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The previous behavior was to simply set the accountcode of an outgoing
channel to the accountcode of the channel initiating the call. It was
done this way a long time ago to allow the accountcode set on the SIP/100
channel to be propagated to a local channel so the dialplan execution on
the Local;2 channel would have the SIP/100 accountcode available.
SIP/100 -> Local;1/Local;2 -> SIP/200
Propagating the SIP/100 accountcode to the local channels is very useful.
Without any dialplan manipulation, all channels in this call would have
the same accountcode.
Using dialplan, you can set a different accountcode on the SIP/200 channel
either by setting the accountcode on the Local;2 channel or by the Dial
application's b(pre-dial), M(macro) or U(gosub) options, or by the
FollowMe application's b(pre-dial) option, or by the Queue application's
macro or gosub options. Before Asterisk v12, the altered accountcode on
SIP/200 will remain until the local channels optimize out and the
accountcode would change to the SIP/100 accountcode.
Asterisk v1.8 attempted to add peeraccount support but ultimately had to
punt on the support. The peeraccount support was rendered useless because
of how the CDR code needed to unconditionally force the caller's
accountcode onto the peer channel's accountcode. The CEL events were thus
intentionally made to always use the channel's accountcode as the
peeraccount value.
With the arrival of Asterisk v12, the situation has improved somewhat so
peeraccount support can be made to work. Using the indicated example, the
the accountcode values become as follows when the peeraccount is set on
SIP/100 before calling SIP/200:
SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200
acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200
peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100
If a channel already has an accountcode it can only change by the
following explicit user actions:
1) A channel originate method that can specify an accountcode to use.
2) The calling channel propagating its non-empty peeraccount or its
non-empty accountcode if the peeraccount was empty to the outgoing
channel's accountcode before initiating the dial. e.g., Dial and
FollowMe. The exception to this propagation method is Queue. Queue will
only propagate peeraccounts this way only if the outgoing channel does not
have an accountcode.
3) Dialplan using CHANNEL(accountcode).
4) Dialplan using CHANNEL(peeraccount) on the other end of a local
channel pair.
If a channel does not have an accountcode it can get one from the
following places:
1) The channel driver's configuration at channel creation.
2) Explicit user action as already indicated.
3) Entering a basic or stasis-mixing bridge from a peer channel's
peeraccount value.
You can specify the accountcode for an outgoing channel by setting the
CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue
applications. Queue adds the wrinkle that it will not overwrite an
existing accountcode on the outgoing channel with the calling channels
values.
Accountcode and peeraccount values propagate to an outgoing channel before
dialing. Accountcodes also propagate when channels enter or leave a basic
or stasis-mixing bridge. The peeraccount value only makes sense for
mixing bridges with two channels; it is meaningless otherwise.
* Made peeraccount functional by changing accountcode propagation as
described above.
* Fixed CEL extracting the wrong ie value for the peeraccount. This was
done intentionally in Asterisk v1.8 when that version had to punt on
peeraccount.
* Fixed a few places dealing with accountcodes that were reading from
channels without the lock held.
AFS-65 #close
Review: https://reviewboard.asterisk.org/r/3601/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Reverting the patch since it was causing a regression and after fixing the
regression, there were no performance gains. At least based on my method
for measurement.
ASTERISK-24050
Review: https://reviewboard.asterisk.org/r/3841/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419504 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In Asterisk, it is possible for a device to have a status of ONHOLD. This is
not typically an easy thing to determine, as a channel being on hold is not
a direct channel state. Typically, this has to be calculated outside of the
core independently in channel drivers, notably, chan_sip and chan_pjsip. Both
of these channel drivers already have to calculate device state in a fashion
more complex than the core can handle, as they aggregate all state of all
channels associated with a peer/endpoint; they also independently track
whether or not one of those channels is currently on hold and mark the device
state appropriately.
In 12+, we now have the ability to report an AST_DEVICE_ONHOLD state for all
channels that defer their device state to the core. This is due to channel hold
state actually now being tracked on the channel itself. If a channel driver
defers its device state to the core (which many, such as DAHDI, IAX2, and
others do in most situations), the device state core already goes out to get a
channel associated with the device. As such, it can now also factor the channel
hold state in its calculation.
This patch adds this logic to the device state core. It also uses an existing
mapping between device state and channel state to handle more channel states.
chan_pjsip has been updated slightly as well to make use of this (as it was,
for some reason, reporting a channel state of BUSY as a device state of INUSE,
which feels slightly wrong).
Review: https://reviewboard.asterisk.org/r/3771/
ASTERISK-24038 #close
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419358 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Allow for responses to AMI actions/commands to be documented properly
in XML and displayed via the CLI. Response events are documented
exactly as standard AMI events are documented.
Review: https://reviewboard.asterisk.org/r/3812/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch does two things:
(1) It updates the unit tests to expect additional stasis messages. More
messages are now sent to the endpoint topic, due to forwarding all
channel messages and the forwarding relationship set up between
endpoints themselves.
(2) Remove the technology forwarding subscription during
ast_endpoint_shutdown. This prevents an improper double shutdown of
an endpoint from occurring.
........
Merged revisions 419318 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Unregister manager actions FAXSessions, FAXSession and FAXStats at unload.
* Update ast_manager_register2 use ao2_t_alloc tagged with the action name.
ASTERISK-24058 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3831/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Say you wanted to include variables in an application map and have those
variables substituted and passed along to the application being executed;
currently this does not happen.
This patch adds this ability to pass channel variable values to an
application before being executed.
ASTERISK-22608 #close
Reported by: Michael L. Young
patches:
features_substitute_arguments_v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/3819/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419252 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When updating a row, we are currently doing an INSERT OR REPLACE INTO. The
downside to this is that the row is deleted if it exists and then a new row is
inserted. So, we are hitting the disk twice. One for the deletion and one for
the insertion.
This patch changes this statement to an INSERT INTO and if the insert fails
because a row with that key exists, we will IGNORE the failure. Then we will
attempt to perform an UPDATE on the existing row if that row wasn't just
INSERTed.
ASTERISK-24050 #close
Reported by: Michael L. Young
patches:
astdb-insert-update-io-help_trunk_v2.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/3815/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch serves two purposes:
(1) It fixes some bugs with endpoint subscriptions not reporting all of the
channel events
(2) It serves as the preliminary work needed for ASTERISK-23692, which allows
for sending/receiving arbitrary out of call text messages through ARI in a
technology agnostic fashion.
The messaging functionality described on ASTERISK-23692 requires two things:
(1) The ability to send/receive messages associated with an endpoint. This is
relatively straight forwards with the endpoint core in Asterisk now.
(2) The ability to send/receive messages associated with a technology and an
arbitrary technology defined URI. This is less straight forward, as
endpoints are formed from a tech + resource pair. We don't have a
mechanism to note that a technology that *may* have endpoints exists.
This patch provides such a mechanism, and fixes a few bugs along the way.
The first major bug this patch fixes is the forwarding of channel messages
to their respective endpoints. Prior to this patch, there were two problems:
(1) Channel caching messages weren't forwarded. Thus, the endpoints missed
most of the interesting bits (such as channel creation, destruction, state
changes, etc.)
(2) Channels weren't associated with their endpoint until after creation.
This resulted in endpoints missing the channel creation message, which
limited the usefulness of the subscription in the first place (a major use
case being 'tell me when this endpoint has a channel'). Unfortunately,
this meant another parameter to ast_channel_alloc. Since not all channel
technologies support an ast_endpoint, this patch makes such a call
optional and opts for a new function, ast_channel_alloc_with_endpoint.
When endpoints are created, they will implicitly create a technology endpoint
for their technology (if one does not already exist). A technology endpoint is
special in that it has no state, cannot have channels created for it, cannot
be created explicitly, and cannot be destroyed except on shutdown. It does,
however, have all messages from other endpoints in its technology forwarded to
it.
Combined with the bug fixes, we now have Stasis messages being properly
forwarded. Consider the following scenario: two PJSIP endpoints (foo and bar),
where bar has a single channel associated with it and foo has two channels
associated with it. The messages would be forwarded as follows:
channel PJSIP/foo-1 --
\
--> endpoint PJSIP/foo --
/ \
channel PJSIP/foo-2 -- \
---- > endpoint PJSIP
/
channel PJSIP/bar-1 -----> endpoint PJSIP/bar --
ARI, through the applications resource, can:
- subscribe to endpoint:PJSIP/foo and get notifications for channels
PJSIP/foo-1,PJSIP/foo-2 and endpoint PJSIP/foo
- subscribe to endpoint:PJSIP/bar and get notifications for channels
PJSIP/bar-1 and endpoint PJSIP/bar
- subscribe to endpoint:PJSIP and get notifications for channels
PJSIP/foo-1,PJSIP/foo-2,PJSIP/bar-1 and endpoints PJSIP/foo,PJSIP/bar
Note that since endpoint PJSIP never changes, it never has events itself. It
merely provides an aggregation point for all other endpoints in its technology
(which in turn aggregate all channel messages associated with that endpoint).
This patch also adds endpoints to res_xmpp and chan_motif, because the actual
messaging work will need it (messaging without XMPP is just sad).
Review: https://reviewboard.asterisk.org/r/3760/
ASTERISK-23692
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Merged revisions 419196 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419203 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This prevents a crash in the Dial API triggered by use of the Page()
application where a format capability struct was used before checking
whether it was NULL.
ASTERISK-24074 #close
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In the old times media formats were represented using a bit field. This was
fast but had a few limitations.
1. Asterisk was limited in how many formats it could handle.
2. Formats, being a bit field, could not include any attribute information.
A format was strictly its type, e.g., "this is ulaw".
This was changed in Asterisk 10 (see
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for
notes on that work) which led to the creation of the ast_format structure.
This structure allowed Asterisk to handle attributes and bundle information
with a format.
Additionally, ast_format_cap was created to act as a container for multiple
formats that, together, formed the capability of some entity. Another
mechanism was added to allow logic to be registered which performed format
attribute negotiation. Everywhere throughout the codebase Asterisk was
changed to use this strategy.
Unfortunately, in software, there is no free lunch. These new capabilities
came at a cost.
Performance analysis and profiling showed that we spend an inordinate
amount of time comparing, copying, and generally manipulating formats and
their related structures. Basic prototyping has shown that a reasonably
large performance improvement could be made in this area. This patch is the
result of that project, which overhauled the media format architecture
and its usage in Asterisk to improve performance.
Generally, the new philosophy for handling formats is as follows:
* The ast_format structure is reference counted. This removed a large amount
of the memory allocations and copying that was done in prior versions.
* In order to prevent race conditions while keeping things performant, the
ast_format structure is immutable by convention and lock-free. Violate this
tenet at your peril!
* Because formats are reference counted, codecs are also reference counted.
The Asterisk core generally provides built-in codecs and caches the
ast_format structures created to represent them. Generally, to prevent
inordinate amounts of module reference bumping, codecs and formats can be
added at run-time but cannot be removed.
* All compatibility with the bit field representation of codecs/formats has
been moved to a compatibility API. The primary user of this representation
is chan_iax2, which must continue to maintain its bit-field usage of formats
for interoperability concerns.
* When a format is negotiated with attributes, or when a format cannot be
represented by one of the cached formats, a new format object is created or
cloned from an existing format. That format may have the same codec
underlying it, but is a different format than a version of the format with
different attributes or without attributes.
* While formats are reference counted objects, the reference count maintained
on the format should be manipulated with care. Formats are generally cached
and will persist for the lifetime of Asterisk and do not explicitly need
to have their lifetime modified. An exception to this is when the user of a
format does not know where the format came from *and* the user may outlive
the provider of the format. This occurs, for example, when a format is read
from a channel: the channel may have a format with attributes (hence,
non-cached) and the user of the format may last longer than the channel (if
the reference to the channel is released prior to the format's reference).
For more information on this work, see the API design notes:
https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
Finally, this work was the culmination of a large number of developer's
efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the
work in the Asterisk core, chan_sip, and was an invaluable resource in peer
reviews throughout this project.
There were a substantial number of patches contributed during this work; the
following issues/patch names simply reflect some of the work (and will cause
the release scripts to give attribution to the individuals who work on them).
Reviews:
https://reviewboard.asterisk.org/r/3814
https://reviewboard.asterisk.org/r/3808
https://reviewboard.asterisk.org/r/3805
https://reviewboard.asterisk.org/r/3803
https://reviewboard.asterisk.org/r/3801
https://reviewboard.asterisk.org/r/3798
https://reviewboard.asterisk.org/r/3800
https://reviewboard.asterisk.org/r/3794
https://reviewboard.asterisk.org/r/3793
https://reviewboard.asterisk.org/r/3792
https://reviewboard.asterisk.org/r/3791
https://reviewboard.asterisk.org/r/3790
https://reviewboard.asterisk.org/r/3789
https://reviewboard.asterisk.org/r/3788
https://reviewboard.asterisk.org/r/3787
https://reviewboard.asterisk.org/r/3786
https://reviewboard.asterisk.org/r/3784
https://reviewboard.asterisk.org/r/3783
https://reviewboard.asterisk.org/r/3778
https://reviewboard.asterisk.org/r/3774
https://reviewboard.asterisk.org/r/3775
https://reviewboard.asterisk.org/r/3772
https://reviewboard.asterisk.org/r/3761
https://reviewboard.asterisk.org/r/3754
https://reviewboard.asterisk.org/r/3753
https://reviewboard.asterisk.org/r/3751
https://reviewboard.asterisk.org/r/3750
https://reviewboard.asterisk.org/r/3748
https://reviewboard.asterisk.org/r/3747
https://reviewboard.asterisk.org/r/3746
https://reviewboard.asterisk.org/r/3742
https://reviewboard.asterisk.org/r/3740
https://reviewboard.asterisk.org/r/3739
https://reviewboard.asterisk.org/r/3738
https://reviewboard.asterisk.org/r/3737
https://reviewboard.asterisk.org/r/3736
https://reviewboard.asterisk.org/r/3734
https://reviewboard.asterisk.org/r/3722
https://reviewboard.asterisk.org/r/3713
https://reviewboard.asterisk.org/r/3703
https://reviewboard.asterisk.org/r/3689
https://reviewboard.asterisk.org/r/3687
https://reviewboard.asterisk.org/r/3674
https://reviewboard.asterisk.org/r/3671
https://reviewboard.asterisk.org/r/3667
https://reviewboard.asterisk.org/r/3665
https://reviewboard.asterisk.org/r/3625
https://reviewboard.asterisk.org/r/3602
https://reviewboard.asterisk.org/r/3519
https://reviewboard.asterisk.org/r/3518
https://reviewboard.asterisk.org/r/3516
https://reviewboard.asterisk.org/r/3515
https://reviewboard.asterisk.org/r/3512
https://reviewboard.asterisk.org/r/3506
https://reviewboard.asterisk.org/r/3413
https://reviewboard.asterisk.org/r/3410
https://reviewboard.asterisk.org/r/3387
https://reviewboard.asterisk.org/r/3388
https://reviewboard.asterisk.org/r/3389
https://reviewboard.asterisk.org/r/3390
https://reviewboard.asterisk.org/r/3321
https://reviewboard.asterisk.org/r/3320
https://reviewboard.asterisk.org/r/3319
https://reviewboard.asterisk.org/r/3318
https://reviewboard.asterisk.org/r/3266
https://reviewboard.asterisk.org/r/3265
https://reviewboard.asterisk.org/r/3234
https://reviewboard.asterisk.org/r/3178
ASTERISK-23114 #close
Reported by: mjordan
media_formats_translation_core.diff uploaded by kharwell (License 6464)
rb3506.diff uploaded by mjordan (License 6283)
media_format_app_file.diff uploaded by kharwell (License 6464)
misc-2.diff uploaded by file (License 5000)
chan_mild-3.diff uploaded by file (License 5000)
chan_obscure.diff uploaded by file (License 5000)
jingle.diff uploaded by file (License 5000)
funcs.diff uploaded by file (License 5000)
formats.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
bridges.diff uploaded by file (License 5000)
mf-codecs-2.diff uploaded by file (License 5000)
mf-app_fax.diff uploaded by file (License 5000)
mf-apps-3.diff uploaded by file (License 5000)
media-formats-3.diff uploaded by file (License 5000)
ASTERISK-23715
rb3713.patch uploaded by coreyfarrell (License 5909)
rb3689.patch uploaded by mjordan (License 6283)
ASTERISK-23957
rb3722.patch uploaded by mjordan (License 6283)
mf-attributes-3.diff uploaded by file (License 5000)
ASTERISK-23958
Tested by: jrose
rb3822.patch uploaded by coreyfarrell (License 5909)
rb3800.patch uploaded by jrose (License 6182)
chan_sip.diff uploaded by mjordan (License 6283)
rb3747.patch uploaded by jrose (License 6182)
ASTERISK-23959 #close
Tested by: sgriepentrog, mjordan, coreyfarrell
sip_cleanup.diff uploaded by opticron (License 6273)
chan_sip_caps.diff uploaded by mjordan (License 6283)
rb3751.patch uploaded by coreyfarrell (License 5909)
chan_sip-3.diff uploaded by file (License 5000)
ASTERISK-23960 #close
Tested by: opticron
direct_media.diff uploaded by opticron (License 6273)
pjsip-direct-media.diff uploaded by file (License 5000)
format_cap_remove.diff uploaded by opticron (License 6273)
media_format_fixes.diff uploaded by opticron (License 6273)
chan_pjsip-2.diff uploaded by file (License 5000)
ASTERISK-23966 #close
Tested by: rmudgett
rb3803.patch uploaded by rmudgetti (License 5621)
chan_dahdi.diff uploaded by file (License 5000)
ASTERISK-24064 #close
Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose
rb3814.patch uploaded by rmudgett (License 5621)
moh_cleanup.diff uploaded by opticron (License 6273)
bridge_leak.diff uploaded by opticron (License 6273)
translate.diff uploaded by file (License 5000)
rb3795.patch uploaded by rmudgett (License 5621)
tls_fix.diff uploaded by mjordan (License 6283)
fax-mf-fix-2.diff uploaded by file (License 5000)
rtp_transfer_stuff uploaded by mjordan (License 6283)
rb3787.patch uploaded by rmudgett (License 5621)
media-formats-explicit-translate-format-3.diff uploaded by file (License 5000)
format_cache_case_fix.diff uploaded by opticron (License 6273)
rb3774.patch uploaded by rmudgett (License 5621)
rb3775.patch uploaded by rmudgett (License 5621)
rtp_engine_fix.diff uploaded by opticron (License 6273)
rtp_crash_fix.diff uploaded by opticron (License 6273)
rb3753.patch uploaded by mjordan (License 6283)
rb3750.patch uploaded by mjordan (License 6283)
rb3748.patch uploaded by rmudgett (License 5621)
media_format_fixes.diff uploaded by opticron (License 6273)
rb3740.patch uploaded by mjordan (License 6283)
rb3739.patch uploaded by mjordan (License 6283)
rb3734.patch uploaded by mjordan (License 6283)
rb3689.patch uploaded by mjordan (License 6283)
rb3674.patch uploaded by coreyfarrell (License 5909)
rb3671.patch uploaded by coreyfarrell (License 5909)
rb3667.patch uploaded by coreyfarrell (License 5909)
rb3665.patch uploaded by mjordan (License 6283)
rb3625.patch uploaded by coreyfarrell (License 5909)
rb3602.patch uploaded by coreyfarrell (License 5909)
format_compatibility-2.diff uploaded by file (License 5000)
core.diff uploaded by file (License 5000)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This fixes a build failure introduced by r3821. struct stasis_topic is
opaque, so topic->name is unavailable. Switch to using stasis_topic_name().
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Merged revisions 419019 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Add tags to stasis objects using the name. This makes it
easier to track the source of certain stasis ref leaks.
Review: https://reviewboard.asterisk.org/r/3821/
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Merged revisions 418996 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418997 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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If a reference count goes negative, instead of
just logging that fact, be more helpful with a
backtrace and an assert that will DO_CRASH.
This patch also removes the duplicate ao2_bt()
function and cleans up extraneous usage of the
ast_log_backtrace() call.
Review: https://reviewboard.asterisk.org/r/3765/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418963 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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If the features.conf is missing, the cfg->featurgroups
and cfg->applicationmaps is not initialized, resulting
in assert on ao2_find of a null container. This patch
changes the initialization call and adds asserts for a
safeguard.
Review: https://reviewboard.asterisk.org/r/3809/
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Merged revisions 418886 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Whenever possible, audiohooks and framehooks will now be copied over
to the channel that the masquerading channel gets cloned into. This
should occur for all audiohooks and most framehooks. As a result,
in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now
deprecated and its behavior is essentially the new default for all
audiohooks, plus some additional audiohooks/framehooks.
Review: https://reviewboard.asterisk.org/r/3721/
........
Merged revisions 418914 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Ensure that three-way transfers can be reported even if featuremap is
non-NULL.
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Merged revisions 418810 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Technically, ADPCM is a method that can be applied to several codecs.
Asterisk's ADPCM codec is the Dialogic ADPCM or VOX codec.
See http://en.wikipedia.org/wiki/Dialogic_ADPCM for more information
about said codec.
Review: https://reviewboard.asterisk.org/r/3744
patches:
rb3744.patch uploaded by dennis.guse (License 6513)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When the PresenceState action is executed, the nominal path fails to include
the ActionID in the successful response. This patch adds a call to
astman_start_ack, which guarantees that an ActionID (if provided) will be
sent back to the AMI client.
Unlike the Asterisk 11 and 12 patches, this patch also deprecates the
duplicate Message key in the response to the action, replacing it with the
key 'PresenceMessage'.
Review: https://reviewboard.asterisk.org/r/3776/
ASTERISK-23985 #close
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This fixes two reference leaks that would occur when TEST_FRAMEWORK was
enabled and features were successfully executed.
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It's been 2014 for like... 6 months.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Update refcounter.py to use delta's to track the current reference count.
* Use result from internal_ao2_ref to write old_refcount to refs_log.
Review: https://reviewboard.asterisk.org/r/3756/
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* Cleanup games object in app_skel.
* Cleanup stasis subscription to TEST_FRAMEWORK in manager.c (12+).
Review: https://reviewboard.asterisk.org/r/3757/
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Tags are useful in hunting down ref imbalances; this patch adds tag variants
for these commonly used macros/functions.
Review: https://reviewboard.asterisk.org/r/3750/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When updated configuration is written back to the conf
file - for example when a user changes their voicemail
pin, make sure that any config hook that wants to know
of changes is informed.
Review: https://reviewboard.asterisk.org/r/3708/
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* Create a Stasis bridge sub-class to propagate linkedids and
accountcodes.
* Fixed the basic bridge sub-class to update peeraccount codes when the
number of channels in the bridge drops back down to two parties.
* Refactored ast_bridge_channel_update_accountcodes() to handle channels
joining/leaving the bridge.
* Fixed the basic bridge sub-class to not call the base bridge class pull
method twice.
AFS-105 #close
ASTERISK-23852 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/3720/
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This corrects two issues with the extra field information in Asterisk
12+ in channel event logs.
It is possible to inject custom values into the dialstatus provided by
ast_channel_dial_type() Stasis messages that fall outside the
enumeration allowed for the DIALSTATUS channel variable. CEL now
filters for the allowed values and ignores other values.
The "hangupsource" extra field key is always blank if the far end
channel is a chan_pjsip channel. This is because the hangupsource is
never set for the pjsip channel driver. This change sets the
hangupsource whenever a hangup is queued for chan_pjsip channels.
This corrects an issue with the pjsip channel driver where the
hangupcause information was not being set properly.
Review: https://reviewboard.asterisk.org/r/3690/
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Egads. Another bad deletion of too much when attempting to remove h323 stuff.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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The previous patch (r418034) fixed the 'glitch' that the channels/h323
Makefile no longer existed. Unfortunately, removing the entire line was a bit
of a blunder, as it meant that build_tools/menuselect-deps was never
generated. Hilarity ensued when actually trying to compile.
But hey! At least configure worked.
This patch fixes *that* glitch, and removes some more of the vestiges of h323.
(It had tendrils in the main Makefile? Crazy.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Billing records are fair,
To get paid is quite bright,
You should really use ODBC;
Good-bye cdr_sqlite.
Microsoft did once push H.323,
Hell, we all remember NetMeeting.
But try to compile chan_h323 now
And you will take quite a beating.
The XMPP and SIP war was fierce,
And in the distant fray
Was birthed res_jabber/chan_jingle;
But neither to stay.
For everyone did care and chase what Google professed.
"Free Internet Calling" was what devotees cried,
But Google did change the specs so often
That the developers were happy the day chan_gtalk died.
And then there was that odd application
Dedicated to the Polish tongue.
app_saycountpl was subsumed by Say;
One could say its bell was rung.
To read and parse a file from the dialplan
You could (I guess) use an application.
app_readfile did fill that purpose, but I think
A function is perhaps better in its creation.
Barging is rude, I'm not sure why we do it.
Inwardly, the caller will probably sigh.
But if you really must do it,
Don't use app_dahdibarge, use ChanSpy.
We all despise the sound of tinny robots
It makes our queues so cold.
To control such an abomination
It's better to not use Wait/SetMusicOnHold.
It's often nice to know properties of a channel
It makes our calls right
We have a nice function called CHANNEL
And so SIPCHANINFO is sent off into the night.
And now things get odd;
Apparently one could delimit with a colon
Properties from the SIPPEER function!
Commas are in; all others are done.
Finally, a word on pipes and commas.
We're sorry. We can't say it enough.
But those compatibility options in asterisk.conf;
To maintain them forever was just too tough.
This patch removes:
* cdr_sqlite
* chan_gtalk
* chan_jingle
* chan_h323
* res_jabber
* app_saycountpl
* app_readfile
* app_dahdibarge
It removes the following applications/functions:
* WaitMusicOnHold
* SetMusicOnHold
* SIPCHANINFO
It removes the colon delimiter from the SIPPEER function.
Finally, it also removes all compatibility options that were configurable from
asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems.
Review: https://reviewboard.asterisk.org/r/3698/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* Removed some incorrect newlines on ast_http_error() messages in
manager.c.
* Removed an incorrect newline in res_ari_channels.c.
Addendum to ASTERISK-23552
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Adds two new manager commands to pbx_config - DialplanExtensionAdd and
DialplanExtensionRemove which allow manager users to create and delete
extensions respectively.
Review: https://reviewboard.asterisk.org/r/3650/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417910 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Persistent HTTP connection support is needed due to the increased usage of
the Asterisk core HTTP transport and the frequency at which REST API calls
are going to be issued.
* Add http.conf session_keep_alive option to enable persistent
connections.
* Parse and discard optional chunked body extension information and
trailing request headers.
* Increased the maximum application/json and
application/x-www-form-urlencoded body size allowed to 4k. The previous
1k was kind of small.
* Removed a couple inlined versions of ast_http_manid_from_vars() by
calling the function. manager.c:generic_http_callback() and
res_http_post.c:http_post_callback()
* Add missing va_end() in ast_ari_response_error().
* Eliminated unnecessary RAII_VAR() use in http.c:auth_create().
ASTERISK-23552 #close
Reported by: Scott Griepentrog
Review: https://reviewboard.asterisk.org/r/3691/
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The patch for ASTERISK-23905 that added PFS support in Asterisk depends on the
elliptic curve library support being present in OpenSSL. As it turns out, some
versions of OpenSSL don't have this library - notably the version running on
our build agents.
This patch fixes the build by providing a configure check for the specific
library calls that the PFS patch relies on.
Review: https://reviewboard.asterisk.org/r/3709/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch enables Perfect Forward Secrecy (PFS) in Asterisk's core TLS API.
Modules that wish to enable PFS should consider the following:
- Ephemeral ECDH (ECDHE) is enabled by default. To disable it, do not
specify a ECDHE cipher suite in a module's configuration, for example:
tlscipher=AES128-SHA:DES-CBC3-SHA
- Ephemeral DH (DHE) is disabled by default. To enable it, add DH parameters
into the private key file, i.e., tlsprivatekey. For an example, see the
default dh2048.pem at
http://www.opensource.apple.com/source/OpenSSL098/OpenSSL098-35.1/src/apps/dh2048.pem?txt
- Because clients expect the server to prefer PFS, and because OpenSSL sorts
its cipher suites by bit strength, (see "openssl ciphers -v DEFAULT")
consider re-ordering your cipher suites in the conf file. For example:
tlscipher=AES128+kEECDH:AES128+kEDH:3DES+kEDH:AES128-SHA:DES-CBC3-SHA:-ADH:-AECDH
will use PFS when offered by the client. Clients which do not offer PFS
fall-back to AES-128 (or even 3DES as recommend by RFC 3261).
Review: https://reviewboard.asterisk.org/r/3647/
ASTERISK-23905 #close
Reported by: Alexander Traud
patches:
tlsPFS_for_HEAD.patch uploaded by Alexander Traud (License 6520)
tlsPFS.patch uploaded by Alexander Traud (License 6520)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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A loop in ast_careful_fwrite exists that will continually attempt to write to
a file stream, even in the presence of EAGAIN/EINTR errors. However, if a
connection that uses ast_careful_fwrite closes suddenly, ast_careful_fwrite's
call to fflush may return EAGAIN/EINTER along with EOF. A subsequent call to
fflush will return EOF but not clear errno, resulting in an infinite loop.
This patch clears errno after it is detected and handled the loop, such that
any subsequent call to fflush will not get erroneously stuck.
Review: https://reviewboard.asterisk.org/r/3704
#ASTERISK-23984 #close
Reported by: Steve Davies
patches:
fflush_loop_fix uploaded by one47 (License 5012)
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res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP.
This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide
a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation
completes. Configuration options to chan_sip and chan_pjsip have also been added to
allow behavior to be tweaked (such as forcing the AVP type media transports in SDP).
ASTERISK-22961 #close
Reported by: Jay Jideliov
Review: https://reviewboard.asterisk.org/r/3679/
Review: https://reviewboard.asterisk.org/r/3686/
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This patch adds support for the Japanese language to both the say family of
applications, as well as for VoiceMail and VoiceMailMain. A new pack of
language sounds will be released at the same time as the next major version
of Asterisk to support the new language features.
The language features can be enabled using a language code of 'ja'.
Review: https://reviewboard.asterisk.org/r/3477
ASTERISK-23324 #close
Reported by: Kevin McCoy
patches:
app_voicemail.c.20140226.jb.patch uploaded by Kevin McCoy (License 6586)
say.c.20140226.jb.patch uploaded by Kevin McCoy (License 6586)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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In v12+ the type values from the table are only used by the CEL unit
tests. Since the unit tests were only comparing a generated expected
event with a real event to see if the ie contents matched and using the
same table IE_PLTYPE values to read the event contents, the type
mismatches were not detected.
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This change ensures that __ao2_ref_debug writes to ref_log when given a
non-NULL pointer to an invalid ao2 object. This is to ensure that we
record any attempt manipulate references of already freed objects.
ASTERISK-23948 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3677/
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Add the "bridge_technology" extra field key to BRIDGE_ENTER and
BRIDGE_EXIT CEL events to convey the bridge technology in use at the
time the record was generated.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This patch allows the current owner of a channel to define various
feature hooks to be made available once the channel has entered a
bridge. This includes any hooks that are setup on the
ast_bridge_features struct such as DTMF hooks, bridge event hooks
(join, leave, etc.), and interval hooks.
Review: https://reviewboard.asterisk.org/r/3649/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When using FEC, with span=3 and entries=4 Asterisk will attempt to repair
the packet with sequence number 5, as it will see that packet -4 is
missing. The result is Asterisk sending garbage packets that can kill a
fax.
This patch adds a check to see if the sequence number is valid before
checking if the packet is missing.
Review: https://reviewboard.asterisk.org/r/3657/
#ASTERISK-23908 #close
Reported by: Torrey Searle
patches:
udptl_fec.patch uploaded by Torrey Searle (License 5334)
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Almost every reference operation against container node's uses
__ao2_alloc or __ao2_ref, thereby preventing ref logging for
the nodes. One node reference is released with ao2_t_ref, causing
refcounter.py to falsely report skews and leaks for many nodes.
ASTERISK-23922 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3670/
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Move eid functions from netsock.c to utils.c. These functions were
already published by utils.h. Flag netsock.h as deprecated and switch
res_pjsip_session.h to use netsock2.h. The only code that still uses
netsock.h is chan_iax2.
ASTERISK-23920 #close
Reported by: Corey Farrell
Review: https://reviewboard.asterisk.org/r/3661/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Appending the ;2 to the user supplied ;1 uniqueid to create the ;2 version
if the user did not also supply an extra uniqueid for the ;2 channel
resulted in allocating a buffer that was one byte too small.
* Fix off by one error in ast_unreal_new_channels() when generating the ;2
uniqueid from the user suppled ;1 version.
* Pulled some long assignment lines from if tests to improve line break
readability in ast_unreal_new_channels().
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Part of a series of AMI command equivalents to existing CLI
commands
Review: https://reviewboard.asterisk.org/r/3651/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416848 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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callback.
* Extract the sayname API call to its own registerd callback. This allows
the app_directory and app_chanspy applications to say a mailbox owner's
name using an alternate provider when app_voicemail is not available
because you are using res_mwi_external. app_directory still uses the
voicemail.conf file.
AFS-64 #close
Reported by: Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Move some implementation specific code from astobj2_container.c into
astobj2_hash.c and astobj2_rbtree.c. This completely removes the need for
astobj2_container to switch on RTTI and it no longer has any knowledge of
the implementation details.
Also adds AO2_DEBUG as a new compile option in menuselect which controls
astobj2 debugging independently of AST_DEVMODE and REF_DEBUG.
Tested by: George Joseph
Review: https://reviewboard.asterisk.org/r/3593/
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