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This is useful for configuring multiple permanent contacts for an AOR when using
realtime AORs.
Review: https://reviewboard.asterisk.org/r/3462
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configured.
This change fixes a problem where permanent contacts being qualified were not
being updated. This was caused by the permanent contacts getting a uuid and not a
known identifier, causing an inability to look them up when updating in the
qualify code. A bug also existed where the new configuration may not be available
immediately when updating qualifies.
(closes issue ASTERISK-23514)
Reported by: Richard Mudgett
Review: https://reviewboard.asterisk.org/r/3448/
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* Remove unused RAII_VAR() declarations. The compiler cannot catch these
because the cleanup function "references" the unused variable. Some
actually allocated and released resources that were never used.
* Fixed some whitespace issues in stasis_bridges.c.
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This commit contains several changes to sorcery:
1) Application of sorcery configuration based on module name is automatically performed
when sorcery is opened for a module.
2) Sorcery will not attempt to apply the same wizard to an object type more than once.
3) Sorcery gives more exact results when attempting to apply a wizard, whether as the
default or based on configuration.
Sorcery unit tests still pass for me after making these changes.
Review: https://reviewboard.asterisk.org/r/3326
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contact.
* Fixed bad use of ao2_find() in on_endpoint().
* Replaced use of find_endpoints() with find_an_endpoint() since only the
first found endpoint is ever needed.
* Fixed qualify_contact_cb() to update the contact with the aor
authenticate_qualify setting. Otherwise, permanent contacts in the aor
type sections would have a config line order dependancy.
* Fixed off nominal path contact ref leak in qualify_contact(). The
comment saying the unref is not needed was wrong.
* Fixed off nominal path use of the endpoint parameter if it is NULL in
send_out_of_dialog_request().
* Added missing off nominal path unref of pjsip tdata in
send_out_of_dialog_request().
* Fixed off nominal path failing to call the callback in send_request_cb()
when the request is challenged for authentication.
* Eliminated silly RAII_VAR() use in qualify_contact_cb().
* Updated ast_sip_send_request() doxygen to better reflect reality.
(closes issue ASTERISK-23254)
Reported by: rmudgett
Review: https://reviewboard.asterisk.org/r/3381/
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failure.
Thanks Walter Doekes!
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This change enables DNS client support within PJSIP. System
nameservers are automatically discovered using res_init or
res_ninit. If this fails then PJSIP will resort to using
gethostbyname for resolution.
By enabling this support we gain SRV support, failover, and
weight support.
(closes issue ASTERISK-23435)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3343/
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These changes were still up for review and have not been approved
yet. I must have had the changes in my working copy when making
a different change.
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Playing back a file to a channel in an ARI bridge would attempt to wait until
the playback concluded before returning. The method used involved signaling the
waiting thread in the ARI custom playback function.
The problem with this is that there were some corner cases that were not accounted for:
* If a bridge channel could not be found, then we never would attempt the playback but
would still attempt to wait for the playback to complete.
* If the bridge playfile action failed to queue, we would still attempt to wait for the
playback to complete.
* If the bridge playfile action were queued but some circumstance caused the playback
not to occur (the bridge dies, the channel is removed from the bridge), then we would
never be notified.
The solution to this is to move the waiting logic into the bridge code. A new bridge
API function is added to queue a synchronous action on a bridge. The waiting thread
is notified when the queued frame has been freed, either due to an error occurring
or due to successful playback. As a failsafe, the waiting thread has a 10 minute
timeout just in case there is a frame leak somewhere.
Review: https://reviewboard.asterisk.org/r/3338
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(closes issue ASTERISK-23235)
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3324/
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request will have an endpoint.
This change removes the assumption that an outgoing request will always
have an endpoint and makes the authenticate_qualify option work once again.
(closes issue ASTERISK-23210)
Reported by: Joshua Colp
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Created the 'pjsip show channel' and 'pjsip show contact' commands.
Refactored out the hated ast_hashtab. Replaced with ao2_container.
Cleaned up function naming. Internal only, no public name changes.
Cleaned up whitespace and brace formatting in cli code.
Changed some NULL checking from "if"s to ast_asserts.
Fixed some register/unregister ordering to reduce deadlock potential.
Fixed ast_sip_location_add_contact where the 'name' buffer was too short.
Fixed some self-assignment issues in res_pjsip_outbound_registration.
(closes issue ASTERISK-23276)
Review: http://reviewboard.asterisk.org/r/3283/
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In order to prevent confusion over the allow and disallow
list of codecs being the same an option for registering a
field as an alias is added. The alias field will be read
from the configuration file, but afterwards is not listed
as a known field. With disallow set as an alias, the CLI
command pjsip show endpoint # will list the allow= field,
but not the disallow field.
(closes issue ASTERISK-23092)
Review: https://reviewboard.asterisk.org/r/3193/
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This patch creates the AST_SORCERY dialplan function which allows someone to
retrieve any value from a sorcery-based config file. It's similar to
AST_CONFIG.
The creation of the function itself was fairly straightforward but it required
changes to the underlying sorcery infrastructure that rippled into individual
sorcery objects. The changes stemmed from inconsistencies in how sorcery
created ast_variable objectsets from sorcery objects and the inconsistency
in how individual objects used that feature especially when it came to
parameters that can be specified multiple times like contact in aor and match
in identify. You can read more here...
http://lists.digium.com/pipermail/asterisk-dev/2014-February/065202.html
So, what this patch does, besides actually creating the AST_SORCERY function,
is the following...
* Creates ast_variable_list_append which is a helper to append one ast_variable
list to another.
* Modifies the ast_sorcery_object_field_register functions to accept the
already-defined sorcery_fields_handler callback.
* Modifies ast_sorcery_objectset_create to accept a parameter indicating return
type preference...a single ast_variable with all values concatenated or an
ast_variable list with multiple entries. Also fixed a few bugs.
* Modifies individual sorcery object implementations to use the new function
definition of the ast_sorcery_object_field_register functions.
* Modifies location.c and res_pjsip_endpoint_identifier_ip.c to implement
sorcery_fields_handler handlers so they return multiple occurrences as an
ast_variable_list.
* Added a whole bunch of tests to test_sorcery.
(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3254/
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Transport TOS values were interpreted as DSCP values without being documented
as such. Endpoint TOS values (tos_audio/tos_video) behaved normally as TOS
values have historically. This patch makes the transport TOS values behave as
TOS values and makes all TOS values readable as string values (e.g. AF11).
In addition, alembic scripts have been updated to use the proper field types
for all TOS/COS values.
(issue ASTERISK-23235)
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3304/
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Added 'show registrations' and 'show contacts' to pjsip cli to make things
a little more consistent. The output is exactly the same as the list command.
Just needed to add entries to their respective ast_cli_entry structures.
(closes issue ASTERISK-23275)
Review: http://reviewboard.asterisk.org/r/3210/
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Fixed memory leaks in ast_sip_cli_print_sorcery_objectset and
ast_variable_list_sort.
(closes issue ASTERISK-23266)
Review: http://reviewboard.asterisk.org/r/3200/
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In order to retrieve an arbitrary sorcery instance from a dialplan function
(or any place else) there needs to be a registry of sorcery instances.
ast_sorcery_init now creates a hashtab as a registry.
ast_sorcery_open now checks the hashtab for an existing sorcery instance
matching the caller's module name. If it finds one, it bumps the
refcount and returns it. If not, it creates a new sorcery instance,
adds it to the hashtab, then returns it.
ast_sorcery_retrieve_by_module_name is a new function that does a hashtab
lookup by module name. It can be called by the future dialplan function.
res_pjsip/config_system needed a small change to share the main res_pjsip
sorcery instance.
tests/test_sorcery was updated to include a test for the registry.
(closes issue ASTERISK-22537)
Review: http://reviewboard.asterisk.org/r/3184/
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When an endpoint sends a REGISTER request to Asterisk, we now will
associate the User-Agent header with all contacts that were bound in
that REGISTER request.
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If an AOR has no permanent contacts, then the
permanent_contacts container is never allocated.
This makes the code safe in the face of NULLs.
I also changed the variable that counts contacts
from "num" to "total_contacts" since there are now
two variables that are indicate numbers of things.
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* Adds identify, transport, and registration support to the PJSIP CLI.
* Creates three additional callbacks, one for an iterator, one for a
comparator, and one for a container. This eliminates the link dependency
from higher level modules to lower level ones.
* Eliminates duplicate sorting in PJSIP CLI commands.
* Cleans up PJSIP CLI output formatting.
* Pushes CLI command registration down to the implementing source file.
* Adds several ast_sip_destroy_sorcery functions to complement existing
ast_sip_sorcery_initialize functions. The destroy functions unregister
PJSIP CLI commands and PJSIP CLI formatters.
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3104/
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If the global section was not specified in pjsip.conf then the configuration
object does not exist in sorcery so when retrieving "debug" option it would
return NULL. Then the NULL result was passed to ast_false utils function
which would return false because it wasn't set to some representation of
false, thus enabling sip debug logging. Made it so if the global config object
does not exist then it will return a default of "no" for sip debugging.
(issue ASTERISK-23038)
Reported by: Rusty Newton
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(closes issue ASTERISK-23168)
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/3143/
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Added a "debug" configuration option for res_pjsip that when set to "yes"
enables SIP messages to be logged. It is specified under the "system" type.
Also added an alembic script to add the option to realtime.
(closes issue ASTERISK-23038)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3148/
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When the PJSIP pubsub framework was created, subscription handlers were required
to state what event they handled along with what body types they knew how to
generate. While this serves well when implementing a base RFC, it has problems
when trying to extend the body to support non-standard or proprietary body
elements. The code also was NOTIFY-specific, meaning that when the time comes
that we start writing code to send out PUBLISH requests with MWI or presence
bodies, we would likely find ourselves duplicating code that had previously been
written.
This changeset introduces the concept of body generators and body supplements. A
body generator is responsible for allocating a native structure for a given body
type, providing the primary body content, converting the native structure to a
string, and deallocating resources. A body supplement takes the primary body
content (the native structure, not a string) generated by the body generator and
adds nonstandard elements to the body. With these elements living in their own
module, it becomes easy to extend our support for body types and to re-use
resources when sending a PUBLISH request.
Body generators and body supplements register themselves with the pubsub core,
similar to how subscription and publish handlers had done. Now, subscription
handlers do not need to know what type of body content they generate, but they
still need to inform the pubsub core about what the default body type for a
given event package is. The pubsub core keeps track of what body generators and
body supplements have been registered. When a SUBSCRIBE arrives, the pubsub core
will check that there is a subscription handler for the event in the SUBSCRIBE,
then it will check that there is a body generator that can provide the content
specified in the Accept header(s).
Because of the nature of body generators and supplements, it means
res_pjsip_exten_state and res_pjsip_mwi have been completely gutted. They no
longer worry about body types, instead calling
ast_sip_pubsub_generate_body_content() when they need to generate a NOTIFY body.
Review: https://reviewboard.asterisk.org/r/3150
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On some systems the values for INFINITY and NAN are not defined thus causing
a build error on those systems. Added definitions for those if they had
not previously been defined.
(closes issue ASTERISK-23056)
Reported by: capouch
Patches:
inf-nan-patch.txt uploaded by capouch (license 6564)
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If an endpoint had previously dynamically registered a contact and the contact
information was successfully stored in astdb then upon restart the qualify
notifications would not be sent out if the qualify_frequency was set. This was
due to the fact that only permanent contacts were being checked and scheduled
for qualifies on startup. Modified the code to check and schedule all
registered contacts at startup.
(closes issue ASTERISK-23062)
Reported by: Rusty Newton
Review: https://reviewboard.asterisk.org/r/3124/
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This adds Path support to chan_pjsip in res_pjsip_path.c with minimal
additions in res_pjsip_registrar.c to store the path and additions in
res_pjsip_outbound_registration.c to enable advertisement of path
support to registrars and intervening proxies.
Path information is stored on contacts and is enabled via Address of
Record (AoRs) and Registration configuration sections.
While adding path support, it became necessary to be able to add SIP
supplements that handled messages outside of sessions, so a framework
for handling these types of hooks was added in parallel to the
already-existing session supplements and several senders of
out-of-dialog requests were refactored as a result.
(closes issue ASTERISK-21084)
Review: https://reviewboard.asterisk.org/r/3050/
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We were not including an authentication challenge when sending a 401 response
to unmatched endpoints. This was due to the conversion to use a vector for
authentication section names on an endpoint. The vector for artificial endpoints
was empty, resulting in the challenge being sent back containing no challenges.
This is worked around by placing a bogus value in the artificial endpoint's auth
vector. This value is never looked up by anything, since they instead will directly
call ast_sip_get_artificial_auth().
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This fixes several issues with the new res_pjsip CLI tab completion
such as output of headers during tab completion and being able to
tab-complete more items than the code actually handled (further items
would simply be ignored).
(closes issue ASTERISK-23081)
Review: https://reviewboard.asterisk.org/r/3115/
Reported by: xrobau
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Previously, this printed a UUID, which was not very clear when dealing
with an artificial endpoint.
Review: https://reviewboard.asterisk.org/r/3113
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Added a new 'set_var' option for ast_sip_endpoint(s). For each variable
specified that variable gets set upon creation of a pjsip channel involving
the endpoint.
(closes issue ASTERISK-22868)
Reported by: Joshua Colp
Review: https://reviewboard.asterisk.org/r/3095/
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When wanting to pass *free as a function pointer, ast_free_ptr has to be used
instead of ast_free. This allows it to be compiled with MALLOC_DEBUG enabled.
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Implements the following cli commands:
pjsip list aors
pjsip list auths
pjsip list channels
pjsip list contacts
pjsip list endpoints
pjsip show aor(s)
pjsip show auth(s)
pjsip show channels
pjsip show endpoint(s)
Also...
Minor modifications made to the AMI command implementations to facilitate
reuse.
New function ast_variable_list_sort added to config.c and config.h to implement
variable list sorting.
(issue ASTERISK-22610)
patches:
pjsip_cli_v2.patch uploaded by george.joseph (License 6322)
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* Fixed several places where ao2_iterator_destroy() was not called.
* Fixed several iterator loop object variable reference problems.
* Fixed res_parking AMI actions returning non-zero. Only the AMI logoff
action can return non-zero.
Review: https://reviewboard.asterisk.org/r/3087/
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know about.
Under normal conditions it is unlikely we will ever receive a response for a transaction
or dialog we don't know about but if any are received ignore them.
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Objects which are involved in SIP request creation and sending
now allow an outbound proxy to be specified. For cases where
an endpoint is used the outbound proxy specified there will
be applied.
(closes issue ASTERISK-22673)
Reported by: Antti Yrjola
Review: https://reviewboard.asterisk.org/r/3022/
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In some cases messages need to be sent to a direct URI (sip:<ip address>). This
patch adds in that support by using a default outbound endpoint. When sending
messages, if no endpoint can be found then the default one is used.
To facilitate this a new default_outbound_endpoint option was added to the
globals section for pjsip.conf.
Review: https://reviewboard.asterisk.org/r/2944/
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Since Asterisk has a vector API now, places where arrays are manually
resized don't really make sense any more. Since the auth work in PJSIP
was freshly-written, it was easy to reform it to use a vector.
Review: https://reviewboard.asterisk.org/r/3044
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* Make ast_sorcery_observer_remove() accept a const callbacks struct.
* Make ast_sorcery_observer_remove() tolerant of the sorcery parameter
being NULL. Now it can be called within a module unload routine if the
sorcery initialization fails.
* Fix ast_sorcery_observer_add() to fail if the container link fails.
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Transport type determination for security events has been simplified to use
the type present on the message itself instead of searching through configured
transports to find the transport used.
The actual WebSocket transport has also been simplified. It now leverages the
existing PJSIP transport manager for finding the active WebSocket transport
for outgoing messages. This removes the need for res_pjsip_transport_websocket
to store a mapping itself.
(closes issue ASTERISK-22897)
Reported by: Max E. Reyes Vera J.
Review: https://reviewboard.asterisk.org/r/3036/
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The action taken when a redirect occurs is now configurable on a
per-endpoint basis. The redirect can either be treated as a redirect
to a local extension, to a URI that is dialed through the Asterisk
core, or to a URI that is dialed within PJSIP itself.
(closes issue ASTERISK-21710)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2963/
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Certain options available that specify a SIP URI perform validation
on the provided URI using the PJSIP URI parser. This operation
requires that the thread executing it be registered with the PJLIB
library. During reloads this was done on a thread which was NOT
registered with it.
This fixes the problem by creating a task which reloads the
configuration on a PJSIP thread.
(closes issue ASTERISK-22923)
Reported by: Anthony Messina
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Some options (such as call_group and pickup_group) share the same configuration
handler and decide what logic to use based on the name of the option. These
handlers were not updated to check for the new option names and were treating
the options as invalid.
This change simply updates the handlers with the proper names of the options.
(closes issue ASTERISK-22922)
Reported by: Anthony Messina
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Created the following AMI commands and corresponding events for res_pjsip:
PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few
select attributes on each.
Events:
EndpointList - for each endpoint a few attributes.
EndpointlistComplete - after all endpoints have been listed.
PJSIPShowEndpoint - Provides a detail list of attributes for a specified
endpoint.
Events:
EndpointDetail - attributes on an endpoint.
AorDetail - raised for each AOR on an endpoint.
AuthDetail - raised for each associated inbound and outbound auth
TransportDetail - transport attributes.
IdentifyDetail - attributes for the identify object associated with
the endpoint.
EndpointDetailComplete - last event raised after all detail events.
PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound
registrations.
Events:
InboundRegistrationDetail - inbound registration attributes for each
registration.
InboundRegistrationDetailComplete - raised after all detail records have
been listed.
PJSIPShowRegistrationsOutbound - Provides a detail listing of all outbound
registrations.
Events:
OutboundRegistrationDetail - outbound registration attributes for each
registration.
OutboundRegistrationDetailComplete - raised after all detail records
have been listed.
PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions
and their attributes.
Events:
SubscriptionDetail - on each subscription detailed attributes
SubscriptionDetailComplete - raised after all detail records have
been listed.
PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound
subscriptions and their attributes.
Events:
SubscriptionDetail - on each subscription detailed attributes
SubscriptionDetailComplete - raised after all detail records have
been listed.
(issue ASTERISK-22609)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2959/
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Updated the alembic script for pjsip. Also, the dtls config parsing stuff was
expecting strings with no underscores, so removed the underscores from the
option name before passing it to the parser.
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Renamed, where appropriate, the configuration options for chan/res_pjsip to use
snake case (compound words separated by an underscore). For example, faxdetect
will become fax_detect, recordofffeature will become record_off_feature, etc...
Review: https://reviewboard.asterisk.org/r/3002/
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