* Declare 'requires' and 'enhances' text fields on module info structure.
* Rename 'nonoptreq' to 'optional_modules'.
* Update doxygen comments.
Still need to investigate dependencies among modules I cannot compile.
Change-Id: I3ad9547a0a6442409ff4e352a6d897bef2cc04bf
One of the patches for ASTERISK_27147 introduced a deadlock regression.
When the connection oriented transport shut down, the code attempted to
remove the associated contact. However, that same transport had just
requested a registration that we hadn't responded to yet. Depending
upon timing we could deadlock.
* Made send the REGISTER response after we completed processing the
request contacts and released the AOR lock to avoid the deadlock.
ASTERISK-27391
Change-Id: I89a90f87cb7a02facbafb44c75d8845f93417364
When "rewrite_contact" is enabled, the "max_contacts" count option can
block re-registrations because the source port from the endpoint can be
random. When the re-registration is blocked, the endpoint may give up
re-registering and require manual intervention.
* The "remove_existing" option now allows a registration to succeed by
displacing any existing contacts that now exceed the "max_contacts" count.
Any removed contacts are the next to expire. The behaviour change is
beneficial when "rewrite_contact" is enabled and "max_contacts" is greater
than one. The removed contact is likely the old contact created by
"rewrite_contact" that the device is refreshing.
ASTERISK-27192
Change-Id: I64c107a10b70db1697d17136051ae6bf22b5314b
The fix for the issue is broken up into three parts.
This is part two which handles the server side of REGISTER requests when
rewrite_contact is enabled. Any registered reliable transport contact
becomes invalid when the transport connection becomes disconnected.
* Monitor the rewrite_contact's reliable transport REGISTER contact for
shutdown. If it is shutdown then the contact must be removed because it
is no longer valid. Otherwise, when the client attempts to re-REGISTER it
may be blocked because the invalid contact is there. Also if we try to
send a call to the endpoint using the invalid contact then the endpoint is
not likely to see the request. The endpoint either won't be listening on
that port for new connections or a NAT/firewall will block it.
* Prune any rewrite_contact's registered reliable transport contacts on
boot. The reliable transport no longer exists so the contact is invalid.
* Websockets always rewrite the REGISTER contact address and the transport
needs to be monitored for shutdown.
* Made the websocket transport set a unique name since that is what we use
as the ao2 container key. Otherwise, we would not know which transport we
find when one of them shuts down. The names are also used for PJPROJECT
debug logging.
* Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state
event. Now the global keep_alive_interval option, initially idle shutdown
timer, and the server REGISTER contact monitor can work on wetsocket
transports.
* Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction.
Now initially idle websockets will automatically shutdown.
ASTERISK-27147
Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4
Most uses of CMP_STOP are superfluous and are only respected when
OBJ_MULTIPLE is used to search the container.
Change-Id: I20571a202ec0aa1098bb2749eeba18de7ca110b8
This reverts commit 6492e91392.
The change in question was intended to prevent the need to reload in
order to update qualifies on contacts when an AOR changes. However, this
ended up causing a deadlock instead.
Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e
Prior to this change, qualifies would only update in the following
cases:
* A reload of res_pjsip.so was issued.
* A dynamic contact was re-registered after its AOR's qualify_frequency
had been changed
This does not work well if you are using realtime for your AORs. You can
update your database to have a new qualify_frequency, but the permanent
contacts on that AOR will not have their qualifies updated. And the
dynamic contacts on that AOR will not have their qualifies updated until
the next registration, which could be a long time.
This change seeks to fix this problem by making it so that whenever AOR
configuration is applied, the contacts pertaining to that AOR have their
qualifies updated.
Additions from this patch:
* AOR sorcery objects now have an apply handler that calls into a newly
added function in the OPTIONS code. This causes all contacts
associated with that AOR to re-schedule qualifies.
* When it is time to qualify a contact, the OPTIONS code checks to see
if the AOR can still be retrieved. If not, then qualification is
canceled on the contact.
Alterations from this patch:
* The registrar code no longer updates contact's qualify_frequence and
qualify_timeout. There is no point to this since those values already
get updated when the AOR changes.
* Reloading res_pjsip.so no longer calls the OPTIONS initialization
function. Reloading res_pjsip.so results in re-loading AORs, which
results in re-scheduling qualifies.
Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121
The PJSIPShowRegistrationsInbound AMI command was just dumping out
all AORs which was pretty useless and resource heavy since it had
to get all endpoints, then all aors for each endpoint, then all
contacts for each aor.
PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail
events which meets the intended purpose of the other command and has
significantly less overhead. Also, some additional fields that were
added to Contact since the original creation of the ContactStatusDetail
event have been added to the end of the event.
For compatibility purposes, PJSIPShowRegistrationsInbound is left
intact.
ASTERISK-26644 #close
Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a
This implements the chan_sip legacy_useroption_parsing option but with a
better name.
* Made the caller-id number and redirecting number strings obtained from
incoming SIP URI user fields always truncated at the first semicolon.
People don't care about anything after the semicolon showing up on their
displays even though the RFC allows the semicolon.
ASTERISK-26316 #close
Reported by: Kevin Harwell
Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62
Create an alternative to ast_sorcery_generic_alloc which uses astobj2
shared locking. Use this new method for the 'struct ast_sip_aor' allocator.
Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f
The named aor lock was always being locked for writes so a rwlock adds no
benefit and may be slower because rwlocks are biased toward read locking.
Change-Id: I8c5c2c780eb30ce5441832257beeb3506fd12b28
This patch fixes a race condition processing received REGISTER requests
and their retransmissions caused by REGISTER requests being processed by
two threads. The "sip_transaction Unable to register REGISTER transaction
(key exists)" message is a notable symptom of this issue.
This issue was more likely to happen before the pjsip/distributor
serializers were created. Instead of steps one and two below placing the
REGISTER messages into the same pjsip/distributor they were placed in
random pjsip/default serializers.
1) REGISTER requests come in and get placed on the pjsip/distributor
serializer.
2) Before the first request is processed a retransmission comes in and is
placed on the same pjsip/distributor serializer.
3) The first request goes up the pjsip stack and is then shunted off to
the pjsip/aor/<aor> serializer.
4) Before the first request is completed processing in the pjsip/aor/<aor>
serializer, the second request goes up the pjsip stack and is also shunted
off to the pjsip/aor/<aor> serializer.
5) The first request completes processing and sends out its response.
6) The second request completes processing and tries to send out its
response but pjlib complains that the REGISTER transaction key already
exists.
7) Sadness ensues.
* The race is eliminated by removing the pjsip/aor/<aor> serializer and
continuing the processing in the pjsip/distributor serializer. Now any
retransmissions queued in the pjsip/distributor serializer will be
processed after the first message is completely processed.
ASTERISK-26088 #close
Reported by: Richard Mudgett
Change-Id: I842d714346088bf717ea27437f1dd85bff0bab5a
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
The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_*
and res_pjsip_registrar modules should load ASAP
to avoid "No matching endpoint found" for legitimate endpoint.
ASTERISK-25994
Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b
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
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
Recent changes to the PJSIP registrar resulted in tests failing due to
missing AOR_CONTACT_ADDED test events. The reason for this was that the
user_agent string had junk values in it, resulting in being unable to
generate the event.
I'm going to be honest here, I have no idea why this was happening. Here
are the steps needed for the user_agent variable to get messed up:
* REGISTER is received
* First contact in the REGISTER results in a contact being removed
* Second contact in the REGISTER results in a contact being added
* The contact, AOR, expiration, and user agent all have to be passed as
format parameters to the creation of a string. Any subset of those
parameters would not be enough to cause the problem.
Looking into what was happening, the thing that struck me as odd was
that the user_agent variable was meant to be set to the value of the
User-Agent SIP header in the incoming REGISTER. However, when removing a
contact, the user_agent variable would be set (via ast_strdupa inside a
loop) to the stored contact's user_agent. This means that the
user_agent's value would be incorrect when attempting to process further
contacts in the incoming REGISTER.
The fix here is to use a different variable for the stored user agent
when removing a contact. Correcting the behavior to be correct also
means the memory usage is less weird, and the issue no longer occurs.
ASTERISK-25929 #close
Reported by Joshua Colp
Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08
Due to some ignored return values, Asterisk could crash if processing an
incoming REGISTER whose contact URI was above a certain length.
ASTERISK-25707 #close
Reported by George Joseph
Patches:
0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch
AST-2016-004
Change-Id: I3ea7cee16f29c8088794de3085ca7523c1c4833d
Contact expiration can occur in several places: res_pjsip_registrar,
res_pjsip_registrar_expire, and automatically when anyone calls
ast_sip_location_retrieve_aor_contact. At the same time, res_pjsip_registrar
may also be attempting to renew or add a contact. Since none of this was locked
it was possible for one thread to be renewing a contact and another thread to
expire it immediately because it was working off of stale data. This was the
casue of intermittent registration/inbound/nominal/multiple_contacts test
failures.
Now, the new named lock functionality is used to lock the aor during contact
expire and add operations and res_pjsip_registrar_expire now checks the
expiration with the lock held before deleting the contact.
ASTERISK-25885 #close
Reported-by: Josh Colp
Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059
The res_pjsip_mwi previously required a reload to set up the proper
subscriptions to allow unsolicited MWI to work. This change
makes it so the act of registering will also cause this to occur.
This is particularly useful if realtime is involved as no reload
needs to occur within Asterisk to cause the MWI information
to get sent.
ASTERISK-25180 #close
Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252
Asterisk had an issue where retransmissions of MESSAGE requests resulted in
Asterisk processing the retransmission as if it were a new MESSAGE request.
This patch fixes the issue by creating a transaction in PJSIP on the incoming
request. This way, if a retransmission arrives, the PJSIP transaction layer
will resend the response and Asterisk will not ever see the retransmission.
ASTERISK-24920 #close
Reported by Mark Michelson
Review: https://reviewboard.asterisk.org/r/4532/
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Some implementations don't pay attention to the expires for individual contacts.
In this case they may consider the lack of an Expires header in the 200 OK as
unregistered. This change makes it so if an Expires header is present in the REGISTER
we will add one in the 200 OK.
ASTERISK-24785 #close
Reported by: Ross Beer
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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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A number of various PJSIP AMI actions were failing to parse out and place the
ActionID into their responses. This patch updates the various PJSIP actions
such that the passed in ActionID is emitted on any event list complete events,
as well as any intermediate events created as a result of the action.
#ASTERISK-23947 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/3675/
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There was an underlying issue in a realtime backend where database updates
would fail. Since we were not checking for failure, we would end up in a
strange state where the old database entry was still present but Asterisk
thought that it had been updated. Now when an entry fails to update, we
print a warning and delete the old contact from sorcery so there is no
mismatch between foreground and backend state.
Patches:
res_pjsip_registrar.patch by John Hardin (License #6512)
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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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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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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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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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While handling a registration request a race condition could occur if/when two+
clients registered at the same time. This happened when one request obtained a
copy of the current contacts for an AOR and another request did the same before
the first request updated. Thus the second would update and overwrite the first
(or vice-versa depending on which actually updated first). In the case of it
being the same contact two "add" events would be raised.
pjsip registration handling is now serialized to alleviate this issue.
(closes issue AST-1213)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2860/
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The general gist is to have a clear boundary between old SIP stuff
and new SIP stuff by having the word "SIP" for old stuff and "PJSIP"
for new stuff. Here's a brief rundown of the changes:
* The word "Gulp" in dialstrings, functions, and CLI commands is now
"PJSIP"
* chan_gulp.c is now chan_pjsip.c
* Function names in chan_gulp.c that were "gulp_*" are now "chan_pjsip_*"
* All files that were "res_sip*" are now "res_pjsip*"
* The "res_sip" directory is now "res_pjsip"
* Files in the "res_pjsip" directory that began with "sip_*" are now "pjsip_*"
* The configuration file is now "pjsip.conf" instead of "res_sip.conf"
* The module info for all PJSIP-related files now uses "PJSIP" instead of "SIP"
* CLI and AMI commands created by Asterisk's PJSIP modules now have "pjsip" as
the starting word instead of "sip"
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Corey Farrell
Richard Mudgett
Mark Michelson
George Joseph
zuul
Alexei Gradinari
George Joseph
Joshua Colp
Rodrigo Ramírez Norambuena
Kinsey Moore
Matthew Jordan
Kevin Harwell
John Bigelow