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The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f
patch introduced several regressions when the newly created "Updated"
state goes out for each endpoint registration refresh.
1) It restarted any OPTIONS RTT ping cycle.
2) It would interfere with a currently active ping and throw off that
ping's resulting RTT calculation.
3) It cleared the RTT time each time the endpoint was refreshed.
4) The cleared RTT time was sent out as a statsd update each time.
5) It created two AMI events for each update.
* Revert the original patch and reimplement it. Now the current contact
status state is re-sent instead of the state being momentarily toggled
every time the endpoint refreshes its registration. The statsd events are
not created for the re-sent refresh because they are sent after every
OPTIONS ping.
ASTERISK-26160 #close
Reported by: Matt Jordan
Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1
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If specified, incoming SUBSCRIBE requests will be searched for the matching
extension in the indicated context. If no "subscribe_context" is specified,
then the "context" setting is used.
ASTERISK-25471 #close
Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514
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ASTERISK-18995 #close
Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a
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We should not be processing any incoming messages until we are fully
booted. We may not have dialplan or other needed configuration loaded
yet.
ASTERISK-26089 #close
Reported by: Scott Griepentrog
ASTERISK-26088
Reported by: Richard Mudgett
Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264
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Testing has shown that our usage of UnixODBC is problematic
due to bugs within UnixODBC itself as well as the heavy weight
cost of connecting and disconnecting database connections, even
when pooling is enabled.
For users of UnixODBC 2.3.1 and earlier crashes would occur due
to insufficient protection of the disconnect operation. This was
fixed in UnixODBC 2.3.2 and above.
For users of UnixODBC 2.3.3 and higher a slow-down would occur
under heavy database use due to repeated connection establishment.
A regression is present where on each connection the database
configuration is cached again, with the cache growing out of
control.
The connection pool implementation present in this change helps
to mitigate these issues by reducing how much we connect and
disconnect database connections. We also solve the issue of
crashes under UnixODBC 2.3.1 by defaulting the maximum number of
connections to 1, returning us to the previous working behavior.
For users who may have a fixed version the maximum concurrent
connection limit can be increased helping with performance.
The connection pool works by keeping a list of active connections.
If the connection limit has not been reached a new connection is
established. If the connection limit has been reached then the
request waits until a connection becomes available before
continuing.
ASTERISK-26074 #close
ASTERISK-26054 #close
Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff
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Icelandic has some weird grammar rules when dealing with dates and
numbers. There are different genders used depending on which number
you're dealing with, and only a handful of numbers do change depending
on the gender. There is also an implied gender in several cases.
This patch was originally written for asterisk 1.6, and has been in use
for several years without crashes. I cleaned it up a bit and rewrote
what was necessary for Asterisk 13.
The functions were copied from other similar languages and modified
where appropriate. If i recall correctly, the German and Danish
functions were used as a base.
ASTERISK-26087
Reported by: Örn Arnarson
Tested by: Örn Arnarson
Change-Id: Ib7d8bd7b0fede5767921ed821315b5b508c0e665
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Change the awkward and not as flexible UnicastRTP options format
From:
Dial(UnicastRTP/127.0.0.1[/[<engine>][/[<codec>]]])
To:
Dial(UnicastRTP/127.0.0.1[/[<options>]])
Where <options> can be standard Asterisk flag options:
c(<codec>) - Specify which codec/format to use such as 'ulaw'.
e(<engine>) - Specify which RTP engine to use such as 'asterisk'.
More option flags can be easily added later such as the codec's RTP
payload type to use when the codec does not have a static payload type
defined.
Change-Id: I0c297aaf09e2ee515536cb7437bb8042ff8ff3c9
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Added a new channel variable FORWARDERNAME which indicates which
channel was responsible for a forwarding requests received on dial attempt.
Fixed a bug in the app_queue: FORWARD_CONTEXT is not used.
ASTERISK-26059 #close
Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2
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If you create a local channel and don't specify an originator channel
to take capabilities from, we automatically add all audio formats to
the new channel's capabilities. When we try to make the channel
compatible with another, the "best format" functions pick the best
format available, which in this case will be slin192. While this is
great for preserving quality, it's the worst for performance and
overkill for the vast majority of applications.
In the absense of any other information, adding all formats is the
correct thing to do and it's not always possible to supply an
originator so a new parameter 'formats' has been added to the channel
create/originate functions. It's just a comma separated list of formats
to make availalble for the channel. Example: "ulaw,slin,slin16".
'formats' and 'originator' are mutually exclusive.
To facilitate determination of format names, the format name has been
added to "core show codecs".
ASTERISK-26070 #close
Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b
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Dial events up to this point have come in two flavors
* A Dial event with no status to indicate that dialing has begun
* A Dial event with a status to indicate that dialing has ended
With this change, Dial events have been expanded to also give
intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS".
This is especially useful for ARI dialing, as it gives the application
writer the opportunity to place a channel into an early bridge when
early media is detected.
AMI handles these in-progress dial events by sending a new event called
"DialState" that simply indicates that dial state has changed but has
not ended. ARI never distinguished between DialBegin and DialEnd, so no
change was made to the event itself.
Another change here relates to dial forwards. A forward-related event
was previously only sent when a channel was successfully able to forward
a call to a new channel. With this set of changes, if forwarding is
blocked, we send a Dial event with a forwarding destination but no
forwarding channel, since we were prevented from creating one. This is
again useful for ARI since application writers can now handle call
forward attempts from within their own application.
ASTERISK-25925 #close
Reported by Mark Michelson
Change-Id: I42cbec7730d84640a434d143a0d172a740995543
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As res_pjsip_nat rewrites contact's address, only the last Via header
can contain the source address of registered endpoint.
Also Call-Id header may contain the source address of registered
endpoint.
Added "via_addr", "via_port", "call_id" to contact.
Added new fields ViaAddress, CallID to AMI event ContactStatus.
ASTERISK-26011
Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576
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Add the option 'enable_callee_prompt' to followme.conf. Enabled by
default. If disabled, a callee is not prompted to accept or reject
the forwarded call.
ASTERISK-26064 #close
Change-Id: I0a8b19d4cf95c86a07c992813babb9e4a4acfff5
Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>
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recording"
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func_odbc was changed in Asterisk 13.9.0
to make func_odbc use a single database connection per DSN
because of reported bug ASTERISK-25938
with MySQL/MariaDB LAST_INSERT_ID().
This is drawback in performance when func_odbc is used
very often in dialplan.
Single database connection should be optional.
ASTERISK-26010
Change-Id: I7091783a7150252de8eeb455115bd00514dfe843
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This patch adds a new feature to ARI that allows a client to download
the media associated with a stored recording. The new route is
/recordings/stored/{name}/file, and transmits the underlying binary file
using Asterisk's HTTP server's underlying file transfer facilities.
Because this REST route returns non-JSON, a few small enhancements had
to be made to the Python Swagger generation code, as well as the
mustache templates that generate the ARI bindings.
ASTERISK-26042 #close
Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181
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Some SBCs require responses to empty SIP INFO packets
after establishing call via INVITE, if not responded to
they may drop your call after unspecified timeout of X minutes.
They are identified by having no Content-Type, check for this
and respond with 200 - OK message.
ASTERISK-24986 #close
Reported-by: Ilya Trikoz, Federico Santulli
Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0
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Added a new multi_user option that when specified allows a particular
configuration to be used for multiple users. It does this by replacing
the user portion of the server uri with a dynamically created one.
Two new API calls have been added in order to make use of the new
functionality:
ast_sip_publish_user_send - Sends an outgoing publish message based on the
given user. If state for the user already exists it uses that, otherwise
it dynamically creates new outbound publishing state for the user at that
time.
ast_sip_publish_user_remove - Removes all outbound publish state objects
associated with the user. This essentially stops outbound publishing for
the user.
ASTERISK-25965 #close
Change-Id: Ib88dde024cc83c916424645d4f5bb84a0fa936cc
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* Provide consistent indenting of lines in bulleted paragraphs
* Respect the 80 character column width
* Group all like items together, e.g., all dialplan applications under
"Applications", etc.
* Use a single blank line to break up functionality changes within a
larger section
* Use two blanks lines to delineate larger sections
Change-Id: I0488554f5cb7c51da70003d69288a21c9aab9647
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Many ARI applications will want to play multiple media files in a row to
a resource. The most common use case is when building long-ish IVR prompts
made up of multiple, smaller sound files. Today, that requires building a
small state machine, listening for each PlaybackFinished event, and triggering
the next sound file to play. While not especially challenging, it is tedious
work. Since requiring developers to write tedious code to do normal activities
stinks, this patch adds the ability to play back a list of media files to a
resource.
Each of the 'play' operations on supported resources (channels and bridges)
now accepts a comma delineated list of media URIs to play. A single Playback
resource is created as a handle to the entire list. The operation of playing
a list is identical to playing a single media URI, save that a new event,
PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final
media URI. When the entire list is finished being played, a PlaybackFinished
event is raised.
In order to help inform applications where they are in the list playback, the
Playback resource now includes a new, optional attribute, 'next_media_uri',
that contains the next URI in the list to be played.
It's important to note the following:
- If an offset is provided to the 'play' operations, it only applies to the
first media URI, as it would be weird to skip n seconds forward in every
media resource.
- Operations that control the position of the media only affect the current
media being played. For example, once a media resource in the list
completes, a 'reverse' operation on a subsequent media resource will not
start a previously completed media resource at the appropiate offset.
- This patch does not add any new operations to control the list. Hopefully,
user feedback and/or future patches would add that if people want it.
ASTERISK-26022 #close
Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f
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At one point in time, it seemed like a good idea to use the Asterisk
channel name as the HEP correlation UUID. In particular, it felt like
this would be a useful identifier to tie PJSIP messages and RTCP
messages together, along with whatever other data we may eventually send
to Homer. This also had the benefit of keeping the correlation UUID
channel technology agnostic.
In practice, it isn't as useful as hoped, for two reasons:
1) The first INVITE request received doesn't have a channel. As a
result, there is always an 'odd message out', leading it to be
potentially uncorrelated in Homer.
2) Other systems sending capture packets (Kamailio) use the SIP Call-ID.
This causes RTCP information to be uncorrelated to the SIP message
traffic seen by those capture nodes.
In order to support both (in case someone is trying to use res_hep_rtcp
with a non-PJSIP channel), this patch adds a new option, uuid_type, with
two valid values - 'call-id' and 'channel'. The uuid_type option is used
by a module to determine the preferred UUID type. When available, that
source of a correlation UUID is used; when not, the more readily available
source is used.
For res_hep_pjsip:
- uuid_type = call-id: the module uses the SIP Call-ID header value
- uuid_type = channel: the module uses the channel name if available,
falling back to SIP Call-ID if not
For res_hep_rtcp:
- uuid_type = call-id: the module uses the SIP Call-ID header if the
channel type is PJSIP and we have a channel,
falling back to the Stasis event provided
channel name if not
- uuid_type = channel: the module uses the channel name
ASTERISK-25352 #close
Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c
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With the old SIP module we can use IP access controls per peer.
PJSIP module missing this feature.
This patch added next configuration Endpoint options:
"acl" - list of IP ACL section names in acl.conf
"deny" - List of IP addresses to deny access from
"permit" - List of IP addresses to permit access from
"contact_acl" - List of Contact ACL section names in acl.conf
"contact_deny" - List of Contact header addresses to deny
"contact_permit" - List of Contact header addresses to permit
This patch also better logging failed request:
add custom message instead of "No matching endpoint found"
add SIP method to logging
ASTERISK-25900
Change-Id: I456dea3909d929d413864fb347d28578415ebf02
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This patch allows for having app_confbridge register the name of the
conference as an extension into a specific context, similar to
regcontext for chan_sip. This variant is not quite as involved as the
one in chan_sip and doesn't allow for multiple contexts or custom
extensions, you can only specify the context and the conference name
will always be used as the extension to register.
ASTERISK-25989 #close
Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f
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The app_fax set FAXMODE variable, but res_fax missing this feature.
This patch add FAXMODE variable which is set to either "audio" or "T38".
ASTERISK-25980
Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b
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With the old SIP module AMI sends PeerStatus event on every
successfully REGISTER requests, ie, on start registration,
update registration and stop registration.
With PJSIP AMI sends ContactStatus only when status is changed.
Regarding registration:
on start registration - Created
on stop registration - Removed
but on update registration nothing
This patch added contact.updated event.
ASTERISK-25904
Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f
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If the Asterisk system name is set in asterisk.conf, it will be stored
into the "reg_server" field in the ps_contacts table to facilitate
multi-server setups.
ASTERISK-25931
Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8
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A feature of chan_sip that service providers relied upon was the ability to
identify by the Authorization username. This is most often used when customers
have a PBX that needs to register rather than identify by IP address. From my
own experiance, this is pretty common with small businesses who otherwise
don't need a static IP.
In this scenario, a register from the customer's PBX may succeed because From
will usually contain the PBXs account id but an INVITE will contain the caller
id. With nothing recognizable in From, the service provider's Asterisk can
never match to an endpoint and the INVITE just stays unauthorized.
The fixes:
A new value "auth_username" has been added to endpoint/identify_by that
will use the username and digest fields in the Authorization header
instead of username and domain in the the From header to match an endpoint,
or the To header to match an aor. This code as added to
res_pjsip_endpoint_identifier_user rather than creating a new module.
Although identify_by was always a comma-separated list, there was only
1 choice so order wasn't preserved. So to keep the order, a vector was added
to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar
to find the aor. The res_pjsip_endpoint_identifier_* modules are called in
globals/endpoint_identifier_order.
Along the way, the logic in res_pjsip_registrar was corrected to match
most-specific to least-specific as res_pjsip_endpoint_identifier_user does.
The order is:
username@domain
username@domain_alias
username
Auth by username does present 1 problem however, the first INVITE won't have
an Authorization header so the distributor, not finding a match on anything,
sends a securty_alert. It still sends a 401 with a challenge so the next
INVITE will have the Authorization header and presumably succeed. As a result
though, that first security alert is actually a false alarm.
To address this, a new feature has been added to pjsip_distributor that keeps
track of unidentified requests and only sends the security alert if a
configurable number of unidentified requests come from the same IP in a
configurable amout of time. Those configuration options have been added to
the global config object. This feature is only used when auth_username
is enabled.
Finally, default_realm was added to the globals object to replace the hard
coded "asterisk" used when an endpoint is not yet identified.
The testsuite tests all pass but new tests are forthcoming for this new
feature.
ASTERISK-25835 #close
Reported-by: Ross Beer
Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d
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This patch added new global pjsip option 'disable_multi_domain'.
Disabling Multi Domain can improve Realtime performance by reducing
number of database requests.
ASTERISK-25930 #close
Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7
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ChanSpy was creating its audiohook with the flags AST_AUDIOHOOK_TRIGGER_SYNC
and AST_AUDIOHOOK_SMALL_QUEUE, which caused audio frames to be lost when
queues grow too large or when read and write queues go out of sync.
Now these flags are set conditionally:
- AST_AUDIOHOOK_TRIGGER_SYNC is not set if the option "o" is set
- a new option "l" is created: if set, AST_AUDIOHOOK_SMALL_QUEUE will not
be set on the audiohook
ASTERISK-25866
Change-Id: I9c7652f41d9fa72c8691e4e70ec4fd16b047a4dd
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The first available transport of the appropriate type is used now.
This patch adds new config option 'transport' for outbound-publish.
If transport is set then outbound PUBLISH requests will use this transport.
ASTERISK-25901 #close
Change-Id: Ib389130489b70e36795b0003fa5fd386e2680151
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This adds a new ARI method that allows for you to dial a channel that
you previously created in ARI.
By combining this with the create method for channels, it allows for a
workflow where a channel can be created, manipulated, and then dialed.
The channel is under control of the ARI application during all stages of
the Dial and can even be manipulated based on channel state changes
observed within an ARI application.
The overarching goal for this is to eventually be able to add a dialed
channel to a Stasis bridge earlier than the "Up" state. However, at the
moment more work is needed in the Dial and Bridge APIs in order to
facilitate that.
ASTERISK-25889 #close
Change-Id: Ic6c399c791e66c4aa52454222fe4f8b02483a205
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This adds a new ARI method to the channels resource that allows for the
creation of a new channel. The channel is created and then placed into
the specified Stasis application.
This is different from the existing originate method that creates a
channel, dials it, and then places the answered channel into the
dialplan or a Stasis application. This method does not attempt to call
the channel at all. Dialing is left as a later step after channel
creation. This allows for pre-dialing channel manipulation if desired.
ASTERISK-25889
Change-Id: I3c96a0aba914b08e39f6256371a5bd4c92cbded8
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This change introduces the concept of autohints. These are hints
which are created as a result of device state changes occurring within
the core. When this happens a hint will be created (if it does not
exist already) using the device name as the extension.
For example if a device state change is received for "PJSIP/bob"
and autohints are enabled on a context then a hint will exist in
that context for "bob" with a device of "PJSIP/bob".
For virtual or custom device states the name after the type will
be used. For example if the device state of "Custom:bob" changes
then a hint will exist in that context for "bob" with a device of
"Custom:bob".
This functionality can be enabled in extensions.conf by placing
"autohints=yes" in a context.
ASTERISK-25881 #close
Change-Id: I7e444c7da41b7b7d33374420fec658beeb18584e
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Added the ability to show channel statistics to chan_pjsip (cli_functions.c)
Moved the existing 'pjsip show channel(s)' functionality from
pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's
private header so it made sense to move the existing channel commands as well.
Now using stasis_cache_dump to get the channel snapshots rather than retrieving
all endpoints, then getting each one's channel snapshots. Much more efficient.
Change-Id: I03b114522126d27434030b285bf6d531ddd79869
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Blind transfers to a recognized parking extension need to use the parker's
channel variable values to create the dynamic parking lot. This is
because there is always only one parker while the parkee may actually be a
multi-party bridge. A multi-party bridge can never supply the needed
channel variables to create the dynamic parking lot. In the multi-party
bridge blind transfer scenario, the parker's CHANNEL(parkinglot) value and
channel variables are inherited by the local channel used to park the
bridge.
* In park_common_setup(), make use the parker instead of the parkee to
supply the dynamic parking lot channel variable values. In all but one
case, the parkee is the same as the parker. However, in the recognized
parking extension blind transfer scenario for a two party bridge they are
different channels. For consistency, we need to use the parker channel.
* In park_local_transfer(), pass the CHANNEL(parkinglot) value to the
local channel when blind transferring a multi-party bridge to a recognized
parking extension.
* When a local channel starts a call, the Local;2 side needs to inherit
the CHANNEL(parkinglot) value from Local;1.
The DTMF one-touch parking case wasn't even trying to create dynamic
parking lots before it aborted the attempt.
* In parking_park_call(), add missing code to create a dynamic parking
lot.
A DTMF bridge hook is documented as returning -1 to remove the hook.
Though the hook caller is really coded to accept non-zero. See the
ast_bridge_hook_callback typedef.
* In feature_park_call(), don't remove the DTMF one-touch parking hook
because of an error.
ASTERISK-24605 #close
Reported by: Philip Correia
Patches:
call_park.patch (license #6672) patch uploaded by Philip Correia
Change-Id: I221d3a8fcc181877a1158d17004474d35d8016c9
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This patch allows applications/APIs that access media through the core file
APIs to play media in the media cache. Prior to determining if a 'filename'
exists, the filename is passed to the media cache's retrieve API call. If
that call succeeds, the local file specified passed back by the API is
opened for streaming. When used in this fashion, the 'filename' is actually
a URI that the media cache process and understand.
ASTERISK-25654 #close
Change-Id: I73b6e2e90c3e91b8500581c45cdf9c0dc785f5f0
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