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This patch adds a new option to the CHANNEL function that allows for the
extraction of the SIP call-id. It is used in conjunction with the 'pjsip'
option, and will return the Call-ID of the INVITE request that established
the PJSIP channel.
ASTERISK-25352
Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a
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An http request can be sent to rotate a specified log channel.
If the channel does not exist, an error response will be
returned.
The command "curl -v -u user:pass -X PUT 'http://localhost:8088
/ari/asterisk/logging/logChannelName/rotate'" can be run in the
terminal to access this new functionality.
* Added the ability to rotate log files through ARI
ASTERISK-25252
Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01
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This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold'
endpoint options. These allow the channel to be hung up if RTP
is not received from the remote endpoint for a specified number of
seconds.
ASTERISK-25259 #close
Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9
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This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the
chan_sip option, this specifies an interval, in seconds, at which we
will send RTP comfort noise frames. This can be useful for keeping RTP
sessions alive as well as keeping NAT associations alive during lulls.
ASTERISK-25242 #close
Reported by Mark Michelson
Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d
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This patch adds support for push configuration of dynamic, i.e.,
sorcery, objects in Asterisk. It adds three new REST API calls to the
'asterisk' resource:
* GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current
object given its ID. This returns back a list of ConfigTuples, which
define the fields and their present values that make up the object.
* PUT /asterisk/{configClass}/{objectType}/{id}: create or update an
object. A body may be passed with the request that contains fields to
populate in the object. The same format as what is retrieved using
the GET operation is used for the body, save that we specify that the
list of fields to update are contained in the "fields" attribute.
* DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic
object from its backing storage.
Note that the success/failure of these operations is somewhat
configuration dependent, i.e., you must be using a sorcery wizard that
supports the operation in question. If a sorcery wizard does not support
the create or delete mechanisms, then the REST API call will fail with a
403 forbidden.
ASTERISK-25238 #close
Change-Id: I28cd5c7bf6f67f8e9e437ff097f8fd171d30ff5c
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An http request can be sent to reload an Asterisk module. If the
module can not be reloaded or is not already loaded, an error
response will be returned.
The command "curl -v -u user:pass -X PUT 'http://localhost:8088
/ari/asterisk/modules/{moduleName}'" (or something similar, based
on configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Asterisk modules can be reloaded through http requests
ASTERISK-25173
Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1
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An http request can be sent to unload an Asterisk module. If the
module can not be unloaded or is already unloaded, an error response
will be returned.
The command "curl -v -u user:pass -X DELETE 'http://localhost:8088
/ari/asterisk/modules/{moduleName}'" (or something similar, depending
on configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Asterisk modules can be unloaded through http requests
ASTERISK-25173
Change-Id: I535a95f5676deb02651522761ecbdc0b00b5ac57
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An http request can be sent to load an Asterisk module. If the
module can not be loaded or is loaded already, an error response
will be returned.
The command curl -v -u user:pass -X POST 'http://localhost:8088/ari
/asterisk/modules/{moduleName}'" (or something similar, depending on
configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Asterisk modules can be loaded through http requests
ASTERISK-25173
Change-Id: I9e05d5b8c5c666ecfef341504f9edc1aa84fda33
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An http request can be sent to retrieve information on a single
module, including the resource name, description, use count, status,
and support level.
The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari
/asterisk/modules/{moduleName}'" (or something similar, depending on
configuration) can be run in the terminal to access this new
functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Information on a single module can now be retrieved
ASTERISK-25173
Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463
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An http request can be sent to retrieve a list of all existing modules,
including the resource name, description, use count, status, and
support level.
The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/
asterisk/modules" (or something similar, depending on configuration)
can be run in the terminal to access this new functionality.
For more information, see:
https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource
* Added new ARI functionality
* Information on modules can now be retrieved
Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0
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ASTERISK-25189 #close
Reported by: John Hardin
Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac
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Some phones send g.726 audio packed for AAL2, which differs from what is
recommended by RFC 3351. If Asterisk receives audio formatted as such when
negotiating g.726 then it sounds a bit distorted. Added an option to
res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726
AAL2 packed.
ASTERISK-25158 #close
Reported by: Steve Pitts
Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615
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Add a new ContactStatus AMI event.
Publish the following status/state changes:
Created
Removed
Reachable
Unreachable
Unknown
Contact URI, new status/state, aor and endpoint names, and the
last qualify rtt result are included in the event.
ASTERISK-25114 #close
Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e
Reported-by: George Joseph <george.joseph@fairview5.com>
Tested-by: George Joseph <george.joseph@fairview5.com>
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verification.
This way one X.509 certificate can be used for hosts that
can be reached under multiple DNS names or for multiple hosts.
Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name>
ASTERISK-25063 #close
Change-Id: I13302c80490a0b44c43f1b45376c9bd7b15a538f
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Previous versions of Asterisk processed command-line options before
processing asterisk.conf. This meant that if an option was set in
asterisk.conf, it could not be overridden with the equivelent command
line option. This change causes Asterisk to process the command-line
twice. First it processes options that are needed to load asterisk.conf,
then it processes the remaining options after the config is read.
This changes the function of -X slightly. Previously using -X without
disabling execincludes in asterisk.conf caused #exec to be usable in any
config. Now -X only enables #exec for the load of asterisk.conf, if it
is wanted in the rest of the system it must be enabled with execincludes
in asterisk.conf. Updated 'asterisk -h' and 'man asterisk' to reflect
the limited function of -X.
ASTERISK-25042 #close
Reported by: Corey Farrell
Change-Id: I1450d45c15b4467274b871914d893ed4f6564cd7
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Add feature to set optional schema parameter on configuration file via
'schema' setting.
Fix query to get columns from table while considering schema. If in
the database there exists two tables with same name in distinct schemas
it will return an error when inserting record.
ASTERISK-24967 #close
Change-Id: I691fd2cbc277fcba10e615f5884f8de5d8152f2c
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Added the ability to set the character to quote identifiers. This
allows adding the character at the start and end of table and column
names. This setting is configurable for cdr_adaptive_odbc via the
quoted_identifiers in configuration file cdr_adaptive_odbc.conf.
ASTERISK-25006
Change-Id: I0b9a56b79ca13a727a803d88ed3b8643e37632b8
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This patch adds a new option to cdr.conf, 'newcdrcolumns', that will handle CDR
columns added in Asterisk 1.8. The columns are:
* peeraccount
* linkedid
* sequence
When enabled, the columns in the database entry will be populated with the data
from the CDR.
ASTERISK-24976 #close
Change-Id: I51a57063f4ae5e194a9d933a8df45dc8a4534f0b
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Some telco switches occasionally ignore ISDN RESTART requests. The fix
for ASTERISK-19608 added an escape clause for B channels in the restarting
state if the telco ignores a RESTART request. If the telco fails to
acknowledge the RESTART then Asterisk will assume the telco acknowledged
the RESTART on the second call attempt requesting the B channel by the
telco. The escape clause is good for dealing with RESTART requests in
general but it does cause the next call for the restarting B channel to be
rejected if the telco insists the call must go on that B channel.
chan_dahdi doesn't really need to issue a RESTART request in response to
receiving a cause 44 (Requested channel not available) code. Sending the
RESTART in such a situation is not required (nor prohibited) by the
standards. I think chan_dahdi does this for historical reasons to deal
with buggy peers to get channels unstuck in a similar fashion as the
chan_dahdi.conf resetinterval option.
* Add the chan_dahdi.conf force_restart_unavailable_chans compatability
option that when disabled will prevent chan_dahdi from trying to RESTART
the channel in response to a cause 44 code.
ASTERISK-25034 #close
Reported by: Richard Mudgett
Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65
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Version"
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Change-Id: I534ea0f22759e3633585dfa9b145b4a284efe67f
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Add new column to INSERT new columns added in cdr 1.8 version. The columns are:
* peeraccount
* linkedid
* sequence
This feature is configurable in cdr_odbc.conf using a new configuration
option, 'newcdrcolumns'.
ASTERISK-24976 #close
Change-Id: Ibe0c7540a88305c6012786f438a0813ad8b19127
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Change-Id: I6b43e43474bf6fb77b8227eadb036036f8e90521
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* changes:
res_pjsip: Add global option to limit the maximum time for initial qualifies
pjsip_options: Add qualify_timeout processing and eventing
res_pjsip: Refactor endpt_send_request to include transaction timeout
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Currently when Asterisk starts initial qualifies of contacts are spread out
randomly between 0 and qualify_timeout to prevent network and system overload.
If a contact's qualify_frequency is 5 minutes however, that contact may be
unavailable to accept calls for the entire 5 minutes after startup. So while
staggering the initial qualifies is a good idea, basing the time on
qualify_timeout could leave contacts unavailable for too long.
This patch adds a new global parameter "max_initial_qualify_time" that sets the
maximum time for the initial qualifies. This way you could make sure that all
your contacts are initialy, randomly qualified within say 30 seconds but still
have the contact's ongoing qualifies at a 5 minute interval.
If max_initial_qualify_time is > 0, the formula is initial_interval =
min(max_initial_interval, qualify_timeout * random(). If not set,
qualify_timeout is used.
The default is "0" (disabled).
ASTERISK-24863 #close
Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4
Tested-by: George Joseph <george.joseph@fairview5.com>
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This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the
discussion at
http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html
The basic issues are that changes in contact status don't cause events to be
emitted for the associated endpoint. Only dynamic contact add/delete actions
update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds
which is a long time.
This patch makes use of the new transaction timeout feature in r4585 and
provides the following capabilities...
1. A new aor/contact variable 'qualify_timeout' has been added that allows the
user to specify the maximum time in milliseconds to wait for a response to an
OPTIONS message. The default is 3000ms. When the timer expires, the contact is
marked unavailable.
2. Contact status changes are now propagated up to the endpoint as follows...
When any contact is 'Available', the endpoint is marked as 'Reachable'. When
all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The
existing endpoint events are generated appropriately.
ASTERISK-24863 #close
Change-Id: Id0ce0528e58014da1324856ea537e7765466044a
Tested-by: Dmitriy Serov
Tested-by: George Joseph <george.joseph@fairview5.com>
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This change adds the following:
1. A query set implementation. This is an API that allows queries to be executed in parallel and once all have completed a callback is invoked.
2. Unit tests for the query set implementation.
3. An external PJSIP resolver which uses the DNS core API to do NAPTR, SRV, AAAA, and A lookups.
For the resolver it will do NAPTR, SRV, and AAAA/A lookups in parallel. If NAPTR or SRV
are available it will then do more queries. And so on. Preference is NAPTR > SRV > AAAA/A,
with IPv6 preferred over IPv4. For transport it will prefer TLS > TCP > UDP if no explicit
transport has been provided. Configured transports on the system are taken into account to
eliminate resolved addresses which have no hope of completing.
ASTERISK-24947 #close
Reported by: Joshua Colp
Change-Id: I56cb03ce4f9d3d600776f36928e0b3e379b5d71e
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After the "progressinband" value setting of "never" was updated to never send a
183 this separated its use from the "no" value. Since "never" was the default,
but most users probably expect "no" this patch updates the default for the
"progressinband" setting to "no."
ASTERISK-24835 #close
Reported by: Andrew Nagy
Review: https://reviewboard.asterisk.org/r/4606/
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This patch adds a new configuration parameter, 'calltokenexpiration', that
controls how long before an authentication call token is expired. The default
maintains the RFC specified 10 seconds. Setting it to a higher value may be
useful in lossy networks.
Review: https://reviewboard.asterisk.org/r/4588
ASTERISK-24939 #close
Reported by: Y Ateya
patches:
ctoken_configuration.diff submitted by Y Ateya (License 6693)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434563 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Added a new CLI command for res_pjsip that shows both global and system
configuration settings: pjsip show settings
ASTERISK-24918 #close
Reported by: Scott Griepentrog
Review: https://reviewboard.asterisk.org/r/4597/
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This patch adds a new option to cel_pgsl, "usegmtime", which causes timestamps
to be logged in GMT.
Review: https://reviewboard.asterisk.org/r/4571/
ASTERISK-23186 #close
Reported by: Rodrigo Ramirez Norambuena
patches:
cel_pgsql.c_add_usegmtime2.patch submitted by Rodrigo Ramirez Norambuena (License 6577)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434284 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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For some applications - such as SLA - a phone pressing hold should not behave
in the fashion that the Asterisk core would like it to. Instead, the hold
action has some application specific behaviour associated with it - such as
disconnecting the channel that initiated the hold; only playing MoH to channels
in the bridge if the channels are of a particular type, etc.
One way of accomplishing this is to use a framehook to intercept the
hold/unhold frames, raise an event, and eat the frame. Tasty. This patch
accomplishes that using a new dialplan function, HOLD_INTERCEPT.
In addition, some general cleanup of raising hold/unhold Stasis messages was
done, including removing some RAII_VAR usage.
Review: https://reviewboard.asterisk.org/r/4549/
ASTERISK-24922 #close
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messages.
Incoming PJSIP call legs that have not been answered yet send unnecessary
"180 Ringing" or "183 Progress" messages every time a connected line
update happens. If the outgoing channel is also PJSIP then the incoming
channel will always send a "180 Ringing" or "183 Progress" message when
the outgoing channel sends the INVITE.
Consequences of these unnecessary messages:
* The caller can start hearing ringback before the far end even gets the
call.
* Many phones tend to grab the first connected line information and refuse
to update the display if it changes. The first information is not likely
to be correct if the call goes to an endpoint not under the control of the
first Asterisk box.
When connected line first went into Asterisk in v1.8, chan_sip received an
undocumented option "rpid_immediate" that defaults to disabled. When
enabled, the option immediately passes connected line update information
to the caller in "180 Ringing" or "183 Progress" messages as described
above.
* Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or
"183 Progress" messages. The default is "no" to disable sending the
unnecessary messages.
ASTERISK-24781 #close
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/4473/
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This patch fixes previously reverted code that caused binary incompatibility
problems with some modules. And like the original patch it makes sure that
no matter what order the endpoint identifier modules were loaded, priority is
given based on the ones specified in the new global 'endpoint_identifier_order'
option.
ASTERISK-24840
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4489/
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In some countries, privacy laws specify that SMS content cannot be saved by a
provider. This patch adds a new option to the SMS application, 'n', which
prevents the SMS content from being written to the SMS log.
ASTERISK-22591 #close
Reported by: Jan Juergens
patches:
DisableSmsContentLoggingByParam.patch uploaded by Jan Juergens (License 6538)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Due to a break in binary compatibility with some other modules these changes
are being reverted until the issue can be resolved.
ASTERISK-24840
Reported by: Mark Michelson
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It's possible to have a scenario that will create a conflict between endpoint
identifiers. For instance an incoming call could be identified by two different
endpoint identifiers and the one chosen depended upon which identifier module
loaded first. This of course causes problems when, for example, the incoming
call is expected to be identified by username, but instead is identified by ip.
This patch adds a new 'global' option to res_pjsip called
'endpoint_identifier_order'. It is a comma separated list of endpoint
identifier names that specifies the order by which identifiers are processed
and checked.
ASTERISK-24840 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4455/
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This patch adds a new feature to ARI to redirect a channel to another server,
and fixes a few bugs in PJSIP's handling of the Transfer dialplan
application/ARI redirect capability.
*New Feature*
A new operation has been added to the ARI channels resource, redirect. With
this, a channel in a Stasis application can be redirected to another endpoint
of the same underlying channel technology.
*Bug fixes*
In the process of writing this new feature, two bugs were fixed in the PJSIP
stack:
(1) The existing .transfer channel callback had the limitation that it could
only transfer channels to a SIP URI, i.e., you had to pass
'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is
still supported, it is somewhat unintuitive - particularly in a world full
of endpoints. As such, we now also support specifying the PJSIP endpoint to
transfer to.
(2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by
updating its Contact header. Alas, that resulted in the forwarding
destination set by the dialplan application/ARI resource/whatever being
rewritten with very incorrect information. Hence, we now don't bother
updating an outgoing response if it is a 302. Since this took a looong time
to find, some additional debug statements have been added to those modules
that update the Contact headers.
Review: https://reviewboard.asterisk.org/r/4316/
ASTERISK-24015 #close
Reported by: Private Name
ASTERISK-24703 #close
Reported by: Matt Jordan
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across a bridge.
Calling ast_channel_bridge_peer() cannot be done while holding any channel
locks. The reported issue hit the deadlock in chan_iax2, but an audit of
the ast_channel_bridge_peer() calls found three more locations where the
same deadlock can occur.
* Made CHANNEL(peer), res_fax, and the SNMP agent not call
ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I
had to rework the logic to not hold the channel lock.
* Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done
for legacy reasons that no longer apply.
* Removed the iax.conf forcejitterbuffer option. It is now always enabled
when the jitterbuffer option is enabled. If you put a jitter buffer on a
channel it will be on the channel.
ASTERISK-24600 #close
Reported by: Jeff Collell
Review: https://reviewboard.asterisk.org/r/4342/
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CHANGES file.
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This allows for a path to be specified that has a collection of CA
certificates in it.
ASTERISK-24575 #close
Reported by cloos
Patches:
pj-ca-path-trunk.diff uploaded by cloos (License #5956)
Review: https://reviewboard.asterisk.org/r/4344
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* Made the following AMI actions use list API calls for consistency:
Agents
BridgeInfo
BridgeList
BridgeTechnologyList
ConfbridgeLIst
ConfbridgeLIstRooms
CoreShowChannels
DAHDIShowChannels
DBGet
DeviceStateList
ExtensionStateList
FAXSessions
Hangup
IAXpeerlist
IAXpeers
IAXregistry
MeetmeList
MeetmeListRooms
MWIGet
ParkedCalls
Parkinglots
PJSIPShowEndpoint
PJSIPShowEndpoints
PJSIPShowRegistrationsInbound
PJSIPShowRegistrationsOutbound
PJSIPShowResourceLists
PJSIPShowSubscriptionsInbound
PJSIPShowSubscriptionsOutbound
PresenceStateList
PRIShowSpans
QueueStatus
QueueSummary
ShowDialPlan
SIPpeers
SIPpeerstatus
SIPshowregistry
SKINNYdevices
SKINNYlines
Status
VoicemailUsersList
* Incremented the AMI version to 2.7.0.
* Changed astman_send_listack() to not use the listflag parameter and
always set the value to "Start" so the start capitalization is consistent.
i.e., The FAXSessions used "Start" while the rest of the system used
"start". The corresponding complete event always used "Complete".
* Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the
AMI ActionID for all of its list events.
* Fixed off-nominal AMI protocol error in manager_bridge_info(),
manager_parking_status_single_lot(), and
manager_parking_status_all_lots(). Use of astman_send_error() after
responding to the original AMI action request violates the action response
pattern by sending two responses.
* Fixed minor protocol error in action_getconfig() when no requested
categories are found. Each line needs to be formatted as "Header: text".
* Fixed off-nominal memory leak in manager_build_parked_call_string().
* Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail().
ASTERISK-24049 #close
Reported by: Jonathan Rose
Review: https://reviewboard.asterisk.org/r/4315/
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Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This change makes the T.38 negotiation timeout configurable via
't38timeout' in res_fax.conf or FAXOPT(t38timeout). It was previously
hard coded to be 5000 milliseconds.
This change also handles T.38 switch failures by aborting the fax since
in the case where this can happen, both sides have agreed to switch to
T.38 and Asterisk is unable to do so.
Review: https://reviewboard.asterisk.org/r/4320/
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Merged revisions 430415 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 430416 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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With this patch, the following two ARI commands
POST /channels
POST /channels/{id}/continue
Accept a new parameter, label, that can be used to continue to or originate
to a priority label in the dialplan.
Because this is adding a new parameter to ARI commands, the API version of
ARI has been bumped from 1.6.0 to 1.7.0.
This patch comes courtesy of Nir Simionovich from Greenfield Tech. Thanks!
ASTERISK-24412 #close
Reported by Nir Simionovich
Review: https://reviewboard.asterisk.org/r/4285
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Merged revisions 430337 from http://svn.asterisk.org/svn/asterisk/branches/13
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The current behavior of 'pjsip send unregister' is to send the unregister
(REGISTER with 0 exp) but let the next scheduled register proceed normally.
I don't think that's a good idea. If you unregister, it should stay
unregistered until you decide to start registrations again. So this patch
just adds a cancel_registration call to the current unregister_task to
cancel the timer.
Of course, now you need a way to start registration again so I've added
a 'pjsip send register' command that unregisters and cancels any existing
registration (the same as send unregister), then sends an immediate
registration and starts the timer back up again.
Both changes also ripple to AMI. There's a new PJSIPRegister command.
There's no harm in calling either command repeatedly. They don't care
about the actual state.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4301/
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Merged revisions 430223 from http://svn.asterisk.org/svn/asterisk/branches/13
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