats. Unfortunately, this attempt to be helpful
made the CDR and CEL tests go from simply FRACKing to outright
failing and in some cases, failing so badly as to crash Asterisk.

This commit seeks to correct past mistakes by adding the ulaw format
to channels created by the CDR and CEL unit tests. This makes setting
up translation paths succeed, eliminates previously-seen FRACKs, and
ultimately causes the unit tests to succeed again.

Review: https://reviewboard.asterisk.org/r/4014
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6 years agochan_sip: On INVITE retransmission, don't add an extra 503 response.
Walter Doekes [Mon, 22 Sep 2014 19:49:30 +0000 (19:49 +0000)]
chan_sip: On INVITE retransmission, don't add an extra 503 response.

INVITE arrives to asterisk, asterisk responds Busy(). If the INVITE is
retransmitted, asterisk would generate a 503 in addition to the 486.

Thanks Torrey Searle for providing a working regression test.

ASTERISK-24335 #close

Review: https://reviewboard.asterisk.org/r/4003/
Patches:
  retrans_486_invite.patch uploaded by Torrey Searle (License #5334)
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6 years agocli.c: Fix tab completion "module load" when MALLOC_DEBUG is enabled.
Walter Doekes [Mon, 22 Sep 2014 17:42:26 +0000 (17:42 +0000)]
cli.c: Fix tab completion "module load" when MALLOC_DEBUG is enabled.

r421600 conflicted with r155763.

ASTERISK-24348 #close
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6 years agomain/channel: Unlock channel in off-nominal path
Matthew Jordan [Sun, 21 Sep 2014 01:16:05 +0000 (01:16 +0000)]
main/channel: Unlock channel in off-nominal path

In r423414 (13) / r423415 (trunk), an API call that determines if a format
capability structure is empty was added. This returns true if the format
capability structure is completely empty or "none". A check for this was added
in channel.c's set_format call. Unfortunately, when this check was true, it
returned from the function while still holding the channel lock. This caused
the CDR unit tests - which have a tendency to create channels with no formats -
to deadlock. Whoops.

This patch unlocks the channel on the off-nominal path.
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6 years agorest-api/api-docs/events.json: Remove non-compliant 'extends' attribute
Matthew Jordan [Sat, 20 Sep 2014 23:55:45 +0000 (23:55 +0000)]
rest-api/api-docs/events.json: Remove non-compliant 'extends' attribute

Prior to the release of Swagger 1.2, the attribute 'extends' was being
promoted as a possible way to show that a particular object extends an existing
object. Instead, the Swagger specification went with the 'subTypes' attribute
in the base object. This patch removes the unsupported attribute; the object
that the offending objects proposed to extend already lists them in its
'subTypes' attribute.

ASTERISK-24300 #close
Reported by: Bradley Watkins
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6 years agorest-api/api-docs: Correct basePath in resources to match top resources file
Matthew Jordan [Sat, 20 Sep 2014 23:41:55 +0000 (23:41 +0000)]
rest-api/api-docs: Correct basePath in resources to match top resources file

The resources.json file that defines the resource JSON files used with ARI
references a basePath of 'http://localhost:8088/ari'. This does not match what
is defined in the resource files themselves, 'http://localhost:8088/stasis'.
The correct base path is the one that includes 'ari' in the URL; this patch
updates the various resource JSON files to have the correct basePath.

ASTERISK-24339 #close
Reported by: Bradley Watkins
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6 years agores_pjsip_notify: Fix crash on unload/load and don't say the module doesn't exist...
Joshua Colp [Fri, 19 Sep 2014 19:51:50 +0000 (19:51 +0000)]
res_pjsip_notify: Fix crash on unload/load and don't say the module doesn't exist on reload.

When unloading the module did not unregister the CLI commands causing a crash upon
load when they were registered again.

When reloading the module the return value from the config options framework was not
checked to determine if an error occurred or not. This caused a message to be output
saying the module did not exist when reloading if no changes were present.

AST-1433 #close
AST-1434 #close
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6 years agores_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated.
Richard Mudgett [Fri, 19 Sep 2014 17:16:32 +0000 (17:16 +0000)]
res_pjsip_sdp_rtp.c: Fix native formats containing formats that were not negotiated.

Outgoing PJSIP calls can result in non-negotiated formats listed in the
channel's native formats if video formats are listed in the endpoint's
configuration.  The resulting call could then use a non-negotiated format
resulting in one way audio.

* Simplified the update of session->req_caps in set_caps().  Why do
something in five steps when only one is needed?

AFS-162 #close

Review: https://reviewboard.asterisk.org/r/4000/
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6 years agoStasis_channels: Resolve unfinished Dials when doing masquerades
Jonathan Rose [Fri, 19 Sep 2014 15:54:20 +0000 (15:54 +0000)]
Stasis_channels: Resolve unfinished Dials when doing masquerades

Masquerades into channels that are in the dialing state don't end their dial
and this goes against the model for things like CDRs and generating Dial end
manager actions and such.

ASTERISK-24237 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/3990/
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6 years agochan_iax2: Fix a crash when using chan_iax2 jitterbuffer settings
Jonathan Rose [Fri, 19 Sep 2014 15:11:06 +0000 (15:11 +0000)]
chan_iax2: Fix a crash when using chan_iax2 jitterbuffer settings

Caused by format changes in Asterisk 13

ASTERISK-24265 #close
Reported by: Dafi Ni
Review: https://reviewboard.asterisk.org/r/3999/
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6 years agoPJSIP: Prevent T38 framehook being put on wrong channel
Kinsey Moore [Fri, 19 Sep 2014 12:50:08 +0000 (12:50 +0000)]
PJSIP: Prevent T38 framehook being put on wrong channel

This change gives framehooks a reverse-direction masquerade callback in
addition to chan_fixup_cb similar to the callback added to datastores
to handle the same situation. The new callback provides the same
parameters as the fixup callback, but is called on the new channel's
framehooks before moving framehooks from the old channel to the new
channel. This gives the framehooks an oppurtunity to decide whether
they should remain on the new channel or be removed.

This new callback is used to prevent the PJSIP T.38 framehook from
remaining on a masqueraded channel if the new channel is not also a
PJSIP channel. This was causing a crash when a local channel was
masqueraded into a PJSIP channel and the framehook was executed on the
local channel since the channel's tech private data was not structured
as expected.

Review: https://reviewboard.asterisk.org/r/4001/
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6 years agores_pjsip: Don't require a password when doing userpass authentication.
Sean Bright [Thu, 18 Sep 2014 19:31:03 +0000 (19:31 +0000)]
res_pjsip: Don't require a password when doing userpass authentication.

An empty password is valid for username/password authentication so we should
allow password to be empty/not supplied.

Review: https://reviewboard.asterisk.org/r/3988
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423483 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoutils: Create ast_strsep function that ignores separators inside quotes
George Joseph [Thu, 18 Sep 2014 19:23:39 +0000 (19:23 +0000)]
utils: Create ast_strsep function that ignores separators inside quotes

This function acts like strsep with three exceptions...
* The separator is a single character instead of a string.
* Separators inside quotes are treated literally instead of like separators.
* You can elect to have leading and trailing whitespace and quotes
stripped from the result and have '\' sequences unescaped.

Like strsep, ast_strsep maintains no internal state and you can call it
recursively using different separators on the same storage.

Also like strsep, for consistent results, consecutive separators are not
collapsed so you may get an empty string as a valid result.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3989/
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6 years agoAdd subscription state test events.
Mark Michelson [Thu, 18 Sep 2014 18:56:54 +0000 (18:56 +0000)]
Add subscription state test events.

These are needed for a set of batched notification RLS tests that are
about to be committed to the testsuite.

Review: https://reviewboard.asterisk.org/r/3967
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6 years agores_pjsip_endpoint_identifier_ip: Fix parsing of match value with CIDR
Jonathan Rose [Thu, 18 Sep 2014 17:22:03 +0000 (17:22 +0000)]
res_pjsip_endpoint_identifier_ip: Fix parsing of match value with CIDR

Also fixes comma separates match lists

ASTERISK-24290 #close
Reported by: Ray Crumrine
Review: https://reviewboard.asterisk.org/r/3995/
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6 years agobridge_softmix.c: Made use ao2_replace() instead of the inline equivalent.
Richard Mudgett [Thu, 18 Sep 2014 17:10:50 +0000 (17:10 +0000)]
bridge_softmix.c: Made use ao2_replace() instead of the inline equivalent.

* Clarified some read/write format comments.

* Fixed a doxygen tag typo.
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6 years agoastobj2.c/refcounter.py: Fix to deal with invalid object refs.
Richard Mudgett [Thu, 18 Sep 2014 16:56:40 +0000 (16:56 +0000)]
astobj2.c/refcounter.py: Fix to deal with invalid object refs.

* Make astob2 REF_DEBUG output an invalid object line when an invalid ao2
object ref/unref is attempted.  This is similar to the
constructor/destructor lines.

* Fixed refcounter.py to handle skewed objects that have
constructor/destructor states.

* Made refcounter.py highlight the invalid ao2 object refs by putting them
in their own section of the processed output file.

* Made refcounter.py highlight unreffing an object by more than one that
results in a negative ref count and the object being destroyed.  The
abnormally destroyed object is reported in the invalid and finalized
object sections of the output.

Review: https://reviewboard.asterisk.org/r/3971/
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Merged revisions 423416 from http://svn.asterisk.org/svn/asterisk/branches/12
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6 years agoAdd API call to determine if format capability structure is "empty".
Mark Michelson [Thu, 18 Sep 2014 16:38:26 +0000 (16:38 +0000)]
Add API call to determine if format capability structure is "empty".

Empty here means that there are no formats in the format_cap structure
or the only format in it is the "none" format.

I've added calls to check the emptiness of a format_cap in a few places
in order to short-circuit operations that would otherwise be pointless
as well as to prevent some assertions from being triggered in cases
where channels with no formats are used.
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6 years agores_fax_spandsp: Properly handle cleanup before starting FAXes.
Mark Michelson [Thu, 18 Sep 2014 16:24:48 +0000 (16:24 +0000)]
res_fax_spandsp: Properly handle cleanup before starting FAXes.

If faxing fails at a very early stage, then it is possible for
us to pass a NULL t30 state pointer to spandsp, which spandsp
is none too pleased with.

This patch ensures that we pass the correct pointer to spandsp
in the situation where we have not yet set our local t30 state
pointer.

ASTERISK-24301 #close
Reported by Matt Jordan
Patches:
ASTERISK-24301-fax.diff Uploaded by Mark Michelson (License #5049)
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6 years agores_pjsip_pubsub: Add some type safety when generating NOTIFY bodies.
Mark Michelson [Thu, 18 Sep 2014 16:09:25 +0000 (16:09 +0000)]
res_pjsip_pubsub: Add some type safety when generating NOTIFY bodies.

res_pjsip_pubsub has two separate checks that it makes when a SUBSCRIBE
arrives.
* It checks that there is a subscription handler for the Event
* It checks that there are body generators for the types in the Accept header

The problem is, there's nothing that ensures that these two things will
actually mesh with each other. For instance, Asterisk will accept a subscription
to MWI that accepts pidf+xml bodies. That doesn't make sense.

With this commit, we add some type information to the mix. Subscription
handlers state they generate data of type X, and body generators state
that they consume data of type X. This way, Asterisk doesn't end up in
some hilariously mismatched situation like the one in the previous paragraph.

ASTERISK-24136 #close
Reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/3877
Review: https://reviewboard.asterisk.org/r/3878
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423350 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agores_pjsip: ami: Fix error in AMI output when an endpoint has no transport
George Joseph [Thu, 18 Sep 2014 15:14:38 +0000 (15:14 +0000)]
res_pjsip: ami: Fix error in AMI output when an endpoint has no transport

When no transport is associated to an endpoint, the AMI output for
PJSIPShowEndpoint indicates an error instead of silently ignoring the
missing transport.

This patch causes the error to appear only if a transport was specified
on the endpoint and the transport doesn't exist.  It also fixes an issue
with counting the objects that were actually found.

ASTERISK-24161 #close
ASTERISK-24331 #close
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3998/
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6 years agoOnly install dahdi_span_config_hook if DAHDI is enabled
David M. Lee [Thu, 18 Sep 2014 15:07:12 +0000 (15:07 +0000)]
Only install dahdi_span_config_hook if DAHDI is enabled

This patch changes the install to only install the hook script if
DAHDI is enabled. It also adds the script to the uninstall task, and
moves the DAHDI_UDEV_HOOK_DIR variable so that it's not between the
_MAKEOPTS variables and their comment.

This allows installs which specify a --prefix to work normally, as
long as they don't enable DAHDI.

Review: https://reviewboard.asterisk.org/r/3972/
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6 years agoconfig: bug: Fix SEGV in ast_category_insert when matching category isn't found
George Joseph [Thu, 18 Sep 2014 14:46:12 +0000 (14:46 +0000)]
config: bug: Fix SEGV in ast_category_insert when matching category isn't found

If you call ast_category_insert with a match category that doesn't exist, the
list traverse runs out of 'next' categories and you get a SEGV.  This patch
adds check for the end-of-list condition and changes the signature to return
an int for success/failure indication instead of a void.

The only consumer of this function is manager and it was also changed to use
the return value.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3993/
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6 years agores_rtp_asterisk: Ensure that the thread terminating pj stuff is registered.
Joshua Colp [Wed, 17 Sep 2014 18:06:06 +0000 (18:06 +0000)]
res_rtp_asterisk: Ensure that the thread terminating pj stuff is registered.
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6 years agopbx/Makefile: Revert r423237
Matthew Jordan [Tue, 16 Sep 2014 22:46:29 +0000 (22:46 +0000)]
pbx/Makefile: Revert r423237

This patch was supposed to go into a team branch, but I was a bit fast on the
gun before 'svn switch' had apparently moved the target branch over.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423238 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd some pbx python stuff
Matthew Jordan [Tue, 16 Sep 2014 22:42:09 +0000 (22:42 +0000)]
Add some pbx python stuff

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423237 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoMultiple revisions 423209,423212
Joshua Colp [Tue, 16 Sep 2014 21:06:22 +0000 (21:06 +0000)]
Multiple revisions 423209,423212

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  r423209 | file | 2014-09-16 17:35:34 -0300 (Tue, 16 Sep 2014) | 8 lines

  res_rtp_asterisk: Fix building when pjproject is not used.
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  r423212 | file | 2014-09-16 18:03:59 -0300 (Tue, 16 Sep 2014) | 10 lines

  res_rtp_asterisk: Fix 100% CPU usage due to timer heap thread spinning.

  Side note: I need a vacation.
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6 years agoVoicemail: get correct duration when copying file to vm
Scott Griepentrog [Tue, 16 Sep 2014 16:33:53 +0000 (16:33 +0000)]
Voicemail: get correct duration when copying file to vm

Changes made during format improvements resulted in the
recording to voicemail option 'm' of the MixMonitor app
writing a zero length duration in the msgXXXX.txt file.

This change introduces a new function ast_ratestream(),
which provides the sample rate of the format associated
with the stream, and updates the app_voicemail function
for ast_app_copy_recording_to_vm to calculate the right
duration.

Review: https://reviewboard.asterisk.org/r/3996/
ASTERISK-24328 #close
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6 years agores_pjsip_session: Fix usage of wrong memory pool when creating local SDP.
Joshua Colp [Tue, 16 Sep 2014 12:12:36 +0000 (12:12 +0000)]
res_pjsip_session: Fix usage of wrong memory pool when creating local SDP.
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6 years agores_rtp_asterisk: Fix a myriad of TURN client issues.
Joshua Colp [Tue, 16 Sep 2014 11:12:30 +0000 (11:12 +0000)]
res_rtp_asterisk: Fix a myriad of TURN client issues.

1. The number of file descriptors an ioqueue instance can handle is fixed, so we
now spawn the required number to handle the load.
2. Our transport identifiers were exceeding the range supported by pjnath.
3. The TURN client did not set up client binding causing needless bandwidth usage.
4. The code no longer updates address information on each packet.
5. STUN traffic was getting looped back to Asterisk instead of going through the
TURN server.
6. Synchronization now ensures things are completely setup or destroyed.
7. Logging now reflects the target the TURN server is sending to/receiving from
on our behalf.

ASTERISK-23577 #close
Reported by: Jay Jideliov

ASTERISK-23634 #close
Reported by: Roman Skvirsky

Review: https://reviewboard.asterisk.org/r/3982/
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6 years agocontrib: Fix verifyi typo in alembic DB script ps_transport table.
Walter Doekes [Mon, 15 Sep 2014 10:50:11 +0000 (10:50 +0000)]
contrib: Fix verifyi typo in alembic DB script ps_transport table.

Reported by: Zogot (on IRC)
Patches:
  tmp.diff uploaded by Zogot, cleaned up by me.
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6 years agochan_sip: Clarify that sipdebug=yes cannot be undone by the CLI.
Walter Doekes [Sun, 14 Sep 2014 15:54:22 +0000 (15:54 +0000)]
chan_sip: Clarify that sipdebug=yes cannot be undone by the CLI.

Document it in sip.conf.

ASTERISK-24249 #close
Reported by: Avinash Mohod

Review: https://reviewboard.asterisk.org/r/3926/
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6 years agomusiconhold: Add sort=randstart, and deprecate old stuff.
Walter Doekes [Sun, 14 Sep 2014 15:41:58 +0000 (15:41 +0000)]
musiconhold: Add sort=randstart, and deprecate old stuff.

- adds sort=randstart (next to sort=, sort=random, sort=alpha)
- combines duplicate moh option parsing code into a single function
- adds deprecationwarnings for application=r to sort randomly
- adds deprecationwarnings for random=yes to sort randomly
- removes invisible code that was supposed to stay until 1.8

The sort=randstart works like sort=alpha, except we start at a random
position.

Review: https://reviewboard.asterisk.org/r/3991/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423065 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agochan_rtp: Add unicast RTP support.
Joshua Colp [Fri, 12 Sep 2014 17:42:15 +0000 (17:42 +0000)]
chan_rtp: Add unicast RTP support.

This module supports sending both unicast and multicast RTP
to a specified target. Multicast functionality is the same as
chan_multicast_rtp was. In the case of unicast a specific
IP address and port can be specified, along with optional RTP
engine and format in the form of:

UnicastRTP/<ip address>:<port>/<engine>/<format>

This can be useful for sending a copy of a media stream to
another application for processing.

Review: https://reviewboard.asterisk.org/r/3981/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423004 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoRealtime: Fix a bug that caused realtime destroy command to crash
Jonathan Rose [Fri, 12 Sep 2014 16:19:19 +0000 (16:19 +0000)]
Realtime: Fix a bug that caused realtime destroy command to crash

Also has could affect with anything that goes through ast_destroy_realtime.
If a CLI user used the command 'realtime destroy <family>' with only a single
column/value pair, Asterisk would crash when trying to create a variable list
from a NULL value.

ASTERISK-24231 #close
Reported by: Niklas Larsson
Review: https://reviewboard.asterisk.org/r/3985/
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6 years agoRemove undocumented default behavior of ast_play_and_record_full acceptdtmf.
Mark Michelson [Thu, 11 Sep 2014 22:17:39 +0000 (22:17 +0000)]
Remove undocumented default behavior of ast_play_and_record_full acceptdtmf.

ast_play_and_record_full() has a parameter called "acceptdtmf" that is a
string of acceptable DTMF digits that may be pressed by a caller to end
and accept the recording.

ARI uses this function in order to perform recording, and it provides
options for what is passed as acceptdtmf to ast_play_and_record_full().
By default, ARI passes an empty string, with the intention that no DTMF
can be used to end the recording.

The problem is that ast_play_and_record_full() attempts to be "helpful"
by setting "#" as the acceptdtmf if an empty string or NULL pointer
has been passed in. With ARI, this results in unexpected behavior
occurring if you have attempted to intercept "#" yourself in order
to perform some other manipulation of the live recording.

This change removes the "helpful" behavior by no longer accepting
"#" as a default acceptdtmf if none is specified by the caller of
ast_play_and_record_full(). This makes the ARI scenario work as
expected.

The other callers of ast_play_and_record_full() are app_voicemail
and app_minivm, and in both cases, they pass an explicit "#" to
ast_play_and_record_full() as acceptdtmf, so they are unaffected
by this change.
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6 years agoconfig: bug: fix truncation of included config files on permissions error
George Joseph [Wed, 10 Sep 2014 16:07:04 +0000 (16:07 +0000)]
config: bug: fix truncation of included config files on permissions error

ast_config_text_file_save() currently truncates include files as they
are processed.  If a subsequent include file or the main config file has
a permissions error that prevents writing, earlier include files are left
truncated resulting in a frantic search for backups.

This patch causes ast_config_text_file_save to check for write access
on all files before it truncates any of them.

Will be applied 1.8 > trunk.

Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3986/
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6 years agopjsip/config_auth.c: Add missing whitespace to log messages.
Sean Bright [Wed, 10 Sep 2014 16:00:36 +0000 (16:00 +0000)]
pjsip/config_auth.c: Add missing whitespace to log messages.

The errors generated when validating 'auth' settings are missing a space which
makes the messages a little confusing.
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6 years agoUpdate CHANGES for CHANNEL(onhold).
Richard Mudgett [Tue, 9 Sep 2014 20:15:57 +0000 (20:15 +0000)]
Update CHANGES for CHANNEL(onhold).

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422885 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoSounds/BuildSystem: Modifications to include new releases and Japanese language.
Rusty Newton [Tue, 9 Sep 2014 20:11:10 +0000 (20:11 +0000)]
Sounds/BuildSystem: Modifications to include new releases and Japanese language.

Modifying Makefile and sounds.xml to include new core 1.4.26 and extra 1.4.15
sound prompt releases, plus the new Japanese core sound prompts contributed
by QLOOG.

ASTERISK-23324
Reported by: Kevin McCoy
Tested by: Rusty Newton
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6 years agofunc_channel: Add CHANNEL(onhold) item to get the current hold status of the channel.
Richard Mudgett [Tue, 9 Sep 2014 16:14:02 +0000 (16:14 +0000)]
func_channel: Add CHANNEL(onhold) item to get the current hold status of the channel.

It would be useful to get the current hold status of a channel.

Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for
the hold status of a channel.

ASTERISK-24038
Reported by: Matt Jordan

AFS-113 #close
Reported by: Mark Michelson

Review: https://reviewboard.asterisk.org/r/3983/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422870 65c4cc65-6c06-0410-ace0-fbb531ad65f3

6 years agoAdd note about configuring list_items on a single line.
Mark Michelson [Mon, 8 Sep 2014 18:04:01 +0000 (18:04 +0000)]
Add note about configuring list_items on a single line.
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6 years agoAdd sample configuration for resource lists.
Mark Michelson [Mon, 8 Sep 2014 17:53:33 +0000 (17:53 +0000)]
Add sample configuration for resource lists.

On review /r/3977, it was recommended to note in the
sample configuration about the size limitation for
resource lists. However, since there was no section in
the sample configuration at all for resource list
subscriptions, I decided to make a separate commit
where I have added the necessary sample configuration
as well as the size limitation warning.
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6 years agoPre-allocate transmission data buffer for RLS NOTIFY requests.
Mark Michelson [Mon, 8 Sep 2014 17:35:02 +0000 (17:35 +0000)]
Pre-allocate transmission data buffer for RLS NOTIFY requests.

PJSIP, unless a constant is modified at compilation time, limits
SIP requests to 4000 bytes. Full-state RLS notifications can easily
exceed this limit with moderately small lists.

This changeset allows for Asterisk to work around this size limit by
performing its own allocation of the transmission data buffer. This
way, Asterisk can allocate a buffer that exceeds the built-in maximum.

We still impose our own limit of 64000 bytes, mainly because making
allocations larger than that is a bit absurd.

ASTERISK-24181 #close
Reported by Mark Michelson

Review: https://reviewboard.asterisk.org/r/3977
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6 years agores_pjsip_pubsub: Check supported headers for eventlist when subscribing to
Jonathan Rose [Mon, 8 Sep 2014 15:58:24 +0000 (15:58 +0000)]
res_pjsip_pubsub: Check supported headers for eventlist when subscribing to
resource list

https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan
According to the off-nominal plan, if evenlist support is not specified in a
SUBSCRIBE's supported header(s), that subscription should be rejected with an
error.

ASTERISK-23871
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3960/diff/#index_header
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6 years agomain/cdr: Copy over location information during a fork
Matthew Jordan [Sat, 6 Sep 2014 22:50:24 +0000 (22:50 +0000)]
main/cdr: Copy over location information during a fork

When a CDR is forked, a new CDR is created and appended to the CDR chain for
the Party A. The forked CDR starts life off as a clone of the last
non-finalized for the particular Party A. In the past, merely copying over
the snapshots for Party A/Party B would be sufficient. However, as the CDRs
now contain cached information from Party A - specifically application/data,
context, and extension - we need to copy that over during a fork as well.

Huzzah for unit tests catching this when the context/extension were derived
from a cached value on the CDR instead of on Party A.
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6 years agomain/rtp_engine: Format NTP timestamps as unsigned ints
Matthew Jordan [Sat, 6 Sep 2014 22:22:34 +0000 (22:22 +0000)]
main/rtp_engine: Format NTP timestamps as unsigned ints

On some systems, a timeval's tv_sec/tv_usec will be unsigned lont ints, as
opposed to long ints. When the RTP engine formats these as strings, it was
previously formatting them as signed integers, which can result in some
odd negative timestamp values (particularly on 32-bit systems). This patch
formats the values as unsigned long integers.
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6 years agores_pjsip_sdp_rtp: Fix retrieval of "ice-pwd" attribute if in session and not media...
Joshua Colp [Sat, 6 Sep 2014 19:13:10 +0000 (19:13 +0000)]
res_pjsip_sdp_rtp: Fix retrieval of "ice-pwd" attribute if in session and not media stream.
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6 years agomain/cdrs: Preserve context/extension when executing a Macro or GoSub
Matthew Jordan [Fri, 5 Sep 2014 22:04:33 +0000 (22:04 +0000)]
main/cdrs: Preserve context/extension when executing a Macro or GoSub

The context/extension in a CDR is generally considered the destination of a
call. When looking at a 2-party call CDR, users will typically be presented
with the following:

context    exten      channel     dest_channel app  data
default    1000       SIP/8675309 SIP/1000     Dial SIP/1000,,20

However, if the Dial actually takes place in a Macro, the current behaviour
in 12 will result in the following CDR:

context    exten      channel     dest_channel app  data
macro-dial s          SIP/8675309 SIP/1000     Dial SIP/1000,,20

The same is true of a GoSub:

context    exten      channel     dest_channel app  data
subs       dial_stuff SIP/8675309 SIP/1000     Dial SIP/1000,,20

This generally makes the context/exten fields less than useful.

It isn't hard to preserve these values in the CDR state machine; however, we
need to have something that informs us when a channel is executing a
subroutine. Prior to this patch, there isn't anything that does this.

This patch solves this problem by adding a new channel flag,
AST_FLAG_SUBROUTINE_EXEC. This flag is set on a channel when it executes a
Macro or a GoSub. The CDR engine looks for this value when updating a Party A
snapshot; if the flag is present, we don't override the context/exten on the
main CDR object. In a funny quirk, executing a hangup handler must *not* abide
by this logic, as the endbeforehexten logic assumes that the user wants to see
data that occurs in hangup logic, which includes those subroutines. Since
those execute outside of a typical Dial operation (and will typically have
their own dedicated CDR anyway), this is unlikely to cause any heartburn.

Review: https://reviewboard.asterisk.org/r/3962/

ASTERISK-24254 #close
Reported by: tm1000, Tony Lewis
Tested by: Tony Lewis
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6 years agomain/cdr: Fix crash/memory consumption in CDRs in multi-party bridge scenarios
Matthew Jordan [Fri, 5 Sep 2014 21:56:52 +0000 (21:56 +0000)]
main/cdr: Fix crash/memory consumption in CDRs in multi-party bridge scenarios

This patch fixes an issue where CDRs would get stuck generating an infinite
number of CDRs, eventually crashing Asterisk (and consuming a lot of memory
along the way).

When a channel enters into a multi-party bridge, the CDR engine creates
mappings of each participant to each other participant, picking the 'A' party
as it goes. So, if we have four channels in a multi-party bridge (Alice, Bob,
Charlie, Denise), we would have something like:

Alice => Bob
Alice => Charlie
Alice => Denise
Bob => Charlie
Bob => Denise
Charlie => Denise

This works fine when participants enter the bridge a single time.

When a participant leaves a bridge, the CDRs for that channel are transitioned
to a finalized state.

The bug occurs if Bob rejoins. When the CDR engine creates mappings between the
channels, it walks through all the participants currently in the bridge, and
realizes that no one in the bridge can create a CDR with the channel (Bob).
As such it creates a new CDR for the candidate and appends it to that
candidate's chain. Unfortunately, on this particular code path, it doesn't
stop traversing the candidate's chain. Since we just added ourselves to the
chain, this causes the loop to keep going, constantly adding new CDRs.

This patch makes it so the engine bails when it creates a CDR match in this
case.

Review: https://reviewboard.asterisk.org/r/3964/

ASTERISK-24241 #close
Reported by: Deepak Singh Rawat
Tested by: Deepak Singh Rawat

ASTERISK-24208
Reported by: Frankie Chin
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6 years agofunc_channel.c: Add missing locking to some CHANNEL() requests.
Richard Mudgett [Fri, 5 Sep 2014 20:38:27 +0000 (20:38 +0000)]
func_channel.c: Add missing locking to some CHANNEL() requests.

* The CHANNEL() audionativeformat, videonativeformat, audioreadformat, and
audiowriteformat now need locking since the media format rework when
accessing the channel's format pointers.

* Increased the buffer size for CHANNEL() audionativeformat and
videonativeformat output strings since the allow=all can be a lengthy
list.

* Tweaked the CHANNEL() XML documentation for secure_bridge_signaling,
secure_bridge_media, and state.

* Ensured the output buffer is initialized for secure_bridge_signaling and
secure_bridge_media.

* Made use the locked_copy_string() macro instead of inlining it for trace
and checkhangup.
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6 years agoDial API: Add a dial option to indicate the dialed channel will replace dialer
Jonathan Rose [Fri, 5 Sep 2014 20:22:12 +0000 (20:22 +0000)]
Dial API: Add a dial option to indicate the dialed channel will replace dialer

Adds an option to the dial API that marks an outgoing dial as replacing the dialing channel for the purpose of propagating accountcode. When it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on the involved channels with ast_channel_req_accountcodes.

Review: https://reviewboard.asterisk.org/r/3968/
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6 years agoCall IDs: Fix appearance of call ID in core show channels when NULL
Jonathan Rose [Fri, 5 Sep 2014 19:39:04 +0000 (19:39 +0000)]
Call IDs: Fix appearance of call ID in core show channels when NULL

NULL call IDs were meant to appear as '(none)' but instead were showing
the contents of an uninitialized character buffer.

ASTERISK-24223
Review: https://reviewboard.asterisk.org/r/3979/
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